diff options
Diffstat (limited to 'sound')
387 files changed, 44360 insertions, 10630 deletions
diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 270790d384e2..4cedc6950d72 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -997,45 +997,10 @@ static void onyx_exit_codec(struct aoa_codec *codec) onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx); } -static int onyx_create(struct i2c_adapter *adapter, - struct device_node *node, - int addr) -{ - struct i2c_board_info info; - struct i2c_client *client; - - memset(&info, 0, sizeof(struct i2c_board_info)); - strlcpy(info.type, "aoa_codec_onyx", I2C_NAME_SIZE); - info.addr = addr; - info.platform_data = node; - client = i2c_new_device(adapter, &info); - if (!client) - return -ENODEV; - - /* - * We know the driver is already loaded, so the device should be - * already bound. If not it means binding failed, which suggests - * the device doesn't really exist and should be deleted. - * Ideally this would be replaced by better checks _before_ - * instantiating the device. - */ - if (!client->driver) { - i2c_unregister_device(client); - return -ENODEV; - } - - /* - * Let i2c-core delete that device on driver removal. - * This is safe because i2c-core holds the core_lock mutex for us. - */ - list_add_tail(&client->detected, &client->driver->clients); - return 0; -} - static int onyx_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - struct device_node *node = client->dev.platform_data; + struct device_node *node = client->dev.of_node; struct onyx *onyx; u8 dummy; @@ -1071,40 +1036,6 @@ static int onyx_i2c_probe(struct i2c_client *client, return -ENODEV; } -static int onyx_i2c_attach(struct i2c_adapter *adapter) -{ - struct device_node *busnode, *dev = NULL; - struct pmac_i2c_bus *bus; - - bus = pmac_i2c_adapter_to_bus(adapter); - if (bus == NULL) - return -ENODEV; - busnode = pmac_i2c_get_bus_node(bus); - - while ((dev = of_get_next_child(busnode, dev)) != NULL) { - if (of_device_is_compatible(dev, "pcm3052")) { - const u32 *addr; - printk(KERN_DEBUG PFX "found pcm3052\n"); - addr = of_get_property(dev, "reg", NULL); - if (!addr) - return -ENODEV; - return onyx_create(adapter, dev, (*addr)>>1); - } - } - - /* if that didn't work, try desperate mode for older - * machines that have stuff missing from the device tree */ - - if (!of_device_is_compatible(busnode, "k2-i2c")) - return -ENODEV; - - printk(KERN_DEBUG PFX "found k2-i2c, checking if onyx chip is on it\n"); - /* probe both possible addresses for the onyx chip */ - if (onyx_create(adapter, NULL, 0x46) == 0) - return 0; - return onyx_create(adapter, NULL, 0x47); -} - static int onyx_i2c_remove(struct i2c_client *client) { struct onyx *onyx = i2c_get_clientdata(client); @@ -1117,16 +1048,16 @@ static int onyx_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id onyx_i2c_id[] = { - { "aoa_codec_onyx", 0 }, + { "MAC,pcm3052", 0 }, { } }; +MODULE_DEVICE_TABLE(i2c,onyx_i2c_id); static struct i2c_driver onyx_driver = { .driver = { .name = "aoa_codec_onyx", .owner = THIS_MODULE, }, - .attach_adapter = onyx_i2c_attach, .probe = onyx_i2c_probe, .remove = onyx_i2c_remove, .id_table = onyx_i2c_id, diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index 8e63d1f35ce1..c491ae0f749c 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -883,43 +883,10 @@ static void tas_exit_codec(struct aoa_codec *codec) } -static int tas_create(struct i2c_adapter *adapter, - struct device_node *node, - int addr) -{ - struct i2c_board_info info; - struct i2c_client *client; - - memset(&info, 0, sizeof(struct i2c_board_info)); - strlcpy(info.type, "aoa_codec_tas", I2C_NAME_SIZE); - info.addr = addr; - info.platform_data = node; - - client = i2c_new_device(adapter, &info); - if (!client) - return -ENODEV; - /* - * We know the driver is already loaded, so the device should be - * already bound. If not it means binding failed, and then there - * is no point in keeping the device instantiated. - */ - if (!client->driver) { - i2c_unregister_device(client); - return -ENODEV; - } - - /* - * Let i2c-core delete that device on driver removal. - * This is safe because i2c-core holds the core_lock mutex for us. - */ - list_add_tail(&client->detected, &client->driver->clients); - return 0; -} - static int tas_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - struct device_node *node = client->dev.platform_data; + struct device_node *node = client->dev.of_node; struct tas *tas; tas = kzalloc(sizeof(struct tas), GFP_KERNEL); @@ -953,47 +920,6 @@ static int tas_i2c_probe(struct i2c_client *client, return -EINVAL; } -static int tas_i2c_attach(struct i2c_adapter *adapter) -{ - struct device_node *busnode, *dev = NULL; - struct pmac_i2c_bus *bus; - - bus = pmac_i2c_adapter_to_bus(adapter); - if (bus == NULL) - return -ENODEV; - busnode = pmac_i2c_get_bus_node(bus); - - while ((dev = of_get_next_child(busnode, dev)) != NULL) { - if (of_device_is_compatible(dev, "tas3004")) { - const u32 *addr; - printk(KERN_DEBUG PFX "found tas3004\n"); - addr = of_get_property(dev, "reg", NULL); - if (!addr) - continue; - return tas_create(adapter, dev, ((*addr) >> 1) & 0x7f); - } - /* older machines have no 'codec' node with a 'compatible' - * property that says 'tas3004', they just have a 'deq' - * node without any such property... */ - if (strcmp(dev->name, "deq") == 0) { - const u32 *_addr; - u32 addr; - printk(KERN_DEBUG PFX "found 'deq' node\n"); - _addr = of_get_property(dev, "i2c-address", NULL); - if (!_addr) - continue; - addr = ((*_addr) >> 1) & 0x7f; - /* now, if the address doesn't match any of the two - * that a tas3004 can have, we cannot handle this. - * I doubt it ever happens but hey. */ - if (addr != 0x34 && addr != 0x35) - continue; - return tas_create(adapter, dev, addr); - } - } - return -ENODEV; -} - static int tas_i2c_remove(struct i2c_client *client) { struct tas *tas = i2c_get_clientdata(client); @@ -1011,16 +937,16 @@ static int tas_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id tas_i2c_id[] = { - { "aoa_codec_tas", 0 }, + { "MAC,tas3004", 0 }, { } }; +MODULE_DEVICE_TABLE(i2c,tas_i2c_id); static struct i2c_driver tas_driver = { .driver = { .name = "aoa_codec_tas", .owner = THIS_MODULE, }, - .attach_adapter = tas_i2c_attach, .probe = tas_i2c_probe, .remove = tas_i2c_remove, .id_table = tas_i2c_id, diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index afef72c4f0d3..0d7b25e81643 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -108,7 +108,7 @@ static struct pxa2xx_pcm_client pxa2xx_ac97_pcm_client = { #ifdef CONFIG_PM -static int pxa2xx_ac97_do_suspend(struct snd_card *card, pm_message_t state) +static int pxa2xx_ac97_do_suspend(struct snd_card *card) { pxa2xx_audio_ops_t *platform_ops = card->dev->platform_data; @@ -144,7 +144,7 @@ static int pxa2xx_ac97_suspend(struct device *dev) int ret = 0; if (card) - ret = pxa2xx_ac97_do_suspend(card, PMSG_SUSPEND); + ret = pxa2xx_ac97_do_suspend(card); return ret; } @@ -160,10 +160,7 @@ static int pxa2xx_ac97_resume(struct device *dev) return ret; } -static const struct dev_pm_ops pxa2xx_ac97_pm_ops = { - .suspend = pxa2xx_ac97_suspend, - .resume = pxa2xx_ac97_resume, -}; +static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, pxa2xx_ac97_suspend, pxa2xx_ac97_resume); #endif static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index f7c2bb08055d..eb4ceb71123e 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -535,9 +535,9 @@ out_put_pclk: } #ifdef CONFIG_PM -static int atmel_abdac_suspend(struct platform_device *pdev, pm_message_t msg) +static int atmel_abdac_suspend(struct device *pdev) { - struct snd_card *card = platform_get_drvdata(pdev); + struct snd_card *card = dev_get_drvdata(pdev); struct atmel_abdac *dac = card->private_data; dw_dma_cyclic_stop(dac->dma.chan); @@ -547,9 +547,9 @@ static int atmel_abdac_suspend(struct platform_device *pdev, pm_message_t msg) return 0; } -static int atmel_abdac_resume(struct platform_device *pdev) +static int atmel_abdac_resume(struct device *pdev) { - struct snd_card *card = platform_get_drvdata(pdev); + struct snd_card *card = dev_get_drvdata(pdev); struct atmel_abdac *dac = card->private_data; clk_enable(dac->pclk); @@ -559,9 +559,11 @@ static int atmel_abdac_resume(struct platform_device *pdev) return 0; } + +static SIMPLE_DEV_PM_OPS(atmel_abdac_pm, atmel_abdac_suspend, atmel_abdac_resume); +#define ATMEL_ABDAC_PM_OPS &atmel_abdac_pm #else -#define atmel_abdac_suspend NULL -#define atmel_abdac_resume NULL +#define ATMEL_ABDAC_PM_OPS NULL #endif static int __devexit atmel_abdac_remove(struct platform_device *pdev) @@ -589,9 +591,9 @@ static struct platform_driver atmel_abdac_driver = { .remove = __devexit_p(atmel_abdac_remove), .driver = { .name = "atmel_abdac", + .owner = THIS_MODULE, + .pm = ATMEL_ABDAC_PM_OPS, }, - .suspend = atmel_abdac_suspend, - .resume = atmel_abdac_resume, }; static int __init atmel_abdac_init(void) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 115313ef54d6..bf47025bdf45 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -991,6 +991,8 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) gpio_direction_output(pdata->reset_pin, 1); chip->reset_pin = pdata->reset_pin; } + } else { + chip->reset_pin = -EINVAL; } snd_card_set_dev(card, &pdev->dev); @@ -1133,9 +1135,9 @@ err_snd_card_new: } #ifdef CONFIG_PM -static int atmel_ac97c_suspend(struct platform_device *pdev, pm_message_t msg) +static int atmel_ac97c_suspend(struct device *pdev) { - struct snd_card *card = platform_get_drvdata(pdev); + struct snd_card *card = dev_get_drvdata(pdev); struct atmel_ac97c *chip = card->private_data; if (cpu_is_at32ap7000()) { @@ -1149,9 +1151,9 @@ static int atmel_ac97c_suspend(struct platform_device *pdev, pm_message_t msg) return 0; } -static int atmel_ac97c_resume(struct platform_device *pdev) +static int atmel_ac97c_resume(struct device *pdev) { - struct snd_card *card = platform_get_drvdata(pdev); + struct snd_card *card = dev_get_drvdata(pdev); struct atmel_ac97c *chip = card->private_data; clk_enable(chip->pclk); @@ -1163,9 +1165,11 @@ static int atmel_ac97c_resume(struct platform_device *pdev) } return 0; } + +static SIMPLE_DEV_PM_OPS(atmel_ac97c_pm, atmel_ac97c_suspend, atmel_ac97c_resume); +#define ATMEL_AC97C_PM_OPS &atmel_ac97c_pm #else -#define atmel_ac97c_suspend NULL -#define atmel_ac97c_resume NULL +#define ATMEL_AC97C_PM_OPS NULL #endif static int __devexit atmel_ac97c_remove(struct platform_device *pdev) @@ -1208,9 +1212,9 @@ static struct platform_driver atmel_ac97c_driver = { .remove = __devexit_p(atmel_ac97c_remove), .driver = { .name = "atmel_ac97c", + .owner = THIS_MODULE, + .pm = ATMEL_AC97C_PM_OPS, }, - .suspend = atmel_ac97c_suspend, - .resume = atmel_ac97c_resume, }; static int __init atmel_ac97c_init(void) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index a68aed7fce02..ec2118d0e27a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -502,10 +502,8 @@ static int snd_compr_pause(struct snd_compr_stream *stream) if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) return -EPERM; retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); - if (!retval) { + if (!retval) stream->runtime->state = SNDRV_PCM_STATE_PAUSED; - wake_up(&stream->runtime->sleep); - } return retval; } @@ -544,6 +542,10 @@ static int snd_compr_stop(struct snd_compr_stream *stream) if (!retval) { stream->runtime->state = SNDRV_PCM_STATE_SETUP; wake_up(&stream->runtime->sleep); + stream->runtime->hw_pointer = 0; + stream->runtime->app_pointer = 0; + stream->runtime->total_bytes_available = 0; + stream->runtime->total_bytes_transferred = 0; } return retval; } diff --git a/sound/core/jack.c b/sound/core/jack.c index 471e1e3b0a99..a06b1651fcba 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -155,7 +155,7 @@ EXPORT_SYMBOL(snd_jack_new); * @jack: The jack to configure * @parent: The device to set as parent for the jack. * - * Set the parent for the jack input device in the device tree. This + * Set the parent for the jack devices in the device tree. This * function is only valid prior to registration of the jack. If no * parent is configured then the parent device will be the sound card. */ @@ -179,6 +179,9 @@ EXPORT_SYMBOL(snd_jack_set_parent); * mapping is provided but keys are enabled in the jack type then * BTN_n numeric buttons will be reported. * + * If jacks are not reporting via the input API this call will have no + * effect. + * * Note that this is intended to be use by simple devices with small * numbers of keys that can be reported. It is also possible to * access the input device directly - devices with complex input diff --git a/sound/core/misc.c b/sound/core/misc.c index 768167925409..30e027ecf4da 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -68,6 +68,7 @@ void __snd_printk(unsigned int level, const char *path, int line, { va_list args; #ifdef CONFIG_SND_VERBOSE_PRINTK + int kern_level; struct va_format vaf; char verbose_fmt[] = KERN_DEFAULT "ALSA %s:%d %pV"; #endif @@ -81,12 +82,16 @@ void __snd_printk(unsigned int level, const char *path, int line, #ifdef CONFIG_SND_VERBOSE_PRINTK vaf.fmt = format; vaf.va = &args; - if (format[0] == '<' && format[2] == '>') { - memcpy(verbose_fmt, format, 3); - vaf.fmt = format + 3; + + kern_level = printk_get_level(format); + if (kern_level) { + const char *end_of_header = printk_skip_level(format); + memcpy(verbose_fmt, format, end_of_header - format); + vaf.fmt = end_of_header; } else if (level) - memcpy(verbose_fmt, KERN_DEBUG, 3); + memcpy(verbose_fmt, KERN_DEBUG, sizeof(KERN_DEBUG) - 1); printk(verbose_fmt, sanity_file_name(path), line, &vaf); + #else vprintk(format, args); #endif diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 4d18941178e6..7ae671923393 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -313,9 +313,22 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_base; snd_pcm_sframes_t hdelta, delta; unsigned long jdelta; + unsigned long curr_jiffies; + struct timespec curr_tstamp; old_hw_ptr = runtime->status->hw_ptr; + + /* + * group pointer, time and jiffies reads to allow for more + * accurate correlations/corrections. + * The values are stored at the end of this routine after + * corrections for hw_ptr position + */ pos = substream->ops->pointer(substream); + curr_jiffies = jiffies; + if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) + snd_pcm_gettime(runtime, (struct timespec *)&curr_tstamp); + if (pos == SNDRV_PCM_POS_XRUN) { xrun(substream); return -EPIPE; @@ -343,7 +356,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, delta = runtime->hw_ptr_interrupt + runtime->period_size; if (delta > new_hw_ptr) { /* check for double acknowledged interrupts */ - hdelta = jiffies - runtime->hw_ptr_jiffies; + hdelta = curr_jiffies - runtime->hw_ptr_jiffies; if (hdelta > runtime->hw_ptr_buffer_jiffies/2) { hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) @@ -388,7 +401,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, * Without regular period interrupts, we have to check * the elapsed time to detect xruns. */ - jdelta = jiffies - runtime->hw_ptr_jiffies; + jdelta = curr_jiffies - runtime->hw_ptr_jiffies; if (jdelta < runtime->hw_ptr_buffer_jiffies / 2) goto no_delta_check; hdelta = jdelta - delta * HZ / runtime->rate; @@ -430,7 +443,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (hdelta < runtime->delay) goto no_jiffies_check; hdelta -= runtime->delay; - jdelta = jiffies - runtime->hw_ptr_jiffies; + jdelta = curr_jiffies - runtime->hw_ptr_jiffies; if (((hdelta * HZ) / runtime->rate) > jdelta + HZ/100) { delta = jdelta / (((runtime->period_size * HZ) / runtime->rate) @@ -492,9 +505,9 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, } runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; - runtime->hw_ptr_jiffies = jiffies; + runtime->hw_ptr_jiffies = curr_jiffies; if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) - snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); + runtime->status->tstamp = curr_tstamp; return snd_pcm_update_state(substream, runtime); } @@ -1237,10 +1250,10 @@ static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params, int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_list *l) + const struct snd_pcm_hw_constraint_list *l) { return snd_pcm_hw_rule_add(runtime, cond, var, - snd_pcm_hw_rule_list, l, + snd_pcm_hw_rule_list, (void *)l, var, -1); } @@ -1894,6 +1907,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t xfer = 0; snd_pcm_uframes_t offset = 0; + snd_pcm_uframes_t avail; int err = 0; if (size == 0) @@ -1917,13 +1931,12 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, } runtime->twake = runtime->control->avail_min ? : 1; + if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) + snd_pcm_update_hw_ptr(substream); + avail = snd_pcm_playback_avail(runtime); while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; - snd_pcm_uframes_t avail; snd_pcm_uframes_t cont; - if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) - snd_pcm_update_hw_ptr(substream); - avail = snd_pcm_playback_avail(runtime); if (!avail) { if (nonblock) { err = -EAGAIN; @@ -1971,6 +1984,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, offset += frames; size -= frames; xfer += frames; + avail -= frames; if (runtime->status->state == SNDRV_PCM_STATE_PREPARED && snd_pcm_playback_hw_avail(runtime) >= (snd_pcm_sframes_t)runtime->start_threshold) { err = snd_pcm_start(substream); @@ -2111,6 +2125,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t xfer = 0; snd_pcm_uframes_t offset = 0; + snd_pcm_uframes_t avail; int err = 0; if (size == 0) @@ -2141,13 +2156,12 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, } runtime->twake = runtime->control->avail_min ? : 1; + if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) + snd_pcm_update_hw_ptr(substream); + avail = snd_pcm_capture_avail(runtime); while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; - snd_pcm_uframes_t avail; snd_pcm_uframes_t cont; - if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) - snd_pcm_update_hw_ptr(substream); - avail = snd_pcm_capture_avail(runtime); if (!avail) { if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) { @@ -2202,6 +2216,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, offset += frames; size -= frames; xfer += frames; + avail -= frames; } _end_unlock: runtime->twake = 0; diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 9c9eff9afbac..d4fc1bfbe457 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -488,3 +488,21 @@ unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate) return SNDRV_PCM_RATE_KNOT; } EXPORT_SYMBOL(snd_pcm_rate_to_rate_bit); + +/** + * snd_pcm_rate_bit_to_rate - converts SNDRV_PCM_RATE_xxx bit to sample rate + * @rate_bit: the rate bit to convert + * + * Returns the sample rate that corresponds to the given SNDRV_PCM_RATE_xxx flag + * or 0 for an unknown rate bit + */ +unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit) +{ + unsigned int i; + + for (i = 0; i < snd_pcm_known_rates.count; i++) + if ((1u << i) == rate_bit) + return snd_pcm_known_rates.list[i]; + return 0; +} +EXPORT_SYMBOL(snd_pcm_rate_bit_to_rate); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 3fe99e644eb8..53b5ada8f7c3 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1360,7 +1360,14 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream, static int snd_pcm_pre_drain_init(struct snd_pcm_substream *substream, int state) { - substream->runtime->trigger_master = substream; + struct snd_pcm_runtime *runtime = substream->runtime; + switch (runtime->status->state) { + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_DISCONNECTED: + case SNDRV_PCM_STATE_SUSPENDED: + return -EBADFD; + } + runtime->trigger_master = substream; return 0; } @@ -1379,6 +1386,9 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state) case SNDRV_PCM_STATE_RUNNING: runtime->status->state = SNDRV_PCM_STATE_DRAINING; break; + case SNDRV_PCM_STATE_XRUN: + runtime->status->state = SNDRV_PCM_STATE_SETUP; + break; default: break; } diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index c70092043061..e9528333e36d 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -35,7 +35,7 @@ #include <linux/sound.h> #include <linux/mutex.h> -#define SNDRV_OSS_MINORS 128 +#define SNDRV_OSS_MINORS 256 static struct snd_minor *snd_oss_minors[SNDRV_OSS_MINORS]; static DEFINE_MUTEX(sound_oss_mutex); @@ -111,7 +111,7 @@ int snd_register_oss_device(int type, struct snd_card *card, int dev, int register1 = -1, register2 = -1; struct device *carddev = snd_card_get_device_link(card); - if (card && card->number >= 8) + if (card && card->number >= SNDRV_MINOR_OSS_DEVICES) return 0; /* ignore silently */ if (minor < 0) return minor; @@ -170,7 +170,7 @@ int snd_unregister_oss_device(int type, struct snd_card *card, int dev) int track2 = -1; struct snd_minor *mptr; - if (card && card->number >= 8) + if (card && card->number >= SNDRV_MINOR_OSS_DEVICES) return 0; if (minor < 0) return minor; diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index ad079b63b8ba..1128b35b2b05 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -117,6 +117,7 @@ struct loopback_pcm { /* timer stuff */ unsigned int irq_pos; /* fractional IRQ position */ unsigned int period_size_frac; + unsigned int last_drift; unsigned long last_jiffies; struct timer_list timer; }; @@ -264,6 +265,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) return err; dpcm->last_jiffies = jiffies; dpcm->pcm_rate_shift = 0; + dpcm->last_drift = 0; spin_lock(&cable->lock); cable->running |= stream; cable->pause &= ~stream; @@ -444,34 +446,30 @@ static void copy_play_buf(struct loopback_pcm *play, } } -#define BYTEPOS_UPDATE_POSONLY 0 -#define BYTEPOS_UPDATE_CLEAR 1 -#define BYTEPOS_UPDATE_COPY 2 - -static void loopback_bytepos_update(struct loopback_pcm *dpcm, - unsigned int delta, - unsigned int cmd) +static inline unsigned int bytepos_delta(struct loopback_pcm *dpcm, + unsigned int jiffies_delta) { - unsigned int count; unsigned long last_pos; + unsigned int delta; last_pos = byte_pos(dpcm, dpcm->irq_pos); - dpcm->irq_pos += delta * dpcm->pcm_bps; - count = byte_pos(dpcm, dpcm->irq_pos) - last_pos; - if (!count) - return; - if (cmd == BYTEPOS_UPDATE_CLEAR) - clear_capture_buf(dpcm, count); - else if (cmd == BYTEPOS_UPDATE_COPY) - copy_play_buf(dpcm->cable->streams[SNDRV_PCM_STREAM_PLAYBACK], - dpcm->cable->streams[SNDRV_PCM_STREAM_CAPTURE], - count); - dpcm->buf_pos += count; - dpcm->buf_pos %= dpcm->pcm_buffer_size; + dpcm->irq_pos += jiffies_delta * dpcm->pcm_bps; + delta = byte_pos(dpcm, dpcm->irq_pos) - last_pos; + if (delta >= dpcm->last_drift) + delta -= dpcm->last_drift; + dpcm->last_drift = 0; if (dpcm->irq_pos >= dpcm->period_size_frac) { dpcm->irq_pos %= dpcm->period_size_frac; dpcm->period_update_pending = 1; } + return delta; +} + +static inline void bytepos_finish(struct loopback_pcm *dpcm, + unsigned int delta) +{ + dpcm->buf_pos += delta; + dpcm->buf_pos %= dpcm->pcm_buffer_size; } static unsigned int loopback_pos_update(struct loopback_cable *cable) @@ -481,7 +479,7 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable) struct loopback_pcm *dpcm_capt = cable->streams[SNDRV_PCM_STREAM_CAPTURE]; unsigned long delta_play = 0, delta_capt = 0; - unsigned int running; + unsigned int running, count1, count2; unsigned long flags; spin_lock_irqsave(&cable->lock, flags); @@ -500,12 +498,13 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable) goto unlock; if (delta_play > delta_capt) { - loopback_bytepos_update(dpcm_play, delta_play - delta_capt, - BYTEPOS_UPDATE_POSONLY); + count1 = bytepos_delta(dpcm_play, delta_play - delta_capt); + bytepos_finish(dpcm_play, count1); delta_play = delta_capt; } else if (delta_play < delta_capt) { - loopback_bytepos_update(dpcm_capt, delta_capt - delta_play, - BYTEPOS_UPDATE_CLEAR); + count1 = bytepos_delta(dpcm_capt, delta_capt - delta_play); + clear_capture_buf(dpcm_capt, count1); + bytepos_finish(dpcm_capt, count1); delta_capt = delta_play; } @@ -513,8 +512,17 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable) goto unlock; /* note delta_capt == delta_play at this moment */ - loopback_bytepos_update(dpcm_capt, delta_capt, BYTEPOS_UPDATE_COPY); - loopback_bytepos_update(dpcm_play, delta_play, BYTEPOS_UPDATE_POSONLY); + count1 = bytepos_delta(dpcm_play, delta_play); + count2 = bytepos_delta(dpcm_capt, delta_capt); + if (count1 < count2) { + dpcm_capt->last_drift = count2 - count1; + count1 = count2; + } else if (count1 > count2) { + dpcm_play->last_drift = count1 - count2; + } + copy_play_buf(dpcm_play, dpcm_capt, count1); + bytepos_finish(dpcm_play, count1); + bytepos_finish(dpcm_capt, count1); unlock: spin_unlock_irqrestore(&cable->lock, flags); return running; @@ -1169,10 +1177,9 @@ static int __devexit loopback_remove(struct platform_device *devptr) } #ifdef CONFIG_PM -static int loopback_suspend(struct platform_device *pdev, - pm_message_t state) +static int loopback_suspend(struct device *pdev) { - struct snd_card *card = platform_get_drvdata(pdev); + struct snd_card *card = dev_get_drvdata(pdev); struct loopback *loopback = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -1182,13 +1189,18 @@ static int loopback_suspend(struct platform_device *pdev, return 0; } -static int loopback_resume(struct platform_device *pdev) +static int loopback_resume(struct device *pdev) { - struct snd_card *card = platform_get_drvdata(pdev); + struct snd_card *card = dev_get_drvdata(pdev); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(loopback_pm, loopback_suspend, loopback_resume); +#define LOOPBACK_PM_OPS &loopback_pm +#else +#define LOOPBACK_PM_OPS NULL #endif #define SND_LOOPBACK_DRIVER "snd_aloop" @@ -1196,12 +1208,10 @@ static int loopback_resume(struct platform_device *pdev) static struct platform_driver loopback_driver = { .probe = loopback_probe, .remove = __devexit_p(loopback_remove), -#ifdef CONFIG_PM - .suspend = loopback_suspend, - .resume = loopback_resume, -#endif .driver = { - .name = SND_LOOPBACK_DRIVER + .name = SND_LOOPBACK_DRIVER, + .owner = THIS_MODULE, + .pm = LOOPBACK_PM_OPS, }, }; diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index ad9434fd6370..f7d3bfc6bca8 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -1065,9 +1065,9 @@ static int __devexit snd_dummy_remove(struct platform_device *devptr) } #ifdef CONFIG_PM -static int snd_dummy_suspend(struct platform_device *pdev, pm_message_t state) +static int snd_dummy_suspend(struct device *pdev) { - struct snd_card *card = platform_get_drvdata(pdev); + struct snd_card *card = dev_get_drvdata(pdev); struct snd_dummy *dummy = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -1075,13 +1075,18 @@ static int snd_dummy_suspend(struct platform_device *pdev, pm_message_t state) return 0; } -static int snd_dummy_resume(struct platform_device *pdev) +static int snd_dummy_resume(struct device *pdev) { - struct snd_card *card = platform_get_drvdata(pdev); + struct snd_card *card = dev_get_drvdata(pdev); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(snd_dummy_pm, snd_dummy_suspend, snd_dummy_resume); +#define SND_DUMMY_PM_OPS &snd_dummy_pm +#else +#define SND_DUMMY_PM_OPS NULL #endif #define SND_DUMMY_DRIVER "snd_dummy" @@ -1089,12 +1094,10 @@ static int snd_dummy_resume(struct platform_device *pdev) static struct platform_driver snd_dummy_driver = { .probe = snd_dummy_probe, .remove = __devexit_p(snd_dummy_remove), -#ifdef CONFIG_PM - .suspend = snd_dummy_suspend, - .resume = snd_dummy_resume, -#endif .driver = { - .name = SND_DUMMY_DRIVER + .name = SND_DUMMY_DRIVER, + .owner = THIS_MODULE, + .pm = SND_DUMMY_PM_OPS, }, }; diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index 86f5fbc2da72..bc03a2046c9c 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -139,7 +139,8 @@ static struct platform_driver snd_mpu401_driver = { .probe = snd_mpu401_probe, .remove = __devexit_p(snd_mpu401_remove), .driver = { - .name = SND_MPU401_DRIVER + .name = SND_MPU401_DRIVER, + .owner = THIS_MODULE, }, }; diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index 76930793fb69..cad73af3860c 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -759,7 +759,8 @@ static struct platform_driver snd_mtpav_driver = { .probe = snd_mtpav_probe, .remove = __devexit_p(snd_mtpav_remove), .driver = { - .name = SND_MTPAV_DRIVER + .name = SND_MTPAV_DRIVER, + .owner = THIS_MODULE, }, }; diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index 621e60e2029f..2d5514b0a290 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -1040,7 +1040,8 @@ static struct platform_driver snd_mts64_driver = { .probe = snd_mts64_probe, .remove = __devexit_p(snd_mts64_remove), .driver = { - .name = PLATFORM_DRIVER + .name = PLATFORM_DRIVER, + .owner = THIS_MODULE, } }; diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 99704e6a2e26..6ca59fc6dcb9 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -200,15 +200,18 @@ static void pcsp_stop_beep(struct snd_pcsp *chip) } #ifdef CONFIG_PM -static int pcsp_suspend(struct platform_device *dev, pm_message_t state) +static int pcsp_suspend(struct device *dev) { - struct snd_pcsp *chip = platform_get_drvdata(dev); + struct snd_pcsp *chip = dev_get_drvdata(dev); pcsp_stop_beep(chip); snd_pcm_suspend_all(chip->pcm); return 0; } + +static SIMPLE_DEV_PM_OPS(pcsp_pm, pcsp_suspend, NULL); +#define PCSP_PM_OPS &pcsp_pm #else -#define pcsp_suspend NULL +#define PCSP_PM_OPS NULL #endif /* CONFIG_PM */ static void pcsp_shutdown(struct platform_device *dev) @@ -221,10 +224,10 @@ static struct platform_driver pcsp_platform_driver = { .driver = { .name = "pcspkr", .owner = THIS_MODULE, + .pm = PCSP_PM_OPS, }, .probe = pcsp_probe, .remove = __devexit_p(pcsp_remove), - .suspend = pcsp_suspend, .shutdown = pcsp_shutdown, }; diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index 3e32bd3d95d9..8364855ed14f 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -829,7 +829,8 @@ static struct platform_driver snd_portman_driver = { .probe = snd_portman_probe, .remove = __devexit_p(snd_portman_remove), .driver = { - .name = PLATFORM_DRIVER + .name = PLATFORM_DRIVER, + .owner = THIS_MODULE, } }; diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index b2d0e8e49bed..86700671d1ac 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -995,7 +995,8 @@ static struct platform_driver snd_serial_driver = { .probe = snd_serial_probe, .remove = __devexit_p( snd_serial_remove), .driver = { - .name = SND_SERIAL_DRIVER + .name = SND_SERIAL_DRIVER, + .owner = THIS_MODULE, }, }; diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index 9d97478a18b3..d7d514df9058 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -142,7 +142,8 @@ static struct platform_driver snd_virmidi_driver = { .probe = snd_virmidi_probe, .remove = __devexit_p(snd_virmidi_remove), .driver = { - .name = SND_VIRMIDI_DRIVER + .name = SND_VIRMIDI_DRIVER, + .owner = THIS_MODULE, }, }; diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index b8e515999bc2..de5055a3b0d0 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -725,7 +725,7 @@ EXPORT_SYMBOL(snd_vx_dsp_load); /* * suspend */ -int snd_vx_suspend(struct vx_core *chip, pm_message_t state) +int snd_vx_suspend(struct vx_core *chip) { unsigned int i; diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 87657dd7714c..ea995af6d049 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -31,6 +31,8 @@ #define INTERRUPT_INTERVAL 16 #define QUEUE_LENGTH 48 +static void pcm_period_tasklet(unsigned long data); + /** * amdtp_out_stream_init - initialize an AMDTP output stream structure * @s: the AMDTP output stream to initialize @@ -47,6 +49,7 @@ int amdtp_out_stream_init(struct amdtp_out_stream *s, struct fw_unit *unit, s->flags = flags; s->context = ERR_PTR(-1); mutex_init(&s->mutex); + tasklet_init(&s->period_tasklet, pcm_period_tasklet, (unsigned long)s); s->packet_index = 0; return 0; @@ -164,6 +167,21 @@ void amdtp_out_stream_set_pcm_format(struct amdtp_out_stream *s, } EXPORT_SYMBOL(amdtp_out_stream_set_pcm_format); +/** + * amdtp_out_stream_pcm_prepare - prepare PCM device for running + * @s: the AMDTP output stream + * + * This function should be called from the PCM device's .prepare callback. + */ +void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s) +{ + tasklet_kill(&s->period_tasklet); + s->pcm_buffer_pointer = 0; + s->pcm_period_pointer = 0; + s->pointer_flush = true; +} +EXPORT_SYMBOL(amdtp_out_stream_pcm_prepare); + static unsigned int calculate_data_blocks(struct amdtp_out_stream *s) { unsigned int phase, data_blocks; @@ -376,11 +394,21 @@ static void queue_out_packet(struct amdtp_out_stream *s, unsigned int cycle) s->pcm_period_pointer += data_blocks; if (s->pcm_period_pointer >= pcm->runtime->period_size) { s->pcm_period_pointer -= pcm->runtime->period_size; - snd_pcm_period_elapsed(pcm); + s->pointer_flush = false; + tasklet_hi_schedule(&s->period_tasklet); } } } +static void pcm_period_tasklet(unsigned long data) +{ + struct amdtp_out_stream *s = (void *)data; + struct snd_pcm_substream *pcm = ACCESS_ONCE(s->pcm); + + if (pcm) + snd_pcm_period_elapsed(pcm); +} + static void out_packet_callback(struct fw_iso_context *context, u32 cycle, size_t header_length, void *header, void *data) { @@ -506,6 +534,24 @@ err_unlock: EXPORT_SYMBOL(amdtp_out_stream_start); /** + * amdtp_out_stream_pcm_pointer - get the PCM buffer position + * @s: the AMDTP output stream that transports the PCM data + * + * Returns the current buffer position, in frames. + */ +unsigned long amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s) +{ + /* this optimization is allowed to be racy */ + if (s->pointer_flush) + fw_iso_context_flush_completions(s->context); + else + s->pointer_flush = true; + + return ACCESS_ONCE(s->pcm_buffer_pointer); +} +EXPORT_SYMBOL(amdtp_out_stream_pcm_pointer); + +/** * amdtp_out_stream_update - update the stream after a bus reset * @s: the AMDTP output stream */ @@ -532,6 +578,7 @@ void amdtp_out_stream_stop(struct amdtp_out_stream *s) return; } + tasklet_kill(&s->period_tasklet); fw_iso_context_stop(s->context); fw_iso_context_destroy(s->context); s->context = ERR_PTR(-1); diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index 537a9cb83581..b680c5ef01d6 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -1,6 +1,7 @@ #ifndef SOUND_FIREWIRE_AMDTP_H_INCLUDED #define SOUND_FIREWIRE_AMDTP_H_INCLUDED +#include <linux/interrupt.h> #include <linux/mutex.h> #include <linux/spinlock.h> #include "packets-buffer.h" @@ -55,6 +56,7 @@ struct amdtp_out_stream { struct iso_packets_buffer buffer; struct snd_pcm_substream *pcm; + struct tasklet_struct period_tasklet; int packet_index; unsigned int data_block_counter; @@ -66,6 +68,7 @@ struct amdtp_out_stream { unsigned int pcm_buffer_pointer; unsigned int pcm_period_pointer; + bool pointer_flush; }; int amdtp_out_stream_init(struct amdtp_out_stream *s, struct fw_unit *unit, @@ -81,6 +84,8 @@ void amdtp_out_stream_stop(struct amdtp_out_stream *s); void amdtp_out_stream_set_pcm_format(struct amdtp_out_stream *s, snd_pcm_format_t format); +void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s); +unsigned long amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s); void amdtp_out_stream_pcm_abort(struct amdtp_out_stream *s); /** @@ -123,18 +128,6 @@ static inline bool amdtp_out_streaming_error(struct amdtp_out_stream *s) } /** - * amdtp_out_stream_pcm_prepare - prepare PCM device for running - * @s: the AMDTP output stream - * - * This function should be called from the PCM device's .prepare callback. - */ -static inline void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s) -{ - s->pcm_buffer_pointer = 0; - s->pcm_period_pointer = 0; -} - -/** * amdtp_out_stream_pcm_trigger - start/stop playback from a PCM device * @s: the AMDTP output stream * @pcm: the PCM device to be started, or %NULL to stop the current device @@ -149,18 +142,6 @@ static inline void amdtp_out_stream_pcm_trigger(struct amdtp_out_stream *s, ACCESS_ONCE(s->pcm) = pcm; } -/** - * amdtp_out_stream_pcm_pointer - get the PCM buffer position - * @s: the AMDTP output stream that transports the PCM data - * - * Returns the current buffer position, in frames. - */ -static inline unsigned long -amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s) -{ - return ACCESS_ONCE(s->pcm_buffer_pointer); -} - static inline bool cip_sfc_is_base_44100(enum cip_sfc sfc) { return sfc & 1; diff --git a/sound/firewire/cmp.c b/sound/firewire/cmp.c index 76294f2ae47f..645cb0ba4293 100644 --- a/sound/firewire/cmp.c +++ b/sound/firewire/cmp.c @@ -84,7 +84,7 @@ static int pcr_modify(struct cmp_connection *c, return 0; io_error: - cmp_error(c, "transaction failed: %s\n", rcode_string(rcode)); + cmp_error(c, "transaction failed: %s\n", fw_rcode_string(rcode)); return -EIO; bus_reset: diff --git a/sound/firewire/lib.c b/sound/firewire/lib.c index 4750cea2210e..14eb41498372 100644 --- a/sound/firewire/lib.c +++ b/sound/firewire/lib.c @@ -14,32 +14,6 @@ #define ERROR_RETRY_DELAY_MS 5 /** - * rcode_string - convert a firewire result code to a string - * @rcode: the result - */ -const char *rcode_string(unsigned int rcode) -{ - static const char *const names[] = { - [RCODE_COMPLETE] = "complete", - [RCODE_CONFLICT_ERROR] = "conflict error", - [RCODE_DATA_ERROR] = "data error", - [RCODE_TYPE_ERROR] = "type error", - [RCODE_ADDRESS_ERROR] = "address error", - [RCODE_SEND_ERROR] = "send error", - [RCODE_CANCELLED] = "cancelled", - [RCODE_BUSY] = "busy", - [RCODE_GENERATION] = "generation", - [RCODE_NO_ACK] = "no ack", - }; - - if (rcode < ARRAY_SIZE(names) && names[rcode]) - return names[rcode]; - else - return "unknown"; -} -EXPORT_SYMBOL(rcode_string); - -/** * snd_fw_transaction - send a request and wait for its completion * @unit: the driver's unit on the target device * @tcode: the transaction code @@ -71,7 +45,7 @@ int snd_fw_transaction(struct fw_unit *unit, int tcode, if (rcode_is_permanent_error(rcode) || ++tries >= 3) { dev_err(&unit->device, "transaction failed: %s\n", - rcode_string(rcode)); + fw_rcode_string(rcode)); return -EIO; } diff --git a/sound/firewire/lib.h b/sound/firewire/lib.h index 064f3fd9ab06..aef301476ea9 100644 --- a/sound/firewire/lib.h +++ b/sound/firewire/lib.h @@ -8,7 +8,6 @@ struct fw_unit; int snd_fw_transaction(struct fw_unit *unit, int tcode, u64 offset, void *buffer, size_t length); -const char *rcode_string(unsigned int rcode); /* returns true if retrying the transaction would not make sense */ static inline bool rcode_is_permanent_error(int rcode) diff --git a/sound/isa/als100.c b/sound/isa/als100.c index d1f4351fb6ee..2d67c78c9f4b 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -7,7 +7,7 @@ Thanks to Pierfrancesco 'qM2' Passerini. Generalised for soundcards based on DT-0196 and ALS-007 chips - by Jonathan Woithe <jwoithe@physics.adelaide.edu.au>: June 2002. + by Jonathan Woithe <jwoithe@just42.net>: June 2002. This program is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index d7ccf28bd66a..f8fbe22515c9 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -135,10 +135,9 @@ struct snd_opti9xx { unsigned long mc_base_size; #ifdef OPTi93X unsigned long mc_indir_index; - unsigned long mc_indir_size; struct resource *res_mc_indir; - struct snd_wss *codec; #endif /* OPTi93X */ + struct snd_wss *codec; unsigned long pwd_reg; spinlock_t lock; @@ -245,10 +244,8 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, case OPTi9XX_HW_82C931: case OPTi9XX_HW_82C933: chip->mc_base = (hardware == OPTi9XX_HW_82C930) ? 0xf8f : 0xf8d; - if (!chip->mc_indir_index) { + if (!chip->mc_indir_index) chip->mc_indir_index = 0xe0e; - chip->mc_indir_size = 2; - } chip->password = 0xe4; chip->pwd_reg = 0; break; @@ -351,7 +348,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask))) -static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, +static int snd_opti9xx_configure(struct snd_opti9xx *chip, long port, int irq, int dma1, int dma2, long mpu_port, int mpu_irq) @@ -403,7 +400,9 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, #else /* OPTi93X */ case OPTi9XX_HW_82C931: - case OPTi9XX_HW_82C933: + /* disable 3D sound (set GPIO1 as output, low) */ + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(20), 0x04, 0x0c); + case OPTi9XX_HW_82C933: /* FALL THROUGH */ /* * The BTC 1817DW has QS1000 wavetable which is connected * to the serial digital input of the OPTI931. @@ -696,8 +695,7 @@ static int __devinit snd_opti9xx_read_check(struct snd_opti9xx *chip) if (value == snd_opti9xx_read(chip, OPTi9XX_MC_REG(1))) return 0; #else /* OPTi93X */ - chip->res_mc_indir = request_region(chip->mc_indir_index, - chip->mc_indir_size, + chip->res_mc_indir = request_region(chip->mc_indir_index, 2, "OPTi93x MC"); if (chip->res_mc_indir == NULL) return -EBUSY; @@ -770,8 +768,9 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, #ifdef OPTi93X port = pnp_port_start(pdev, 0) - 4; fm_port = pnp_port_start(pdev, 1) + 8; - chip->mc_indir_index = pnp_port_start(pdev, 3) + 2; - chip->mc_indir_size = pnp_port_len(pdev, 3) - 2; + /* adjust mc_indir_index - some cards report it at 0xe?d, + other at 0xe?c but it really is always at 0xe?e */ + chip->mc_indir_index = (pnp_port_start(pdev, 3) & ~0xf) | 0xe; #else devmc = pnp_request_card_device(card, pid->devs[2].id, NULL); if (devmc == NULL) @@ -871,9 +870,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) &codec); if (error < 0) return error; -#ifdef OPTi93X chip->codec = codec; -#endif error = snd_wss_pcm(codec, 0, &pcm); if (error < 0) return error; @@ -1054,11 +1051,55 @@ static int __devexit snd_opti9xx_isa_remove(struct device *devptr, return 0; } +#ifdef CONFIG_PM +static int snd_opti9xx_suspend(struct snd_card *card) +{ + struct snd_opti9xx *chip = card->private_data; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + chip->codec->suspend(chip->codec); + return 0; +} + +static int snd_opti9xx_resume(struct snd_card *card) +{ + struct snd_opti9xx *chip = card->private_data; + int error, xdma2; +#if defined(CS4231) || defined(OPTi93X) + xdma2 = dma2; +#else + xdma2 = -1; +#endif + + error = snd_opti9xx_configure(chip, port, irq, dma1, xdma2, + mpu_port, mpu_irq); + if (error) + return error; + chip->codec->resume(chip->codec); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} + +static int snd_opti9xx_isa_suspend(struct device *dev, unsigned int n, + pm_message_t state) +{ + return snd_opti9xx_suspend(dev_get_drvdata(dev)); +} + +static int snd_opti9xx_isa_resume(struct device *dev, unsigned int n) +{ + return snd_opti9xx_resume(dev_get_drvdata(dev)); +} +#endif + static struct isa_driver snd_opti9xx_driver = { .match = snd_opti9xx_isa_match, .probe = snd_opti9xx_isa_probe, .remove = __devexit_p(snd_opti9xx_isa_remove), - /* FIXME: suspend/resume */ +#ifdef CONFIG_PM + .suspend = snd_opti9xx_isa_suspend, + .resume = snd_opti9xx_isa_resume, +#endif .driver = { .name = DEV_NAME }, @@ -1124,12 +1165,29 @@ static void __devexit snd_opti9xx_pnp_remove(struct pnp_card_link * pcard) snd_opti9xx_pnp_is_probed = 0; } +#ifdef CONFIG_PM +static int snd_opti9xx_pnp_suspend(struct pnp_card_link *pcard, + pm_message_t state) +{ + return snd_opti9xx_suspend(pnp_get_card_drvdata(pcard)); +} + +static int snd_opti9xx_pnp_resume(struct pnp_card_link *pcard) +{ + return snd_opti9xx_resume(pnp_get_card_drvdata(pcard)); +} +#endif + static struct pnp_card_driver opti9xx_pnpc_driver = { .flags = PNP_DRIVER_RES_DISABLE, .name = "opti9xx", .id_table = snd_opti9xx_pnpids, .probe = snd_opti9xx_pnp_probe, .remove = __devexit_p(snd_opti9xx_pnp_remove), +#ifdef CONFIG_PM + .suspend = snd_opti9xx_pnp_suspend, + .resume = snd_opti9xx_pnp_resume, +#endif }; #endif diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 49c8a0c2442c..360b08b03e1d 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1456,7 +1456,6 @@ static struct snd_pcm_hardware snd_wss_playback = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_SYNC_START), .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM | SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE), @@ -1657,6 +1656,10 @@ static void snd_wss_resume(struct snd_wss *chip) break; } } + /* Yamaha needs this to resume properly */ + if (chip->hardware == WSS_HW_OPL3SA2) + snd_wss_out(chip, CS4231_PLAYBK_FORMAT, + chip->image[CS4231_PLAYBK_FORMAT]); spin_unlock_irqrestore(&chip->reg_lock, flags); #if 1 snd_wss_mce_down(chip); diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c index 09d46484bc1a..7d8803a00b79 100644 --- a/sound/oss/swarm_cs4297a.c +++ b/sound/oss/swarm_cs4297a.c @@ -69,7 +69,6 @@ #include <linux/sound.h> #include <linux/slab.h> #include <linux/soundcard.h> -#include <linux/ac97_codec.h> #include <linux/pci.h> #include <linux/bitops.h> #include <linux/interrupt.h> @@ -199,6 +198,22 @@ static const char invalid_magic[] = } \ }) +/* AC97 registers */ +#define AC97_MASTER_VOL_STEREO 0x0002 /* Line Out */ +#define AC97_PCBEEP_VOL 0x000a /* none */ +#define AC97_PHONE_VOL 0x000c /* TAD Input (mono) */ +#define AC97_MIC_VOL 0x000e /* MIC Input (mono) */ +#define AC97_LINEIN_VOL 0x0010 /* Line Input (stereo) */ +#define AC97_CD_VOL 0x0012 /* CD Input (stereo) */ +#define AC97_AUX_VOL 0x0016 /* Aux Input (stereo) */ +#define AC97_PCMOUT_VOL 0x0018 /* Wave Output (stereo) */ +#define AC97_RECORD_SELECT 0x001a /* */ +#define AC97_RECORD_GAIN 0x001c +#define AC97_GENERAL_PURPOSE 0x0020 +#define AC97_3D_CONTROL 0x0022 +#define AC97_POWER_CONTROL 0x0026 +#define AC97_VENDOR_ID1 0x007c + struct list_head cs4297a_devs = { &cs4297a_devs, &cs4297a_devs }; typedef struct serdma_descr_s { diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index 643f1113b1d8..7e814a5c3677 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -438,7 +438,7 @@ static __inline__ void li_writeb(lithium_t *lith, int off, unsigned char val) * * Observe that (mask & -mask) is (1 << low_set_bit_of(mask)). * As long as mask is constant, we trust the compiler will change the - * multipy and divide into shifts. + * multiply and divide into shifts. */ #define SHIFT_FIELD(val, mask) (((val) * ((mask) & -(mask))) & (mask)) diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 5ca0939e4223..ff3af6e77d61 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -228,7 +228,7 @@ config SND_OXYGEN Say Y here to include support for sound cards based on the C-Media CMI8788 (Oxygen HD Audio) chip: * Asound A-8788 - * Asus Xonar DG + * Asus Xonar DG/DGX * AuzenTech X-Meridian * AuzenTech X-Meridian 2G * Bgears b-Enspirer diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 9d91d61902b4..e672ff4df2da 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -1062,17 +1062,4 @@ static struct pci_driver ad1889_pci_driver = { .remove = __devexit_p(snd_ad1889_remove), }; -static int __init -alsa_ad1889_init(void) -{ - return pci_register_driver(&ad1889_pci_driver); -} - -static void __exit -alsa_ad1889_fini(void) -{ - pci_unregister_driver(&ad1889_pci_driver); -} - -module_init(alsa_ad1889_init); -module_exit(alsa_ad1889_fini); +module_pci_driver(ad1889_pci_driver); diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index bdd6164e9c7e..ee895f3c8605 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -1884,9 +1884,10 @@ static int __devinit snd_ali_mixer(struct snd_ali * codec) } #ifdef CONFIG_PM -static int ali_suspend(struct pci_dev *pci, pm_message_t state) +static int ali_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_ali *chip = card->private_data; struct snd_ali_image *im; int i, j; @@ -1929,13 +1930,14 @@ static int ali_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int ali_resume(struct pci_dev *pci) +static int ali_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_ali *chip = card->private_data; struct snd_ali_image *im; int i, j; @@ -1982,6 +1984,11 @@ static int ali_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(ali_pm, ali_suspend, ali_resume); +#define ALI_PM_OPS &ali_pm +#else +#define ALI_PM_OPS NULL #endif /* CONFIG_PM */ static int snd_ali_free(struct snd_ali * codec) @@ -2294,26 +2301,14 @@ static void __devexit snd_ali_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver ali5451_driver = { .name = KBUILD_MODNAME, .id_table = snd_ali_ids, .probe = snd_ali_probe, .remove = __devexit_p(snd_ali_remove), -#ifdef CONFIG_PM - .suspend = ali_suspend, - .resume = ali_resume, -#endif + .driver = { + .pm = ALI_PM_OPS, + }, }; -static int __init alsa_card_ali_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_ali_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_ali_init) -module_exit(alsa_card_ali_exit) +module_pci_driver(ali5451_driver); diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 8196e229b2df..68c4469c6d19 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -766,9 +766,10 @@ static int __devinit snd_als300_create(struct snd_card *card, } #ifdef CONFIG_PM -static int snd_als300_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_als300_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_als300 *chip = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -777,13 +778,14 @@ static int snd_als300_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_als300_resume(struct pci_dev *pci) +static int snd_als300_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_als300 *chip = card->private_data; pci_set_power_state(pci, PCI_D0); @@ -802,6 +804,11 @@ static int snd_als300_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(snd_als300_pm, snd_als300_suspend, snd_als300_resume); +#define SND_ALS300_PM_OPS &snd_als300_pm +#else +#define SND_ALS300_PM_OPS NULL #endif static int __devinit snd_als300_probe(struct pci_dev *pci, @@ -852,26 +859,14 @@ static int __devinit snd_als300_probe(struct pci_dev *pci, return 0; } -static struct pci_driver driver = { +static struct pci_driver als300_driver = { .name = KBUILD_MODNAME, .id_table = snd_als300_ids, .probe = snd_als300_probe, .remove = __devexit_p(snd_als300_remove), -#ifdef CONFIG_PM - .suspend = snd_als300_suspend, - .resume = snd_als300_resume, -#endif + .driver = { + .pm = SND_ALS300_PM_OPS, + }, }; -static int __init alsa_card_als300_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_als300_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_als300_init) -module_exit(alsa_card_als300_exit) +module_pci_driver(als300_driver); diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 3269b8011ea9..0eeca49c5754 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -988,9 +988,10 @@ static void __devexit snd_card_als4000_remove(struct pci_dev *pci) } #ifdef CONFIG_PM -static int snd_als4000_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_als4000_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_card_als4000 *acard = card->private_data; struct snd_sb *chip = acard->chip; @@ -1001,13 +1002,14 @@ static int snd_als4000_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_als4000_resume(struct pci_dev *pci) +static int snd_als4000_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_card_als4000 *acard = card->private_data; struct snd_sb *chip = acard->chip; @@ -1033,29 +1035,21 @@ static int snd_als4000_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -#endif /* CONFIG_PM */ +static SIMPLE_DEV_PM_OPS(snd_als4000_pm, snd_als4000_suspend, snd_als4000_resume); +#define SND_ALS4000_PM_OPS &snd_als4000_pm +#else +#define SND_ALS4000_PM_OPS NULL +#endif /* CONFIG_PM */ -static struct pci_driver driver = { +static struct pci_driver als4000_driver = { .name = KBUILD_MODNAME, .id_table = snd_als4000_ids, .probe = snd_card_als4000_probe, .remove = __devexit_p(snd_card_als4000_remove), -#ifdef CONFIG_PM - .suspend = snd_als4000_suspend, - .resume = snd_als4000_resume, -#endif + .driver = { + .pm = SND_ALS4000_PM_OPS, + }, }; -static int __init alsa_card_als4000_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_als4000_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_als4000_init) -module_exit(alsa_card_als4000_exit) +module_pci_driver(als4000_driver); diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 590682f115ef..31020d2a868b 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1462,9 +1462,10 @@ static int __devinit snd_atiixp_mixer_new(struct atiixp *chip, int clock, /* * power management */ -static int snd_atiixp_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_atiixp_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct atiixp *chip = card->private_data; int i; @@ -1484,13 +1485,14 @@ static int snd_atiixp_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_atiixp_resume(struct pci_dev *pci) +static int snd_atiixp_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct atiixp *chip = card->private_data; int i; @@ -1526,6 +1528,11 @@ static int snd_atiixp_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(snd_atiixp_pm, snd_atiixp_suspend, snd_atiixp_resume); +#define SND_ATIIXP_PM_OPS &snd_atiixp_pm +#else +#define SND_ATIIXP_PM_OPS NULL #endif /* CONFIG_PM */ @@ -1700,27 +1707,14 @@ static void __devexit snd_atiixp_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver atiixp_driver = { .name = KBUILD_MODNAME, .id_table = snd_atiixp_ids, .probe = snd_atiixp_probe, .remove = __devexit_p(snd_atiixp_remove), -#ifdef CONFIG_PM - .suspend = snd_atiixp_suspend, - .resume = snd_atiixp_resume, -#endif + .driver = { + .pm = SND_ATIIXP_PM_OPS, + }, }; - -static int __init alsa_card_atiixp_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_atiixp_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_atiixp_init) -module_exit(alsa_card_atiixp_exit) +module_pci_driver(atiixp_driver); diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 524d35f31232..79e204ec623f 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -1117,9 +1117,10 @@ static int __devinit snd_atiixp_mixer_new(struct atiixp_modem *chip, int clock) /* * power management */ -static int snd_atiixp_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_atiixp_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct atiixp_modem *chip = card->private_data; int i; @@ -1133,13 +1134,14 @@ static int snd_atiixp_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_atiixp_resume(struct pci_dev *pci) +static int snd_atiixp_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct atiixp_modem *chip = card->private_data; int i; @@ -1162,8 +1164,12 @@ static int snd_atiixp_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -#endif /* CONFIG_PM */ +static SIMPLE_DEV_PM_OPS(snd_atiixp_pm, snd_atiixp_suspend, snd_atiixp_resume); +#define SND_ATIIXP_PM_OPS &snd_atiixp_pm +#else +#define SND_ATIIXP_PM_OPS NULL +#endif /* CONFIG_PM */ #ifdef CONFIG_PROC_FS /* @@ -1331,27 +1337,14 @@ static void __devexit snd_atiixp_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver atiixp_modem_driver = { .name = KBUILD_MODNAME, .id_table = snd_atiixp_ids, .probe = snd_atiixp_probe, .remove = __devexit_p(snd_atiixp_remove), -#ifdef CONFIG_PM - .suspend = snd_atiixp_suspend, - .resume = snd_atiixp_resume, -#endif + .driver = { + .pm = SND_ATIIXP_PM_OPS, + }, }; - -static int __init alsa_card_atiixp_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_atiixp_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_atiixp_init) -module_exit(alsa_card_atiixp_exit) +module_pci_driver(atiixp_modem_driver); diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index f13ad536b2d5..ffc376f9f4e4 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -375,24 +375,11 @@ static void __devexit snd_vortex_remove(struct pci_dev *pci) } // pci_driver definition -static struct pci_driver driver = { +static struct pci_driver vortex_driver = { .name = KBUILD_MODNAME, .id_table = snd_vortex_ids, .probe = snd_vortex_probe, .remove = __devexit_p(snd_vortex_remove), }; -// initialization of the module -static int __init alsa_card_vortex_init(void) -{ - return pci_register_driver(&driver); -} - -// clean up the module -static void __exit alsa_card_vortex_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_vortex_init) -module_exit(alsa_card_vortex_exit) +module_pci_driver(vortex_driver); diff --git a/sound/pci/au88x0/au88x0_mixer.c b/sound/pci/au88x0/au88x0_mixer.c index 557c782ae4fc..fa13efbebdaf 100644 --- a/sound/pci/au88x0/au88x0_mixer.c +++ b/sound/pci/au88x0/au88x0_mixer.c @@ -10,6 +10,15 @@ #include <sound/core.h> #include "au88x0.h" +static int remove_ctl(struct snd_card *card, const char *name) +{ + struct snd_ctl_elem_id id; + memset(&id, 0, sizeof(id)); + strcpy(id.name, name); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + return snd_ctl_remove_id(card, &id); +} + static int __devinit snd_vortex_mixer(vortex_t * vortex) { struct snd_ac97_bus *pbus; @@ -28,5 +37,7 @@ static int __devinit snd_vortex_mixer(vortex_t * vortex) ac97.scaps = AC97_SCAP_NO_SPDIF; err = snd_ac97_mixer(pbus, &ac97, &vortex->codec); vortex->isquad = ((vortex->codec == NULL) ? 0 : (vortex->codec->ext_id&0x80)); + remove_ctl(vortex->card, "Master Mono Playback Volume"); + remove_ctl(vortex->card, "Master Mono Playback Switch"); return err; } diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 1c5231931462..0f804741825f 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -112,8 +112,6 @@ struct aw2 { /********************************* * FUNCTION DECLARATIONS ********************************/ -static int __init alsa_card_aw2_init(void); -static void __exit alsa_card_aw2_exit(void); static int snd_aw2_dev_free(struct snd_device *device); static int __devinit snd_aw2_create(struct snd_card *card, struct pci_dev *pci, struct aw2 **rchip); @@ -171,13 +169,15 @@ static DEFINE_PCI_DEVICE_TABLE(snd_aw2_ids) = { MODULE_DEVICE_TABLE(pci, snd_aw2_ids); /* pci_driver definition */ -static struct pci_driver driver = { +static struct pci_driver aw2_driver = { .name = KBUILD_MODNAME, .id_table = snd_aw2_ids, .probe = snd_aw2_probe, .remove = __devexit_p(snd_aw2_remove), }; +module_pci_driver(aw2_driver); + /* operators for playback PCM alsa interface */ static struct snd_pcm_ops snd_aw2_playback_ops = { .open = snd_aw2_pcm_playback_open, @@ -217,23 +217,6 @@ static struct snd_kcontrol_new aw2_control __devinitdata = { * FUNCTION IMPLEMENTATIONS ********************************/ -/* initialization of the module */ -static int __init alsa_card_aw2_init(void) -{ - snd_printdd(KERN_DEBUG "aw2: Load aw2 module\n"); - return pci_register_driver(&driver); -} - -/* clean up the module */ -static void __exit alsa_card_aw2_exit(void) -{ - snd_printdd(KERN_DEBUG "aw2: Unload aw2 module\n"); - pci_unregister_driver(&driver); -} - -module_init(alsa_card_aw2_init); -module_exit(alsa_card_aw2_exit); - /* component-destructor */ static int snd_aw2_dev_free(struct snd_device *device) { diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 496f14c1a731..4dddd871548b 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2794,9 +2794,10 @@ snd_azf3328_resume_ac97(const struct snd_azf3328 *chip) } static int -snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state) +snd_azf3328_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_azf3328 *chip = card->private_data; u16 *saved_regs_ctrl_u16; @@ -2824,14 +2825,15 @@ snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } static int -snd_azf3328_resume(struct pci_dev *pci) +snd_azf3328_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); const struct snd_azf3328 *chip = card->private_data; pci_set_power_state(pci, PCI_D0); @@ -2859,37 +2861,21 @@ snd_azf3328_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -#endif /* CONFIG_PM */ +static SIMPLE_DEV_PM_OPS(snd_azf3328_pm, snd_azf3328_suspend, snd_azf3328_resume); +#define SND_AZF3328_PM_OPS &snd_azf3328_pm +#else +#define SND_AZF3328_PM_OPS NULL +#endif /* CONFIG_PM */ -static struct pci_driver driver = { +static struct pci_driver azf3328_driver = { .name = KBUILD_MODNAME, .id_table = snd_azf3328_ids, .probe = snd_azf3328_probe, .remove = __devexit_p(snd_azf3328_remove), -#ifdef CONFIG_PM - .suspend = snd_azf3328_suspend, - .resume = snd_azf3328_resume, -#endif + .driver = { + .pm = SND_AZF3328_PM_OPS, + }, }; -static int __init -alsa_card_azf3328_init(void) -{ - int err; - snd_azf3328_dbgcallenter(); - err = pci_register_driver(&driver); - snd_azf3328_dbgcallleave(); - return err; -} - -static void __exit -alsa_card_azf3328_exit(void) -{ - snd_azf3328_dbgcallenter(); - pci_unregister_driver(&driver); - snd_azf3328_dbgcallleave(); -} - -module_init(alsa_card_azf3328_init) -module_exit(alsa_card_azf3328_exit) +module_pci_driver(azf3328_driver); diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 62d6163fc9d9..b6a95eeca095 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -836,8 +836,6 @@ static struct { {0x7063, 0x2000}, /* pcHDTV HD-2000 TV */ }; -static struct pci_driver driver; - /* return the id of the card, or a negative value if it's blacklisted */ static int __devinit snd_bt87x_detect_card(struct pci_dev *pci) { @@ -964,24 +962,11 @@ static DEFINE_PCI_DEVICE_TABLE(snd_bt87x_default_ids) = { { } }; -static struct pci_driver driver = { +static struct pci_driver bt87x_driver = { .name = KBUILD_MODNAME, .id_table = snd_bt87x_ids, .probe = snd_bt87x_probe, .remove = __devexit_p(snd_bt87x_remove), }; -static int __init alsa_card_bt87x_init(void) -{ - if (load_all) - driver.id_table = snd_bt87x_default_ids; - return pci_register_driver(&driver); -} - -static void __exit alsa_card_bt87x_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_bt87x_init) -module_exit(alsa_card_bt87x_exit) +module_pci_driver(bt87x_driver); diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 08d6ebfe5a61..83277b747b36 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1872,9 +1872,10 @@ static void __devexit snd_ca0106_remove(struct pci_dev *pci) } #ifdef CONFIG_PM -static int snd_ca0106_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_ca0106_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_ca0106 *chip = card->private_data; int i; @@ -1889,13 +1890,14 @@ static int snd_ca0106_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_ca0106_resume(struct pci_dev *pci) +static int snd_ca0106_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_ca0106 *chip = card->private_data; int i; @@ -1922,6 +1924,11 @@ static int snd_ca0106_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(snd_ca0106_pm, snd_ca0106_suspend, snd_ca0106_resume); +#define SND_CA0106_PM_OPS &snd_ca0106_pm +#else +#define SND_CA0106_PM_OPS NULL #endif // PCI IDs @@ -1932,28 +1939,14 @@ static DEFINE_PCI_DEVICE_TABLE(snd_ca0106_ids) = { MODULE_DEVICE_TABLE(pci, snd_ca0106_ids); // pci_driver definition -static struct pci_driver driver = { +static struct pci_driver ca0106_driver = { .name = KBUILD_MODNAME, .id_table = snd_ca0106_ids, .probe = snd_ca0106_probe, .remove = __devexit_p(snd_ca0106_remove), -#ifdef CONFIG_PM - .suspend = snd_ca0106_suspend, - .resume = snd_ca0106_resume, -#endif + .driver = { + .pm = SND_CA0106_PM_OPS, + }, }; -// initialization of the module -static int __init alsa_card_ca0106_init(void) -{ - return pci_register_driver(&driver); -} - -// clean up the module -static void __exit alsa_card_ca0106_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_ca0106_init) -module_exit(alsa_card_ca0106_exit) +module_pci_driver(ca0106_driver); diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 19b06269adc2..b7d6f2b886ef 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -3338,9 +3338,10 @@ static unsigned char saved_mixers[] = { SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, }; -static int snd_cmipci_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_cmipci_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct cmipci *cm = card->private_data; int i; @@ -3361,13 +3362,14 @@ static int snd_cmipci_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_cmipci_resume(struct pci_dev *pci) +static int snd_cmipci_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct cmipci *cm = card->private_data; int i; @@ -3396,28 +3398,21 @@ static int snd_cmipci_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(snd_cmipci_pm, snd_cmipci_suspend, snd_cmipci_resume); +#define SND_CMIPCI_PM_OPS &snd_cmipci_pm +#else +#define SND_CMIPCI_PM_OPS NULL #endif /* CONFIG_PM */ -static struct pci_driver driver = { +static struct pci_driver cmipci_driver = { .name = KBUILD_MODNAME, .id_table = snd_cmipci_ids, .probe = snd_cmipci_probe, .remove = __devexit_p(snd_cmipci_remove), -#ifdef CONFIG_PM - .suspend = snd_cmipci_suspend, - .resume = snd_cmipci_resume, -#endif + .driver = { + .pm = SND_CMIPCI_PM_OPS, + }, }; -static int __init alsa_card_cmipci_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_cmipci_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_cmipci_init) -module_exit(alsa_card_cmipci_exit) +module_pci_driver(cmipci_driver); diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index a9f368f60df6..45a8317085f4 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -1997,9 +1997,10 @@ static int saved_regs[SUSPEND_REGISTERS] = { #define CLKCR1_CKRA 0x00010000L -static int cs4281_suspend(struct pci_dev *pci, pm_message_t state) +static int cs4281_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct cs4281 *chip = card->private_data; u32 ulCLK; unsigned int i; @@ -2040,13 +2041,14 @@ static int cs4281_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int cs4281_resume(struct pci_dev *pci) +static int cs4281_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct cs4281 *chip = card->private_data; unsigned int i; u32 ulCLK; @@ -2082,28 +2084,21 @@ static int cs4281_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(cs4281_pm, cs4281_suspend, cs4281_resume); +#define CS4281_PM_OPS &cs4281_pm +#else +#define CS4281_PM_OPS NULL #endif /* CONFIG_PM */ -static struct pci_driver driver = { +static struct pci_driver cs4281_driver = { .name = KBUILD_MODNAME, .id_table = snd_cs4281_ids, .probe = snd_cs4281_probe, .remove = __devexit_p(snd_cs4281_remove), -#ifdef CONFIG_PM - .suspend = cs4281_suspend, - .resume = cs4281_resume, -#endif + .driver = { + .pm = CS4281_PM_OPS, + }, }; -static int __init alsa_card_cs4281_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_cs4281_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_cs4281_init) -module_exit(alsa_card_cs4281_exit) +module_pci_driver(cs4281_driver); diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index 819d79d0586d..1e007c736a8b 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -30,7 +30,7 @@ #include <linux/init.h> #include <linux/module.h> #include <sound/core.h> -#include <sound/cs46xx.h> +#include "cs46xx.h" #include <sound/initval.h> MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); @@ -161,26 +161,16 @@ static void __devexit snd_card_cs46xx_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver cs46xx_driver = { .name = KBUILD_MODNAME, .id_table = snd_cs46xx_ids, .probe = snd_card_cs46xx_probe, .remove = __devexit_p(snd_card_cs46xx_remove), #ifdef CONFIG_PM - .suspend = snd_cs46xx_suspend, - .resume = snd_cs46xx_resume, + .driver = { + .pm = &snd_cs46xx_pm, + }, #endif }; -static int __init alsa_card_cs46xx_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_cs46xx_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_cs46xx_init) -module_exit(alsa_card_cs46xx_exit) +module_pci_driver(cs46xx_driver); diff --git a/sound/pci/cs46xx/cs46xx.h b/sound/pci/cs46xx/cs46xx.h new file mode 100644 index 000000000000..29d8a8da1ba7 --- /dev/null +++ b/sound/pci/cs46xx/cs46xx.h @@ -0,0 +1,1744 @@ +#ifndef __SOUND_CS46XX_H +#define __SOUND_CS46XX_H + +/* + * Copyright (c) by Jaroslav Kysela <perex@perex.cz>, + * Cirrus Logic, Inc. + * Definitions for Cirrus Logic CS46xx chips + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <sound/pcm.h> +#include <sound/pcm-indirect.h> +#include <sound/rawmidi.h> +#include <sound/ac97_codec.h> +#include "cs46xx_dsp_spos.h" + +/* + * Direct registers + */ + +/* + * The following define the offsets of the registers accessed via base address + * register zero on the CS46xx part. + */ +#define BA0_HISR 0x00000000 +#define BA0_HSR0 0x00000004 +#define BA0_HICR 0x00000008 +#define BA0_DMSR 0x00000100 +#define BA0_HSAR 0x00000110 +#define BA0_HDAR 0x00000114 +#define BA0_HDMR 0x00000118 +#define BA0_HDCR 0x0000011C +#define BA0_PFMC 0x00000200 +#define BA0_PFCV1 0x00000204 +#define BA0_PFCV2 0x00000208 +#define BA0_PCICFG00 0x00000300 +#define BA0_PCICFG04 0x00000304 +#define BA0_PCICFG08 0x00000308 +#define BA0_PCICFG0C 0x0000030C +#define BA0_PCICFG10 0x00000310 +#define BA0_PCICFG14 0x00000314 +#define BA0_PCICFG18 0x00000318 +#define BA0_PCICFG1C 0x0000031C +#define BA0_PCICFG20 0x00000320 +#define BA0_PCICFG24 0x00000324 +#define BA0_PCICFG28 0x00000328 +#define BA0_PCICFG2C 0x0000032C +#define BA0_PCICFG30 0x00000330 +#define BA0_PCICFG34 0x00000334 +#define BA0_PCICFG38 0x00000338 +#define BA0_PCICFG3C 0x0000033C +#define BA0_CLKCR1 0x00000400 +#define BA0_CLKCR2 0x00000404 +#define BA0_PLLM 0x00000408 +#define BA0_PLLCC 0x0000040C +#define BA0_FRR 0x00000410 +#define BA0_CFL1 0x00000414 +#define BA0_CFL2 0x00000418 +#define BA0_SERMC1 0x00000420 +#define BA0_SERMC2 0x00000424 +#define BA0_SERC1 0x00000428 +#define BA0_SERC2 0x0000042C +#define BA0_SERC3 0x00000430 +#define BA0_SERC4 0x00000434 +#define BA0_SERC5 0x00000438 +#define BA0_SERBSP 0x0000043C +#define BA0_SERBST 0x00000440 +#define BA0_SERBCM 0x00000444 +#define BA0_SERBAD 0x00000448 +#define BA0_SERBCF 0x0000044C +#define BA0_SERBWP 0x00000450 +#define BA0_SERBRP 0x00000454 +#ifndef NO_CS4612 +#define BA0_ASER_FADDR 0x00000458 +#endif +#define BA0_ACCTL 0x00000460 +#define BA0_ACSTS 0x00000464 +#define BA0_ACOSV 0x00000468 +#define BA0_ACCAD 0x0000046C +#define BA0_ACCDA 0x00000470 +#define BA0_ACISV 0x00000474 +#define BA0_ACSAD 0x00000478 +#define BA0_ACSDA 0x0000047C +#define BA0_JSPT 0x00000480 +#define BA0_JSCTL 0x00000484 +#define BA0_JSC1 0x00000488 +#define BA0_JSC2 0x0000048C +#define BA0_MIDCR 0x00000490 +#define BA0_MIDSR 0x00000494 +#define BA0_MIDWP 0x00000498 +#define BA0_MIDRP 0x0000049C +#define BA0_JSIO 0x000004A0 +#ifndef NO_CS4612 +#define BA0_ASER_MASTER 0x000004A4 +#endif +#define BA0_CFGI 0x000004B0 +#define BA0_SSVID 0x000004B4 +#define BA0_GPIOR 0x000004B8 +#ifndef NO_CS4612 +#define BA0_EGPIODR 0x000004BC +#define BA0_EGPIOPTR 0x000004C0 +#define BA0_EGPIOTR 0x000004C4 +#define BA0_EGPIOWR 0x000004C8 +#define BA0_EGPIOSR 0x000004CC +#define BA0_SERC6 0x000004D0 +#define BA0_SERC7 0x000004D4 +#define BA0_SERACC 0x000004D8 +#define BA0_ACCTL2 0x000004E0 +#define BA0_ACSTS2 0x000004E4 +#define BA0_ACOSV2 0x000004E8 +#define BA0_ACCAD2 0x000004EC +#define BA0_ACCDA2 0x000004F0 +#define BA0_ACISV2 0x000004F4 +#define BA0_ACSAD2 0x000004F8 +#define BA0_ACSDA2 0x000004FC +#define BA0_IOTAC0 0x00000500 +#define BA0_IOTAC1 0x00000504 +#define BA0_IOTAC2 0x00000508 +#define BA0_IOTAC3 0x0000050C +#define BA0_IOTAC4 0x00000510 +#define BA0_IOTAC5 0x00000514 +#define BA0_IOTAC6 0x00000518 +#define BA0_IOTAC7 0x0000051C +#define BA0_IOTAC8 0x00000520 +#define BA0_IOTAC9 0x00000524 +#define BA0_IOTAC10 0x00000528 +#define BA0_IOTAC11 0x0000052C +#define BA0_IOTFR0 0x00000540 +#define BA0_IOTFR1 0x00000544 +#define BA0_IOTFR2 0x00000548 +#define BA0_IOTFR3 0x0000054C +#define BA0_IOTFR4 0x00000550 +#define BA0_IOTFR5 0x00000554 +#define BA0_IOTFR6 0x00000558 +#define BA0_IOTFR7 0x0000055C +#define BA0_IOTFIFO 0x00000580 +#define BA0_IOTRRD 0x00000584 +#define BA0_IOTFP 0x00000588 +#define BA0_IOTCR 0x0000058C +#define BA0_DPCID 0x00000590 +#define BA0_DPCIA 0x00000594 +#define BA0_DPCIC 0x00000598 +#define BA0_PCPCIR 0x00000600 +#define BA0_PCPCIG 0x00000604 +#define BA0_PCPCIEN 0x00000608 +#define BA0_EPCIPMC 0x00000610 +#endif + +/* + * The following define the offsets of the registers and memories accessed via + * base address register one on the CS46xx part. + */ +#define BA1_SP_DMEM0 0x00000000 +#define BA1_SP_DMEM1 0x00010000 +#define BA1_SP_PMEM 0x00020000 +#define BA1_SP_REG 0x00030000 +#define BA1_SPCR 0x00030000 +#define BA1_DREG 0x00030004 +#define BA1_DSRWP 0x00030008 +#define BA1_TWPR 0x0003000C +#define BA1_SPWR 0x00030010 +#define BA1_SPIR 0x00030014 +#define BA1_FGR1 0x00030020 +#define BA1_SPCS 0x00030028 +#define BA1_SDSR 0x0003002C +#define BA1_FRMT 0x00030030 +#define BA1_FRCC 0x00030034 +#define BA1_FRSC 0x00030038 +#define BA1_OMNI_MEM 0x000E0000 + + +/* + * The following defines are for the flags in the host interrupt status + * register. + */ +#define HISR_VC_MASK 0x0000FFFF +#define HISR_VC0 0x00000001 +#define HISR_VC1 0x00000002 +#define HISR_VC2 0x00000004 +#define HISR_VC3 0x00000008 +#define HISR_VC4 0x00000010 +#define HISR_VC5 0x00000020 +#define HISR_VC6 0x00000040 +#define HISR_VC7 0x00000080 +#define HISR_VC8 0x00000100 +#define HISR_VC9 0x00000200 +#define HISR_VC10 0x00000400 +#define HISR_VC11 0x00000800 +#define HISR_VC12 0x00001000 +#define HISR_VC13 0x00002000 +#define HISR_VC14 0x00004000 +#define HISR_VC15 0x00008000 +#define HISR_INT0 0x00010000 +#define HISR_INT1 0x00020000 +#define HISR_DMAI 0x00040000 +#define HISR_FROVR 0x00080000 +#define HISR_MIDI 0x00100000 +#ifdef NO_CS4612 +#define HISR_RESERVED 0x0FE00000 +#else +#define HISR_SBINT 0x00200000 +#define HISR_RESERVED 0x0FC00000 +#endif +#define HISR_H0P 0x40000000 +#define HISR_INTENA 0x80000000 + +/* + * The following defines are for the flags in the host signal register 0. + */ +#define HSR0_VC_MASK 0xFFFFFFFF +#define HSR0_VC16 0x00000001 +#define HSR0_VC17 0x00000002 +#define HSR0_VC18 0x00000004 +#define HSR0_VC19 0x00000008 +#define HSR0_VC20 0x00000010 +#define HSR0_VC21 0x00000020 +#define HSR0_VC22 0x00000040 +#define HSR0_VC23 0x00000080 +#define HSR0_VC24 0x00000100 +#define HSR0_VC25 0x00000200 +#define HSR0_VC26 0x00000400 +#define HSR0_VC27 0x00000800 +#define HSR0_VC28 0x00001000 +#define HSR0_VC29 0x00002000 +#define HSR0_VC30 0x00004000 +#define HSR0_VC31 0x00008000 +#define HSR0_VC32 0x00010000 +#define HSR0_VC33 0x00020000 +#define HSR0_VC34 0x00040000 +#define HSR0_VC35 0x00080000 +#define HSR0_VC36 0x00100000 +#define HSR0_VC37 0x00200000 +#define HSR0_VC38 0x00400000 +#define HSR0_VC39 0x00800000 +#define HSR0_VC40 0x01000000 +#define HSR0_VC41 0x02000000 +#define HSR0_VC42 0x04000000 +#define HSR0_VC43 0x08000000 +#define HSR0_VC44 0x10000000 +#define HSR0_VC45 0x20000000 +#define HSR0_VC46 0x40000000 +#define HSR0_VC47 0x80000000 + +/* + * The following defines are for the flags in the host interrupt control + * register. + */ +#define HICR_IEV 0x00000001 +#define HICR_CHGM 0x00000002 + +/* + * The following defines are for the flags in the DMA status register. + */ +#define DMSR_HP 0x00000001 +#define DMSR_HR 0x00000002 +#define DMSR_SP 0x00000004 +#define DMSR_SR 0x00000008 + +/* + * The following defines are for the flags in the host DMA source address + * register. + */ +#define HSAR_HOST_ADDR_MASK 0xFFFFFFFF +#define HSAR_DSP_ADDR_MASK 0x0000FFFF +#define HSAR_MEMID_MASK 0x000F0000 +#define HSAR_MEMID_SP_DMEM0 0x00000000 +#define HSAR_MEMID_SP_DMEM1 0x00010000 +#define HSAR_MEMID_SP_PMEM 0x00020000 +#define HSAR_MEMID_SP_DEBUG 0x00030000 +#define HSAR_MEMID_OMNI_MEM 0x000E0000 +#define HSAR_END 0x40000000 +#define HSAR_ERR 0x80000000 + +/* + * The following defines are for the flags in the host DMA destination address + * register. + */ +#define HDAR_HOST_ADDR_MASK 0xFFFFFFFF +#define HDAR_DSP_ADDR_MASK 0x0000FFFF +#define HDAR_MEMID_MASK 0x000F0000 +#define HDAR_MEMID_SP_DMEM0 0x00000000 +#define HDAR_MEMID_SP_DMEM1 0x00010000 +#define HDAR_MEMID_SP_PMEM 0x00020000 +#define HDAR_MEMID_SP_DEBUG 0x00030000 +#define HDAR_MEMID_OMNI_MEM 0x000E0000 +#define HDAR_END 0x40000000 +#define HDAR_ERR 0x80000000 + +/* + * The following defines are for the flags in the host DMA control register. + */ +#define HDMR_AC_MASK 0x0000F000 +#define HDMR_AC_8_16 0x00001000 +#define HDMR_AC_M_S 0x00002000 +#define HDMR_AC_B_L 0x00004000 +#define HDMR_AC_S_U 0x00008000 + +/* + * The following defines are for the flags in the host DMA control register. + */ +#define HDCR_COUNT_MASK 0x000003FF +#define HDCR_DONE 0x00004000 +#define HDCR_OPT 0x00008000 +#define HDCR_WBD 0x00400000 +#define HDCR_WBS 0x00800000 +#define HDCR_DMS_MASK 0x07000000 +#define HDCR_DMS_LINEAR 0x00000000 +#define HDCR_DMS_16_DWORDS 0x01000000 +#define HDCR_DMS_32_DWORDS 0x02000000 +#define HDCR_DMS_64_DWORDS 0x03000000 +#define HDCR_DMS_128_DWORDS 0x04000000 +#define HDCR_DMS_256_DWORDS 0x05000000 +#define HDCR_DMS_512_DWORDS 0x06000000 +#define HDCR_DMS_1024_DWORDS 0x07000000 +#define HDCR_DH 0x08000000 +#define HDCR_SMS_MASK 0x70000000 +#define HDCR_SMS_LINEAR 0x00000000 +#define HDCR_SMS_16_DWORDS 0x10000000 +#define HDCR_SMS_32_DWORDS 0x20000000 +#define HDCR_SMS_64_DWORDS 0x30000000 +#define HDCR_SMS_128_DWORDS 0x40000000 +#define HDCR_SMS_256_DWORDS 0x50000000 +#define HDCR_SMS_512_DWORDS 0x60000000 +#define HDCR_SMS_1024_DWORDS 0x70000000 +#define HDCR_SH 0x80000000 +#define HDCR_COUNT_SHIFT 0 + +/* + * The following defines are for the flags in the performance monitor control + * register. + */ +#define PFMC_C1SS_MASK 0x0000001F +#define PFMC_C1EV 0x00000020 +#define PFMC_C1RS 0x00008000 +#define PFMC_C2SS_MASK 0x001F0000 +#define PFMC_C2EV 0x00200000 +#define PFMC_C2RS 0x80000000 +#define PFMC_C1SS_SHIFT 0 +#define PFMC_C2SS_SHIFT 16 +#define PFMC_BUS_GRANT 0 +#define PFMC_GRANT_AFTER_REQ 1 +#define PFMC_TRANSACTION 2 +#define PFMC_DWORD_TRANSFER 3 +#define PFMC_SLAVE_READ 4 +#define PFMC_SLAVE_WRITE 5 +#define PFMC_PREEMPTION 6 +#define PFMC_DISCONNECT_RETRY 7 +#define PFMC_INTERRUPT 8 +#define PFMC_BUS_OWNERSHIP 9 +#define PFMC_TRANSACTION_LAG 10 +#define PFMC_PCI_CLOCK 11 +#define PFMC_SERIAL_CLOCK 12 +#define PFMC_SP_CLOCK 13 + +/* + * The following defines are for the flags in the performance counter value 1 + * register. + */ +#define PFCV1_PC1V_MASK 0xFFFFFFFF +#define PFCV1_PC1V_SHIFT 0 + +/* + * The following defines are for the flags in the performance counter value 2 + * register. + */ +#define PFCV2_PC2V_MASK 0xFFFFFFFF +#define PFCV2_PC2V_SHIFT 0 + +/* + * The following defines are for the flags in the clock control register 1. + */ +#define CLKCR1_OSCS 0x00000001 +#define CLKCR1_OSCP 0x00000002 +#define CLKCR1_PLLSS_MASK 0x0000000C +#define CLKCR1_PLLSS_SERIAL 0x00000000 +#define CLKCR1_PLLSS_CRYSTAL 0x00000004 +#define CLKCR1_PLLSS_PCI 0x00000008 +#define CLKCR1_PLLSS_RESERVED 0x0000000C +#define CLKCR1_PLLP 0x00000010 +#define CLKCR1_SWCE 0x00000020 +#define CLKCR1_PLLOS 0x00000040 + +/* + * The following defines are for the flags in the clock control register 2. + */ +#define CLKCR2_PDIVS_MASK 0x0000000F +#define CLKCR2_PDIVS_1 0x00000001 +#define CLKCR2_PDIVS_2 0x00000002 +#define CLKCR2_PDIVS_4 0x00000004 +#define CLKCR2_PDIVS_7 0x00000007 +#define CLKCR2_PDIVS_8 0x00000008 +#define CLKCR2_PDIVS_16 0x00000000 + +/* + * The following defines are for the flags in the PLL multiplier register. + */ +#define PLLM_MASK 0x000000FF +#define PLLM_SHIFT 0 + +/* + * The following defines are for the flags in the PLL capacitor coefficient + * register. + */ +#define PLLCC_CDR_MASK 0x00000007 +#ifndef NO_CS4610 +#define PLLCC_CDR_240_350_MHZ 0x00000000 +#define PLLCC_CDR_184_265_MHZ 0x00000001 +#define PLLCC_CDR_144_205_MHZ 0x00000002 +#define PLLCC_CDR_111_160_MHZ 0x00000003 +#define PLLCC_CDR_87_123_MHZ 0x00000004 +#define PLLCC_CDR_67_96_MHZ 0x00000005 +#define PLLCC_CDR_52_74_MHZ 0x00000006 +#define PLLCC_CDR_45_58_MHZ 0x00000007 +#endif +#ifndef NO_CS4612 +#define PLLCC_CDR_271_398_MHZ 0x00000000 +#define PLLCC_CDR_227_330_MHZ 0x00000001 +#define PLLCC_CDR_167_239_MHZ 0x00000002 +#define PLLCC_CDR_150_215_MHZ 0x00000003 +#define PLLCC_CDR_107_154_MHZ 0x00000004 +#define PLLCC_CDR_98_140_MHZ 0x00000005 +#define PLLCC_CDR_73_104_MHZ 0x00000006 +#define PLLCC_CDR_63_90_MHZ 0x00000007 +#endif +#define PLLCC_LPF_MASK 0x000000F8 +#ifndef NO_CS4610 +#define PLLCC_LPF_23850_60000_KHZ 0x00000000 +#define PLLCC_LPF_7960_26290_KHZ 0x00000008 +#define PLLCC_LPF_4160_10980_KHZ 0x00000018 +#define PLLCC_LPF_1740_4580_KHZ 0x00000038 +#define PLLCC_LPF_724_1910_KHZ 0x00000078 +#define PLLCC_LPF_317_798_KHZ 0x000000F8 +#endif +#ifndef NO_CS4612 +#define PLLCC_LPF_25580_64530_KHZ 0x00000000 +#define PLLCC_LPF_14360_37270_KHZ 0x00000008 +#define PLLCC_LPF_6100_16020_KHZ 0x00000018 +#define PLLCC_LPF_2540_6690_KHZ 0x00000038 +#define PLLCC_LPF_1050_2780_KHZ 0x00000078 +#define PLLCC_LPF_450_1160_KHZ 0x000000F8 +#endif + +/* + * The following defines are for the flags in the feature reporting register. + */ +#define FRR_FAB_MASK 0x00000003 +#define FRR_MASK_MASK 0x0000001C +#ifdef NO_CS4612 +#define FRR_CFOP_MASK 0x000000E0 +#else +#define FRR_CFOP_MASK 0x00000FE0 +#endif +#define FRR_CFOP_NOT_DVD 0x00000020 +#define FRR_CFOP_A3D 0x00000040 +#define FRR_CFOP_128_PIN 0x00000080 +#ifndef NO_CS4612 +#define FRR_CFOP_CS4280 0x00000800 +#endif +#define FRR_FAB_SHIFT 0 +#define FRR_MASK_SHIFT 2 +#define FRR_CFOP_SHIFT 5 + +/* + * The following defines are for the flags in the configuration load 1 + * register. + */ +#define CFL1_CLOCK_SOURCE_MASK 0x00000003 +#define CFL1_CLOCK_SOURCE_CS423X 0x00000000 +#define CFL1_CLOCK_SOURCE_AC97 0x00000001 +#define CFL1_CLOCK_SOURCE_CRYSTAL 0x00000002 +#define CFL1_CLOCK_SOURCE_DUAL_AC97 0x00000003 +#define CFL1_VALID_DATA_MASK 0x000000FF + +/* + * The following defines are for the flags in the configuration load 2 + * register. + */ +#define CFL2_VALID_DATA_MASK 0x000000FF + +/* + * The following defines are for the flags in the serial port master control + * register 1. + */ +#define SERMC1_MSPE 0x00000001 +#define SERMC1_PTC_MASK 0x0000000E +#define SERMC1_PTC_CS423X 0x00000000 +#define SERMC1_PTC_AC97 0x00000002 +#define SERMC1_PTC_DAC 0x00000004 +#define SERMC1_PLB 0x00000010 +#define SERMC1_XLB 0x00000020 + +/* + * The following defines are for the flags in the serial port master control + * register 2. + */ +#define SERMC2_LROE 0x00000001 +#define SERMC2_MCOE 0x00000002 +#define SERMC2_MCDIV 0x00000004 + +/* + * The following defines are for the flags in the serial port 1 configuration + * register. + */ +#define SERC1_SO1EN 0x00000001 +#define SERC1_SO1F_MASK 0x0000000E +#define SERC1_SO1F_CS423X 0x00000000 +#define SERC1_SO1F_AC97 0x00000002 +#define SERC1_SO1F_DAC 0x00000004 +#define SERC1_SO1F_SPDIF 0x00000006 + +/* + * The following defines are for the flags in the serial port 2 configuration + * register. + */ +#define SERC2_SI1EN 0x00000001 +#define SERC2_SI1F_MASK 0x0000000E +#define SERC2_SI1F_CS423X 0x00000000 +#define SERC2_SI1F_AC97 0x00000002 +#define SERC2_SI1F_ADC 0x00000004 +#define SERC2_SI1F_SPDIF 0x00000006 + +/* + * The following defines are for the flags in the serial port 3 configuration + * register. + */ +#define SERC3_SO2EN 0x00000001 +#define SERC3_SO2F_MASK 0x00000006 +#define SERC3_SO2F_DAC 0x00000000 +#define SERC3_SO2F_SPDIF 0x00000002 + +/* + * The following defines are for the flags in the serial port 4 configuration + * register. + */ +#define SERC4_SO3EN 0x00000001 +#define SERC4_SO3F_MASK 0x00000006 +#define SERC4_SO3F_DAC 0x00000000 +#define SERC4_SO3F_SPDIF 0x00000002 + +/* + * The following defines are for the flags in the serial port 5 configuration + * register. + */ +#define SERC5_SI2EN 0x00000001 +#define SERC5_SI2F_MASK 0x00000006 +#define SERC5_SI2F_ADC 0x00000000 +#define SERC5_SI2F_SPDIF 0x00000002 + +/* + * The following defines are for the flags in the serial port backdoor sample + * pointer register. + */ +#define SERBSP_FSP_MASK 0x0000000F +#define SERBSP_FSP_SHIFT 0 + +/* + * The following defines are for the flags in the serial port backdoor status + * register. + */ +#define SERBST_RRDY 0x00000001 +#define SERBST_WBSY 0x00000002 + +/* + * The following defines are for the flags in the serial port backdoor command + * register. + */ +#define SERBCM_RDC 0x00000001 +#define SERBCM_WRC 0x00000002 + +/* + * The following defines are for the flags in the serial port backdoor address + * register. + */ +#ifdef NO_CS4612 +#define SERBAD_FAD_MASK 0x000000FF +#else +#define SERBAD_FAD_MASK 0x000001FF +#endif +#define SERBAD_FAD_SHIFT 0 + +/* + * The following defines are for the flags in the serial port backdoor + * configuration register. + */ +#define SERBCF_HBP 0x00000001 + +/* + * The following defines are for the flags in the serial port backdoor write + * port register. + */ +#define SERBWP_FWD_MASK 0x000FFFFF +#define SERBWP_FWD_SHIFT 0 + +/* + * The following defines are for the flags in the serial port backdoor read + * port register. + */ +#define SERBRP_FRD_MASK 0x000FFFFF +#define SERBRP_FRD_SHIFT 0 + +/* + * The following defines are for the flags in the async FIFO address register. + */ +#ifndef NO_CS4612 +#define ASER_FADDR_A1_MASK 0x000001FF +#define ASER_FADDR_EN1 0x00008000 +#define ASER_FADDR_A2_MASK 0x01FF0000 +#define ASER_FADDR_EN2 0x80000000 +#define ASER_FADDR_A1_SHIFT 0 +#define ASER_FADDR_A2_SHIFT 16 +#endif + +/* + * The following defines are for the flags in the AC97 control register. + */ +#define ACCTL_RSTN 0x00000001 +#define ACCTL_ESYN 0x00000002 +#define ACCTL_VFRM 0x00000004 +#define ACCTL_DCV 0x00000008 +#define ACCTL_CRW 0x00000010 +#define ACCTL_ASYN 0x00000020 +#ifndef NO_CS4612 +#define ACCTL_TC 0x00000040 +#endif + +/* + * The following defines are for the flags in the AC97 status register. + */ +#define ACSTS_CRDY 0x00000001 +#define ACSTS_VSTS 0x00000002 +#ifndef NO_CS4612 +#define ACSTS_WKUP 0x00000004 +#endif + +/* + * The following defines are for the flags in the AC97 output slot valid + * register. + */ +#define ACOSV_SLV3 0x00000001 +#define ACOSV_SLV4 0x00000002 +#define ACOSV_SLV5 0x00000004 +#define ACOSV_SLV6 0x00000008 +#define ACOSV_SLV7 0x00000010 +#define ACOSV_SLV8 0x00000020 +#define ACOSV_SLV9 0x00000040 +#define ACOSV_SLV10 0x00000080 +#define ACOSV_SLV11 0x00000100 +#define ACOSV_SLV12 0x00000200 + +/* + * The following defines are for the flags in the AC97 command address + * register. + */ +#define ACCAD_CI_MASK 0x0000007F +#define ACCAD_CI_SHIFT 0 + +/* + * The following defines are for the flags in the AC97 command data register. + */ +#define ACCDA_CD_MASK 0x0000FFFF +#define ACCDA_CD_SHIFT 0 + +/* + * The following defines are for the flags in the AC97 input slot valid + * register. + */ +#define ACISV_ISV3 0x00000001 +#define ACISV_ISV4 0x00000002 +#define ACISV_ISV5 0x00000004 +#define ACISV_ISV6 0x00000008 +#define ACISV_ISV7 0x00000010 +#define ACISV_ISV8 0x00000020 +#define ACISV_ISV9 0x00000040 +#define ACISV_ISV10 0x00000080 +#define ACISV_ISV11 0x00000100 +#define ACISV_ISV12 0x00000200 + +/* + * The following defines are for the flags in the AC97 status address + * register. + */ +#define ACSAD_SI_MASK 0x0000007F +#define ACSAD_SI_SHIFT 0 + +/* + * The following defines are for the flags in the AC97 status data register. + */ +#define ACSDA_SD_MASK 0x0000FFFF +#define ACSDA_SD_SHIFT 0 + +/* + * The following defines are for the flags in the joystick poll/trigger + * register. + */ +#define JSPT_CAX 0x00000001 +#define JSPT_CAY 0x00000002 +#define JSPT_CBX 0x00000004 +#define JSPT_CBY 0x00000008 +#define JSPT_BA1 0x00000010 +#define JSPT_BA2 0x00000020 +#define JSPT_BB1 0x00000040 +#define JSPT_BB2 0x00000080 + +/* + * The following defines are for the flags in the joystick control register. + */ +#define JSCTL_SP_MASK 0x00000003 +#define JSCTL_SP_SLOW 0x00000000 +#define JSCTL_SP_MEDIUM_SLOW 0x00000001 +#define JSCTL_SP_MEDIUM_FAST 0x00000002 +#define JSCTL_SP_FAST 0x00000003 +#define JSCTL_ARE 0x00000004 + +/* + * The following defines are for the flags in the joystick coordinate pair 1 + * readback register. + */ +#define JSC1_Y1V_MASK 0x0000FFFF +#define JSC1_X1V_MASK 0xFFFF0000 +#define JSC1_Y1V_SHIFT 0 +#define JSC1_X1V_SHIFT 16 + +/* + * The following defines are for the flags in the joystick coordinate pair 2 + * readback register. + */ +#define JSC2_Y2V_MASK 0x0000FFFF +#define JSC2_X2V_MASK 0xFFFF0000 +#define JSC2_Y2V_SHIFT 0 +#define JSC2_X2V_SHIFT 16 + +/* + * The following defines are for the flags in the MIDI control register. + */ +#define MIDCR_TXE 0x00000001 /* Enable transmitting. */ +#define MIDCR_RXE 0x00000002 /* Enable receiving. */ +#define MIDCR_RIE 0x00000004 /* Interrupt upon tx ready. */ +#define MIDCR_TIE 0x00000008 /* Interrupt upon rx ready. */ +#define MIDCR_MLB 0x00000010 /* Enable midi loopback. */ +#define MIDCR_MRST 0x00000020 /* Reset interface. */ + +/* + * The following defines are for the flags in the MIDI status register. + */ +#define MIDSR_TBF 0x00000001 /* Tx FIFO is full. */ +#define MIDSR_RBE 0x00000002 /* Rx FIFO is empty. */ + +/* + * The following defines are for the flags in the MIDI write port register. + */ +#define MIDWP_MWD_MASK 0x000000FF +#define MIDWP_MWD_SHIFT 0 + +/* + * The following defines are for the flags in the MIDI read port register. + */ +#define MIDRP_MRD_MASK 0x000000FF +#define MIDRP_MRD_SHIFT 0 + +/* + * The following defines are for the flags in the joystick GPIO register. + */ +#define JSIO_DAX 0x00000001 +#define JSIO_DAY 0x00000002 +#define JSIO_DBX 0x00000004 +#define JSIO_DBY 0x00000008 +#define JSIO_AXOE 0x00000010 +#define JSIO_AYOE 0x00000020 +#define JSIO_BXOE 0x00000040 +#define JSIO_BYOE 0x00000080 + +/* + * The following defines are for the flags in the master async/sync serial + * port enable register. + */ +#ifndef NO_CS4612 +#define ASER_MASTER_ME 0x00000001 +#endif + +/* + * The following defines are for the flags in the configuration interface + * register. + */ +#define CFGI_CLK 0x00000001 +#define CFGI_DOUT 0x00000002 +#define CFGI_DIN_EEN 0x00000004 +#define CFGI_EELD 0x00000008 + +/* + * The following defines are for the flags in the subsystem ID and vendor ID + * register. + */ +#define SSVID_VID_MASK 0x0000FFFF +#define SSVID_SID_MASK 0xFFFF0000 +#define SSVID_VID_SHIFT 0 +#define SSVID_SID_SHIFT 16 + +/* + * The following defines are for the flags in the GPIO pin interface register. + */ +#define GPIOR_VOLDN 0x00000001 +#define GPIOR_VOLUP 0x00000002 +#define GPIOR_SI2D 0x00000004 +#define GPIOR_SI2OE 0x00000008 + +/* + * The following defines are for the flags in the extended GPIO pin direction + * register. + */ +#ifndef NO_CS4612 +#define EGPIODR_GPOE0 0x00000001 +#define EGPIODR_GPOE1 0x00000002 +#define EGPIODR_GPOE2 0x00000004 +#define EGPIODR_GPOE3 0x00000008 +#define EGPIODR_GPOE4 0x00000010 +#define EGPIODR_GPOE5 0x00000020 +#define EGPIODR_GPOE6 0x00000040 +#define EGPIODR_GPOE7 0x00000080 +#define EGPIODR_GPOE8 0x00000100 +#endif + +/* + * The following defines are for the flags in the extended GPIO pin polarity/ + * type register. + */ +#ifndef NO_CS4612 +#define EGPIOPTR_GPPT0 0x00000001 +#define EGPIOPTR_GPPT1 0x00000002 +#define EGPIOPTR_GPPT2 0x00000004 +#define EGPIOPTR_GPPT3 0x00000008 +#define EGPIOPTR_GPPT4 0x00000010 +#define EGPIOPTR_GPPT5 0x00000020 +#define EGPIOPTR_GPPT6 0x00000040 +#define EGPIOPTR_GPPT7 0x00000080 +#define EGPIOPTR_GPPT8 0x00000100 +#endif + +/* + * The following defines are for the flags in the extended GPIO pin sticky + * register. + */ +#ifndef NO_CS4612 +#define EGPIOTR_GPS0 0x00000001 +#define EGPIOTR_GPS1 0x00000002 +#define EGPIOTR_GPS2 0x00000004 +#define EGPIOTR_GPS3 0x00000008 +#define EGPIOTR_GPS4 0x00000010 +#define EGPIOTR_GPS5 0x00000020 +#define EGPIOTR_GPS6 0x00000040 +#define EGPIOTR_GPS7 0x00000080 +#define EGPIOTR_GPS8 0x00000100 +#endif + +/* + * The following defines are for the flags in the extended GPIO ping wakeup + * register. + */ +#ifndef NO_CS4612 +#define EGPIOWR_GPW0 0x00000001 +#define EGPIOWR_GPW1 0x00000002 +#define EGPIOWR_GPW2 0x00000004 +#define EGPIOWR_GPW3 0x00000008 +#define EGPIOWR_GPW4 0x00000010 +#define EGPIOWR_GPW5 0x00000020 +#define EGPIOWR_GPW6 0x00000040 +#define EGPIOWR_GPW7 0x00000080 +#define EGPIOWR_GPW8 0x00000100 +#endif + +/* + * The following defines are for the flags in the extended GPIO pin status + * register. + */ +#ifndef NO_CS4612 +#define EGPIOSR_GPS0 0x00000001 +#define EGPIOSR_GPS1 0x00000002 +#define EGPIOSR_GPS2 0x00000004 +#define EGPIOSR_GPS3 0x00000008 +#define EGPIOSR_GPS4 0x00000010 +#define EGPIOSR_GPS5 0x00000020 +#define EGPIOSR_GPS6 0x00000040 +#define EGPIOSR_GPS7 0x00000080 +#define EGPIOSR_GPS8 0x00000100 +#endif + +/* + * The following defines are for the flags in the serial port 6 configuration + * register. + */ +#ifndef NO_CS4612 +#define SERC6_ASDO2EN 0x00000001 +#endif + +/* + * The following defines are for the flags in the serial port 7 configuration + * register. + */ +#ifndef NO_CS4612 +#define SERC7_ASDI2EN 0x00000001 +#define SERC7_POSILB 0x00000002 +#define SERC7_SIPOLB 0x00000004 +#define SERC7_SOSILB 0x00000008 +#define SERC7_SISOLB 0x00000010 +#endif + +/* + * The following defines are for the flags in the serial port AC link + * configuration register. + */ +#ifndef NO_CS4612 +#define SERACC_CHIP_TYPE_MASK 0x00000001 +#define SERACC_CHIP_TYPE_1_03 0x00000000 +#define SERACC_CHIP_TYPE_2_0 0x00000001 +#define SERACC_TWO_CODECS 0x00000002 +#define SERACC_MDM 0x00000004 +#define SERACC_HSP 0x00000008 +#define SERACC_ODT 0x00000010 /* only CS4630 */ +#endif + +/* + * The following defines are for the flags in the AC97 control register 2. + */ +#ifndef NO_CS4612 +#define ACCTL2_RSTN 0x00000001 +#define ACCTL2_ESYN 0x00000002 +#define ACCTL2_VFRM 0x00000004 +#define ACCTL2_DCV 0x00000008 +#define ACCTL2_CRW 0x00000010 +#define ACCTL2_ASYN 0x00000020 +#endif + +/* + * The following defines are for the flags in the AC97 status register 2. + */ +#ifndef NO_CS4612 +#define ACSTS2_CRDY 0x00000001 +#define ACSTS2_VSTS 0x00000002 +#endif + +/* + * The following defines are for the flags in the AC97 output slot valid + * register 2. + */ +#ifndef NO_CS4612 +#define ACOSV2_SLV3 0x00000001 +#define ACOSV2_SLV4 0x00000002 +#define ACOSV2_SLV5 0x00000004 +#define ACOSV2_SLV6 0x00000008 +#define ACOSV2_SLV7 0x00000010 +#define ACOSV2_SLV8 0x00000020 +#define ACOSV2_SLV9 0x00000040 +#define ACOSV2_SLV10 0x00000080 +#define ACOSV2_SLV11 0x00000100 +#define ACOSV2_SLV12 0x00000200 +#endif + +/* + * The following defines are for the flags in the AC97 command address + * register 2. + */ +#ifndef NO_CS4612 +#define ACCAD2_CI_MASK 0x0000007F +#define ACCAD2_CI_SHIFT 0 +#endif + +/* + * The following defines are for the flags in the AC97 command data register + * 2. + */ +#ifndef NO_CS4612 +#define ACCDA2_CD_MASK 0x0000FFFF +#define ACCDA2_CD_SHIFT 0 +#endif + +/* + * The following defines are for the flags in the AC97 input slot valid + * register 2. + */ +#ifndef NO_CS4612 +#define ACISV2_ISV3 0x00000001 +#define ACISV2_ISV4 0x00000002 +#define ACISV2_ISV5 0x00000004 +#define ACISV2_ISV6 0x00000008 +#define ACISV2_ISV7 0x00000010 +#define ACISV2_ISV8 0x00000020 +#define ACISV2_ISV9 0x00000040 +#define ACISV2_ISV10 0x00000080 +#define ACISV2_ISV11 0x00000100 +#define ACISV2_ISV12 0x00000200 +#endif + +/* + * The following defines are for the flags in the AC97 status address + * register 2. + */ +#ifndef NO_CS4612 +#define ACSAD2_SI_MASK 0x0000007F +#define ACSAD2_SI_SHIFT 0 +#endif + +/* + * The following defines are for the flags in the AC97 status data register 2. + */ +#ifndef NO_CS4612 +#define ACSDA2_SD_MASK 0x0000FFFF +#define ACSDA2_SD_SHIFT 0 +#endif + +/* + * The following defines are for the flags in the I/O trap address and control + * registers (all 12). + */ +#ifndef NO_CS4612 +#define IOTAC_SA_MASK 0x0000FFFF +#define IOTAC_MSK_MASK 0x000F0000 +#define IOTAC_IODC_MASK 0x06000000 +#define IOTAC_IODC_16_BIT 0x00000000 +#define IOTAC_IODC_10_BIT 0x02000000 +#define IOTAC_IODC_12_BIT 0x04000000 +#define IOTAC_WSPI 0x08000000 +#define IOTAC_RSPI 0x10000000 +#define IOTAC_WSE 0x20000000 +#define IOTAC_WE 0x40000000 +#define IOTAC_RE 0x80000000 +#define IOTAC_SA_SHIFT 0 +#define IOTAC_MSK_SHIFT 16 +#endif + +/* + * The following defines are for the flags in the I/O trap fast read registers + * (all 8). + */ +#ifndef NO_CS4612 +#define IOTFR_D_MASK 0x0000FFFF +#define IOTFR_A_MASK 0x000F0000 +#define IOTFR_R_MASK 0x0F000000 +#define IOTFR_ALL 0x40000000 +#define IOTFR_VL 0x80000000 +#define IOTFR_D_SHIFT 0 +#define IOTFR_A_SHIFT 16 +#define IOTFR_R_SHIFT 24 +#endif + +/* + * The following defines are for the flags in the I/O trap FIFO register. + */ +#ifndef NO_CS4612 +#define IOTFIFO_BA_MASK 0x00003FFF +#define IOTFIFO_S_MASK 0x00FF0000 +#define IOTFIFO_OF 0x40000000 +#define IOTFIFO_SPIOF 0x80000000 +#define IOTFIFO_BA_SHIFT 0 +#define IOTFIFO_S_SHIFT 16 +#endif + +/* + * The following defines are for the flags in the I/O trap retry read data + * register. + */ +#ifndef NO_CS4612 +#define IOTRRD_D_MASK 0x0000FFFF +#define IOTRRD_RDV 0x80000000 +#define IOTRRD_D_SHIFT 0 +#endif + +/* + * The following defines are for the flags in the I/O trap FIFO pointer + * register. + */ +#ifndef NO_CS4612 +#define IOTFP_CA_MASK 0x00003FFF +#define IOTFP_PA_MASK 0x3FFF0000 +#define IOTFP_CA_SHIFT 0 +#define IOTFP_PA_SHIFT 16 +#endif + +/* + * The following defines are for the flags in the I/O trap control register. + */ +#ifndef NO_CS4612 +#define IOTCR_ITD 0x00000001 +#define IOTCR_HRV 0x00000002 +#define IOTCR_SRV 0x00000004 +#define IOTCR_DTI 0x00000008 +#define IOTCR_DFI 0x00000010 +#define IOTCR_DDP 0x00000020 +#define IOTCR_JTE 0x00000040 +#define IOTCR_PPE 0x00000080 +#endif + +/* + * The following defines are for the flags in the direct PCI data register. + */ +#ifndef NO_CS4612 +#define DPCID_D_MASK 0xFFFFFFFF +#define DPCID_D_SHIFT 0 +#endif + +/* + * The following defines are for the flags in the direct PCI address register. + */ +#ifndef NO_CS4612 +#define DPCIA_A_MASK 0xFFFFFFFF +#define DPCIA_A_SHIFT 0 +#endif + +/* + * The following defines are for the flags in the direct PCI command register. + */ +#ifndef NO_CS4612 +#define DPCIC_C_MASK 0x0000000F +#define DPCIC_C_IOREAD 0x00000002 +#define DPCIC_C_IOWRITE 0x00000003 +#define DPCIC_BE_MASK 0x000000F0 +#endif + +/* + * The following defines are for the flags in the PC/PCI request register. + */ +#ifndef NO_CS4612 +#define PCPCIR_RDC_MASK 0x00000007 +#define PCPCIR_C_MASK 0x00007000 +#define PCPCIR_REQ 0x00008000 +#define PCPCIR_RDC_SHIFT 0 +#define PCPCIR_C_SHIFT 12 +#endif + +/* + * The following defines are for the flags in the PC/PCI grant register. + */ +#ifndef NO_CS4612 +#define PCPCIG_GDC_MASK 0x00000007 +#define PCPCIG_VL 0x00008000 +#define PCPCIG_GDC_SHIFT 0 +#endif + +/* + * The following defines are for the flags in the PC/PCI master enable + * register. + */ +#ifndef NO_CS4612 +#define PCPCIEN_EN 0x00000001 +#endif + +/* + * The following defines are for the flags in the extended PCI power + * management control register. + */ +#ifndef NO_CS4612 +#define EPCIPMC_GWU 0x00000001 +#define EPCIPMC_FSPC 0x00000002 +#endif + +/* + * The following defines are for the flags in the SP control register. + */ +#define SPCR_RUN 0x00000001 +#define SPCR_STPFR 0x00000002 +#define SPCR_RUNFR 0x00000004 +#define SPCR_TICK 0x00000008 +#define SPCR_DRQEN 0x00000020 +#define SPCR_RSTSP 0x00000040 +#define SPCR_OREN 0x00000080 +#ifndef NO_CS4612 +#define SPCR_PCIINT 0x00000100 +#define SPCR_OINTD 0x00000200 +#define SPCR_CRE 0x00008000 +#endif + +/* + * The following defines are for the flags in the debug index register. + */ +#define DREG_REGID_MASK 0x0000007F +#define DREG_DEBUG 0x00000080 +#define DREG_RGBK_MASK 0x00000700 +#define DREG_TRAP 0x00000800 +#if !defined(NO_CS4612) +#if !defined(NO_CS4615) +#define DREG_TRAPX 0x00001000 +#endif +#endif +#define DREG_REGID_SHIFT 0 +#define DREG_RGBK_SHIFT 8 +#define DREG_RGBK_REGID_MASK 0x0000077F +#define DREG_REGID_R0 0x00000010 +#define DREG_REGID_R1 0x00000011 +#define DREG_REGID_R2 0x00000012 +#define DREG_REGID_R3 0x00000013 +#define DREG_REGID_R4 0x00000014 +#define DREG_REGID_R5 0x00000015 +#define DREG_REGID_R6 0x00000016 +#define DREG_REGID_R7 0x00000017 +#define DREG_REGID_R8 0x00000018 +#define DREG_REGID_R9 0x00000019 +#define DREG_REGID_RA 0x0000001A +#define DREG_REGID_RB 0x0000001B +#define DREG_REGID_RC 0x0000001C +#define DREG_REGID_RD 0x0000001D +#define DREG_REGID_RE 0x0000001E +#define DREG_REGID_RF 0x0000001F +#define DREG_REGID_RA_BUS_LOW 0x00000020 +#define DREG_REGID_RA_BUS_HIGH 0x00000038 +#define DREG_REGID_YBUS_LOW 0x00000050 +#define DREG_REGID_YBUS_HIGH 0x00000058 +#define DREG_REGID_TRAP_0 0x00000100 +#define DREG_REGID_TRAP_1 0x00000101 +#define DREG_REGID_TRAP_2 0x00000102 +#define DREG_REGID_TRAP_3 0x00000103 +#define DREG_REGID_TRAP_4 0x00000104 +#define DREG_REGID_TRAP_5 0x00000105 +#define DREG_REGID_TRAP_6 0x00000106 +#define DREG_REGID_TRAP_7 0x00000107 +#define DREG_REGID_INDIRECT_ADDRESS 0x0000010E +#define DREG_REGID_TOP_OF_STACK 0x0000010F +#if !defined(NO_CS4612) +#if !defined(NO_CS4615) +#define DREG_REGID_TRAP_8 0x00000110 +#define DREG_REGID_TRAP_9 0x00000111 +#define DREG_REGID_TRAP_10 0x00000112 +#define DREG_REGID_TRAP_11 0x00000113 +#define DREG_REGID_TRAP_12 0x00000114 +#define DREG_REGID_TRAP_13 0x00000115 +#define DREG_REGID_TRAP_14 0x00000116 +#define DREG_REGID_TRAP_15 0x00000117 +#define DREG_REGID_TRAP_16 0x00000118 +#define DREG_REGID_TRAP_17 0x00000119 +#define DREG_REGID_TRAP_18 0x0000011A +#define DREG_REGID_TRAP_19 0x0000011B +#define DREG_REGID_TRAP_20 0x0000011C +#define DREG_REGID_TRAP_21 0x0000011D +#define DREG_REGID_TRAP_22 0x0000011E +#define DREG_REGID_TRAP_23 0x0000011F +#endif +#endif +#define DREG_REGID_RSA0_LOW 0x00000200 +#define DREG_REGID_RSA0_HIGH 0x00000201 +#define DREG_REGID_RSA1_LOW 0x00000202 +#define DREG_REGID_RSA1_HIGH 0x00000203 +#define DREG_REGID_RSA2 0x00000204 +#define DREG_REGID_RSA3 0x00000205 +#define DREG_REGID_RSI0_LOW 0x00000206 +#define DREG_REGID_RSI0_HIGH 0x00000207 +#define DREG_REGID_RSI1 0x00000208 +#define DREG_REGID_RSI2 0x00000209 +#define DREG_REGID_SAGUSTATUS 0x0000020A +#define DREG_REGID_RSCONFIG01_LOW 0x0000020B +#define DREG_REGID_RSCONFIG01_HIGH 0x0000020C +#define DREG_REGID_RSCONFIG23_LOW 0x0000020D +#define DREG_REGID_RSCONFIG23_HIGH 0x0000020E +#define DREG_REGID_RSDMA01E 0x0000020F +#define DREG_REGID_RSDMA23E 0x00000210 +#define DREG_REGID_RSD0_LOW 0x00000211 +#define DREG_REGID_RSD0_HIGH 0x00000212 +#define DREG_REGID_RSD1_LOW 0x00000213 +#define DREG_REGID_RSD1_HIGH 0x00000214 +#define DREG_REGID_RSD2_LOW 0x00000215 +#define DREG_REGID_RSD2_HIGH 0x00000216 +#define DREG_REGID_RSD3_LOW 0x00000217 +#define DREG_REGID_RSD3_HIGH 0x00000218 +#define DREG_REGID_SRAR_HIGH 0x0000021A +#define DREG_REGID_SRAR_LOW 0x0000021B +#define DREG_REGID_DMA_STATE 0x0000021C +#define DREG_REGID_CURRENT_DMA_STREAM 0x0000021D +#define DREG_REGID_NEXT_DMA_STREAM 0x0000021E +#define DREG_REGID_CPU_STATUS 0x00000300 +#define DREG_REGID_MAC_MODE 0x00000301 +#define DREG_REGID_STACK_AND_REPEAT 0x00000302 +#define DREG_REGID_INDEX0 0x00000304 +#define DREG_REGID_INDEX1 0x00000305 +#define DREG_REGID_DMA_STATE_0_3 0x00000400 +#define DREG_REGID_DMA_STATE_4_7 0x00000404 +#define DREG_REGID_DMA_STATE_8_11 0x00000408 +#define DREG_REGID_DMA_STATE_12_15 0x0000040C +#define DREG_REGID_DMA_STATE_16_19 0x00000410 +#define DREG_REGID_DMA_STATE_20_23 0x00000414 +#define DREG_REGID_DMA_STATE_24_27 0x00000418 +#define DREG_REGID_DMA_STATE_28_31 0x0000041C +#define DREG_REGID_DMA_STATE_32_35 0x00000420 +#define DREG_REGID_DMA_STATE_36_39 0x00000424 +#define DREG_REGID_DMA_STATE_40_43 0x00000428 +#define DREG_REGID_DMA_STATE_44_47 0x0000042C +#define DREG_REGID_DMA_STATE_48_51 0x00000430 +#define DREG_REGID_DMA_STATE_52_55 0x00000434 +#define DREG_REGID_DMA_STATE_56_59 0x00000438 +#define DREG_REGID_DMA_STATE_60_63 0x0000043C +#define DREG_REGID_DMA_STATE_64_67 0x00000440 +#define DREG_REGID_DMA_STATE_68_71 0x00000444 +#define DREG_REGID_DMA_STATE_72_75 0x00000448 +#define DREG_REGID_DMA_STATE_76_79 0x0000044C +#define DREG_REGID_DMA_STATE_80_83 0x00000450 +#define DREG_REGID_DMA_STATE_84_87 0x00000454 +#define DREG_REGID_DMA_STATE_88_91 0x00000458 +#define DREG_REGID_DMA_STATE_92_95 0x0000045C +#define DREG_REGID_TRAP_SELECT 0x00000500 +#define DREG_REGID_TRAP_WRITE_0 0x00000500 +#define DREG_REGID_TRAP_WRITE_1 0x00000501 +#define DREG_REGID_TRAP_WRITE_2 0x00000502 +#define DREG_REGID_TRAP_WRITE_3 0x00000503 +#define DREG_REGID_TRAP_WRITE_4 0x00000504 +#define DREG_REGID_TRAP_WRITE_5 0x00000505 +#define DREG_REGID_TRAP_WRITE_6 0x00000506 +#define DREG_REGID_TRAP_WRITE_7 0x00000507 +#if !defined(NO_CS4612) +#if !defined(NO_CS4615) +#define DREG_REGID_TRAP_WRITE_8 0x00000510 +#define DREG_REGID_TRAP_WRITE_9 0x00000511 +#define DREG_REGID_TRAP_WRITE_10 0x00000512 +#define DREG_REGID_TRAP_WRITE_11 0x00000513 +#define DREG_REGID_TRAP_WRITE_12 0x00000514 +#define DREG_REGID_TRAP_WRITE_13 0x00000515 +#define DREG_REGID_TRAP_WRITE_14 0x00000516 +#define DREG_REGID_TRAP_WRITE_15 0x00000517 +#define DREG_REGID_TRAP_WRITE_16 0x00000518 +#define DREG_REGID_TRAP_WRITE_17 0x00000519 +#define DREG_REGID_TRAP_WRITE_18 0x0000051A +#define DREG_REGID_TRAP_WRITE_19 0x0000051B +#define DREG_REGID_TRAP_WRITE_20 0x0000051C +#define DREG_REGID_TRAP_WRITE_21 0x0000051D +#define DREG_REGID_TRAP_WRITE_22 0x0000051E +#define DREG_REGID_TRAP_WRITE_23 0x0000051F +#endif +#endif +#define DREG_REGID_MAC0_ACC0_LOW 0x00000600 +#define DREG_REGID_MAC0_ACC1_LOW 0x00000601 +#define DREG_REGID_MAC0_ACC2_LOW 0x00000602 +#define DREG_REGID_MAC0_ACC3_LOW 0x00000603 +#define DREG_REGID_MAC1_ACC0_LOW 0x00000604 +#define DREG_REGID_MAC1_ACC1_LOW 0x00000605 +#define DREG_REGID_MAC1_ACC2_LOW 0x00000606 +#define DREG_REGID_MAC1_ACC3_LOW 0x00000607 +#define DREG_REGID_MAC0_ACC0_MID 0x00000608 +#define DREG_REGID_MAC0_ACC1_MID 0x00000609 +#define DREG_REGID_MAC0_ACC2_MID 0x0000060A +#define DREG_REGID_MAC0_ACC3_MID 0x0000060B +#define DREG_REGID_MAC1_ACC0_MID 0x0000060C +#define DREG_REGID_MAC1_ACC1_MID 0x0000060D +#define DREG_REGID_MAC1_ACC2_MID 0x0000060E +#define DREG_REGID_MAC1_ACC3_MID 0x0000060F +#define DREG_REGID_MAC0_ACC0_HIGH 0x00000610 +#define DREG_REGID_MAC0_ACC1_HIGH 0x00000611 +#define DREG_REGID_MAC0_ACC2_HIGH 0x00000612 +#define DREG_REGID_MAC0_ACC3_HIGH 0x00000613 +#define DREG_REGID_MAC1_ACC0_HIGH 0x00000614 +#define DREG_REGID_MAC1_ACC1_HIGH 0x00000615 +#define DREG_REGID_MAC1_ACC2_HIGH 0x00000616 +#define DREG_REGID_MAC1_ACC3_HIGH 0x00000617 +#define DREG_REGID_RSHOUT_LOW 0x00000620 +#define DREG_REGID_RSHOUT_MID 0x00000628 +#define DREG_REGID_RSHOUT_HIGH 0x00000630 + +/* + * The following defines are for the flags in the DMA stream requestor write + */ +#define DSRWP_DSR_MASK 0x0000000F +#define DSRWP_DSR_BG_RQ 0x00000001 +#define DSRWP_DSR_PRIORITY_MASK 0x00000006 +#define DSRWP_DSR_PRIORITY_0 0x00000000 +#define DSRWP_DSR_PRIORITY_1 0x00000002 +#define DSRWP_DSR_PRIORITY_2 0x00000004 +#define DSRWP_DSR_PRIORITY_3 0x00000006 +#define DSRWP_DSR_RQ_PENDING 0x00000008 + +/* + * The following defines are for the flags in the trap write port register. + */ +#define TWPR_TW_MASK 0x0000FFFF +#define TWPR_TW_SHIFT 0 + +/* + * The following defines are for the flags in the stack pointer write + * register. + */ +#define SPWR_STKP_MASK 0x0000000F +#define SPWR_STKP_SHIFT 0 + +/* + * The following defines are for the flags in the SP interrupt register. + */ +#define SPIR_FRI 0x00000001 +#define SPIR_DOI 0x00000002 +#define SPIR_GPI2 0x00000004 +#define SPIR_GPI3 0x00000008 +#define SPIR_IP0 0x00000010 +#define SPIR_IP1 0x00000020 +#define SPIR_IP2 0x00000040 +#define SPIR_IP3 0x00000080 + +/* + * The following defines are for the flags in the functional group 1 register. + */ +#define FGR1_F1S_MASK 0x0000FFFF +#define FGR1_F1S_SHIFT 0 + +/* + * The following defines are for the flags in the SP clock status register. + */ +#define SPCS_FRI 0x00000001 +#define SPCS_DOI 0x00000002 +#define SPCS_GPI2 0x00000004 +#define SPCS_GPI3 0x00000008 +#define SPCS_IP0 0x00000010 +#define SPCS_IP1 0x00000020 +#define SPCS_IP2 0x00000040 +#define SPCS_IP3 0x00000080 +#define SPCS_SPRUN 0x00000100 +#define SPCS_SLEEP 0x00000200 +#define SPCS_FG 0x00000400 +#define SPCS_ORUN 0x00000800 +#define SPCS_IRQ 0x00001000 +#define SPCS_FGN_MASK 0x0000E000 +#define SPCS_FGN_SHIFT 13 + +/* + * The following defines are for the flags in the SP DMA requestor status + * register. + */ +#define SDSR_DCS_MASK 0x000000FF +#define SDSR_DCS_SHIFT 0 +#define SDSR_DCS_NONE 0x00000007 + +/* + * The following defines are for the flags in the frame timer register. + */ +#define FRMT_FTV_MASK 0x0000FFFF +#define FRMT_FTV_SHIFT 0 + +/* + * The following defines are for the flags in the frame timer current count + * register. + */ +#define FRCC_FCC_MASK 0x0000FFFF +#define FRCC_FCC_SHIFT 0 + +/* + * The following defines are for the flags in the frame timer save count + * register. + */ +#define FRSC_FCS_MASK 0x0000FFFF +#define FRSC_FCS_SHIFT 0 + +/* + * The following define the various flags stored in the scatter/gather + * descriptors. + */ +#define DMA_SG_NEXT_ENTRY_MASK 0x00000FF8 +#define DMA_SG_SAMPLE_END_MASK 0x0FFF0000 +#define DMA_SG_SAMPLE_END_FLAG 0x10000000 +#define DMA_SG_LOOP_END_FLAG 0x20000000 +#define DMA_SG_SIGNAL_END_FLAG 0x40000000 +#define DMA_SG_SIGNAL_PAGE_FLAG 0x80000000 +#define DMA_SG_NEXT_ENTRY_SHIFT 3 +#define DMA_SG_SAMPLE_END_SHIFT 16 + +/* + * The following define the offsets of the fields within the on-chip generic + * DMA requestor. + */ +#define DMA_RQ_CONTROL1 0x00000000 +#define DMA_RQ_CONTROL2 0x00000004 +#define DMA_RQ_SOURCE_ADDR 0x00000008 +#define DMA_RQ_DESTINATION_ADDR 0x0000000C +#define DMA_RQ_NEXT_PAGE_ADDR 0x00000010 +#define DMA_RQ_NEXT_PAGE_SGDESC 0x00000014 +#define DMA_RQ_LOOP_START_ADDR 0x00000018 +#define DMA_RQ_POST_LOOP_ADDR 0x0000001C +#define DMA_RQ_PAGE_MAP_ADDR 0x00000020 + +/* + * The following defines are for the flags in the first control word of the + * on-chip generic DMA requestor. + */ +#define DMA_RQ_C1_COUNT_MASK 0x000003FF +#define DMA_RQ_C1_DESTINATION_SCATTER 0x00001000 +#define DMA_RQ_C1_SOURCE_GATHER 0x00002000 +#define DMA_RQ_C1_DONE_FLAG 0x00004000 +#define DMA_RQ_C1_OPTIMIZE_STATE 0x00008000 +#define DMA_RQ_C1_SAMPLE_END_STATE_MASK 0x00030000 +#define DMA_RQ_C1_FULL_PAGE 0x00000000 +#define DMA_RQ_C1_BEFORE_SAMPLE_END 0x00010000 +#define DMA_RQ_C1_PAGE_MAP_ERROR 0x00020000 +#define DMA_RQ_C1_AT_SAMPLE_END 0x00030000 +#define DMA_RQ_C1_LOOP_END_STATE_MASK 0x000C0000 +#define DMA_RQ_C1_NOT_LOOP_END 0x00000000 +#define DMA_RQ_C1_BEFORE_LOOP_END 0x00040000 +#define DMA_RQ_C1_2PAGE_LOOP_BEGIN 0x00080000 +#define DMA_RQ_C1_LOOP_BEGIN 0x000C0000 +#define DMA_RQ_C1_PAGE_MAP_MASK 0x00300000 +#define DMA_RQ_C1_PM_NONE_PENDING 0x00000000 +#define DMA_RQ_C1_PM_NEXT_PENDING 0x00100000 +#define DMA_RQ_C1_PM_RESERVED 0x00200000 +#define DMA_RQ_C1_PM_LOOP_NEXT_PENDING 0x00300000 +#define DMA_RQ_C1_WRITEBACK_DEST_FLAG 0x00400000 +#define DMA_RQ_C1_WRITEBACK_SRC_FLAG 0x00800000 +#define DMA_RQ_C1_DEST_SIZE_MASK 0x07000000 +#define DMA_RQ_C1_DEST_LINEAR 0x00000000 +#define DMA_RQ_C1_DEST_MOD16 0x01000000 +#define DMA_RQ_C1_DEST_MOD32 0x02000000 +#define DMA_RQ_C1_DEST_MOD64 0x03000000 +#define DMA_RQ_C1_DEST_MOD128 0x04000000 +#define DMA_RQ_C1_DEST_MOD256 0x05000000 +#define DMA_RQ_C1_DEST_MOD512 0x06000000 +#define DMA_RQ_C1_DEST_MOD1024 0x07000000 +#define DMA_RQ_C1_DEST_ON_HOST 0x08000000 +#define DMA_RQ_C1_SOURCE_SIZE_MASK 0x70000000 +#define DMA_RQ_C1_SOURCE_LINEAR 0x00000000 +#define DMA_RQ_C1_SOURCE_MOD16 0x10000000 +#define DMA_RQ_C1_SOURCE_MOD32 0x20000000 +#define DMA_RQ_C1_SOURCE_MOD64 0x30000000 +#define DMA_RQ_C1_SOURCE_MOD128 0x40000000 +#define DMA_RQ_C1_SOURCE_MOD256 0x50000000 +#define DMA_RQ_C1_SOURCE_MOD512 0x60000000 +#define DMA_RQ_C1_SOURCE_MOD1024 0x70000000 +#define DMA_RQ_C1_SOURCE_ON_HOST 0x80000000 +#define DMA_RQ_C1_COUNT_SHIFT 0 + +/* + * The following defines are for the flags in the second control word of the + * on-chip generic DMA requestor. + */ +#define DMA_RQ_C2_VIRTUAL_CHANNEL_MASK 0x0000003F +#define DMA_RQ_C2_VIRTUAL_SIGNAL_MASK 0x00000300 +#define DMA_RQ_C2_NO_VIRTUAL_SIGNAL 0x00000000 +#define DMA_RQ_C2_SIGNAL_EVERY_DMA 0x00000100 +#define DMA_RQ_C2_SIGNAL_SOURCE_PINGPONG 0x00000200 +#define DMA_RQ_C2_SIGNAL_DEST_PINGPONG 0x00000300 +#define DMA_RQ_C2_AUDIO_CONVERT_MASK 0x0000F000 +#define DMA_RQ_C2_AC_NONE 0x00000000 +#define DMA_RQ_C2_AC_8_TO_16_BIT 0x00001000 +#define DMA_RQ_C2_AC_MONO_TO_STEREO 0x00002000 +#define DMA_RQ_C2_AC_ENDIAN_CONVERT 0x00004000 +#define DMA_RQ_C2_AC_SIGNED_CONVERT 0x00008000 +#define DMA_RQ_C2_LOOP_END_MASK 0x0FFF0000 +#define DMA_RQ_C2_LOOP_MASK 0x30000000 +#define DMA_RQ_C2_NO_LOOP 0x00000000 +#define DMA_RQ_C2_ONE_PAGE_LOOP 0x10000000 +#define DMA_RQ_C2_TWO_PAGE_LOOP 0x20000000 +#define DMA_RQ_C2_MULTI_PAGE_LOOP 0x30000000 +#define DMA_RQ_C2_SIGNAL_LOOP_BACK 0x40000000 +#define DMA_RQ_C2_SIGNAL_POST_BEGIN_PAGE 0x80000000 +#define DMA_RQ_C2_VIRTUAL_CHANNEL_SHIFT 0 +#define DMA_RQ_C2_LOOP_END_SHIFT 16 + +/* + * The following defines are for the flags in the source and destination words + * of the on-chip generic DMA requestor. + */ +#define DMA_RQ_SD_ADDRESS_MASK 0x0000FFFF +#define DMA_RQ_SD_MEMORY_ID_MASK 0x000F0000 +#define DMA_RQ_SD_SP_PARAM_ADDR 0x00000000 +#define DMA_RQ_SD_SP_SAMPLE_ADDR 0x00010000 +#define DMA_RQ_SD_SP_PROGRAM_ADDR 0x00020000 +#define DMA_RQ_SD_SP_DEBUG_ADDR 0x00030000 +#define DMA_RQ_SD_OMNIMEM_ADDR 0x000E0000 +#define DMA_RQ_SD_END_FLAG 0x40000000 +#define DMA_RQ_SD_ERROR_FLAG 0x80000000 +#define DMA_RQ_SD_ADDRESS_SHIFT 0 + +/* + * The following defines are for the flags in the page map address word of the + * on-chip generic DMA requestor. + */ +#define DMA_RQ_PMA_LOOP_THIRD_PAGE_ENTRY_MASK 0x00000FF8 +#define DMA_RQ_PMA_PAGE_TABLE_MASK 0xFFFFF000 +#define DMA_RQ_PMA_LOOP_THIRD_PAGE_ENTRY_SHIFT 3 +#define DMA_RQ_PMA_PAGE_TABLE_SHIFT 12 + +#define BA1_VARIDEC_BUF_1 0x000 + +#define BA1_PDTC 0x0c0 /* BA1_PLAY_DMA_TRANSACTION_COUNT_REG */ +#define BA1_PFIE 0x0c4 /* BA1_PLAY_FORMAT_&_INTERRUPT_ENABLE_REG */ +#define BA1_PBA 0x0c8 /* BA1_PLAY_BUFFER_ADDRESS */ +#define BA1_PVOL 0x0f8 /* BA1_PLAY_VOLUME_REG */ +#define BA1_PSRC 0x288 /* BA1_PLAY_SAMPLE_RATE_CORRECTION_REG */ +#define BA1_PCTL 0x2a4 /* BA1_PLAY_CONTROL_REG */ +#define BA1_PPI 0x2b4 /* BA1_PLAY_PHASE_INCREMENT_REG */ + +#define BA1_CCTL 0x064 /* BA1_CAPTURE_CONTROL_REG */ +#define BA1_CIE 0x104 /* BA1_CAPTURE_INTERRUPT_ENABLE_REG */ +#define BA1_CBA 0x10c /* BA1_CAPTURE_BUFFER_ADDRESS */ +#define BA1_CSRC 0x2c8 /* BA1_CAPTURE_SAMPLE_RATE_CORRECTION_REG */ +#define BA1_CCI 0x2d8 /* BA1_CAPTURE_COEFFICIENT_INCREMENT_REG */ +#define BA1_CD 0x2e0 /* BA1_CAPTURE_DELAY_REG */ +#define BA1_CPI 0x2f4 /* BA1_CAPTURE_PHASE_INCREMENT_REG */ +#define BA1_CVOL 0x2f8 /* BA1_CAPTURE_VOLUME_REG */ + +#define BA1_CFG1 0x134 /* BA1_CAPTURE_FRAME_GROUP_1_REG */ +#define BA1_CFG2 0x138 /* BA1_CAPTURE_FRAME_GROUP_2_REG */ +#define BA1_CCST 0x13c /* BA1_CAPTURE_CONSTANT_REG */ +#define BA1_CSPB 0x340 /* BA1_CAPTURE_SPB_ADDRESS */ + +/* + * + */ + +#define CS46XX_MODE_OUTPUT (1<<0) /* MIDI UART - output */ +#define CS46XX_MODE_INPUT (1<<1) /* MIDI UART - input */ + +/* + * + */ + +#define SAVE_REG_MAX 0x10 +#define POWER_DOWN_ALL 0x7f0f + +/* maxinum number of AC97 codecs connected, AC97 2.0 defined 4 */ +#define MAX_NR_AC97 4 +#define CS46XX_PRIMARY_CODEC_INDEX 0 +#define CS46XX_SECONDARY_CODEC_INDEX 1 +#define CS46XX_SECONDARY_CODEC_OFFSET 0x80 +#define CS46XX_DSP_CAPTURE_CHANNEL 1 + +/* capture */ +#define CS46XX_DSP_CAPTURE_CHANNEL 1 + +/* mixer */ +#define CS46XX_MIXER_SPDIF_INPUT_ELEMENT 1 +#define CS46XX_MIXER_SPDIF_OUTPUT_ELEMENT 2 + + +struct snd_cs46xx_pcm { + struct snd_dma_buffer hw_buf; + + unsigned int ctl; + unsigned int shift; /* Shift count to trasform frames in bytes */ + struct snd_pcm_indirect pcm_rec; + struct snd_pcm_substream *substream; + + struct dsp_pcm_channel_descriptor * pcm_channel; + + int pcm_channel_id; /* Fron Rear, Center Lfe ... */ +}; + +struct snd_cs46xx_region { + char name[24]; + unsigned long base; + void __iomem *remap_addr; + unsigned long size; + struct resource *resource; +}; + +struct snd_cs46xx { + int irq; + unsigned long ba0_addr; + unsigned long ba1_addr; + union { + struct { + struct snd_cs46xx_region ba0; + struct snd_cs46xx_region data0; + struct snd_cs46xx_region data1; + struct snd_cs46xx_region pmem; + struct snd_cs46xx_region reg; + } name; + struct snd_cs46xx_region idx[5]; + } region; + + unsigned int mode; + + struct { + struct snd_dma_buffer hw_buf; + + unsigned int ctl; + unsigned int shift; /* Shift count to trasform frames in bytes */ + struct snd_pcm_indirect pcm_rec; + struct snd_pcm_substream *substream; + } capt; + + + int nr_ac97_codecs; + struct snd_ac97_bus *ac97_bus; + struct snd_ac97 *ac97[MAX_NR_AC97]; + + struct pci_dev *pci; + struct snd_card *card; + struct snd_pcm *pcm; + + struct snd_rawmidi *rmidi; + struct snd_rawmidi_substream *midi_input; + struct snd_rawmidi_substream *midi_output; + + spinlock_t reg_lock; + unsigned int midcr; + unsigned int uartm; + + int amplifier; + void (*amplifier_ctrl)(struct snd_cs46xx *, int); + void (*active_ctrl)(struct snd_cs46xx *, int); + void (*mixer_init)(struct snd_cs46xx *); + + int acpi_port; + struct snd_kcontrol *eapd_switch; /* for amplifier hack */ + int accept_valid; /* accept mmap valid (for OSS) */ + int in_suspend; + + struct gameport *gameport; + +#ifdef CONFIG_SND_CS46XX_NEW_DSP + struct mutex spos_mutex; + + struct dsp_spos_instance * dsp_spos_instance; + + struct snd_pcm *pcm_rear; + struct snd_pcm *pcm_center_lfe; + struct snd_pcm *pcm_iec958; +#else /* for compatibility */ + struct snd_cs46xx_pcm *playback_pcm; + unsigned int play_ctl; +#endif + +#ifdef CONFIG_PM + u32 *saved_regs; +#endif +}; + +int snd_cs46xx_create(struct snd_card *card, + struct pci_dev *pci, + int external_amp, int thinkpad, + struct snd_cs46xx **rcodec); +extern const struct dev_pm_ops snd_cs46xx_pm; + +int snd_cs46xx_pcm(struct snd_cs46xx *chip, int device, struct snd_pcm **rpcm); +int snd_cs46xx_pcm_rear(struct snd_cs46xx *chip, int device, struct snd_pcm **rpcm); +int snd_cs46xx_pcm_iec958(struct snd_cs46xx *chip, int device, struct snd_pcm **rpcm); +int snd_cs46xx_pcm_center_lfe(struct snd_cs46xx *chip, int device, struct snd_pcm **rpcm); +int snd_cs46xx_mixer(struct snd_cs46xx *chip, int spdif_device); +int snd_cs46xx_midi(struct snd_cs46xx *chip, int device, struct snd_rawmidi **rmidi); +int snd_cs46xx_start_dsp(struct snd_cs46xx *chip); +int snd_cs46xx_gameport(struct snd_cs46xx *chip); + +#endif /* __SOUND_CS46XX_H */ diff --git a/sound/pci/cs46xx/cs46xx_dsp_scb_types.h b/sound/pci/cs46xx/cs46xx_dsp_scb_types.h new file mode 100644 index 000000000000..080857ad0ca2 --- /dev/null +++ b/sound/pci/cs46xx/cs46xx_dsp_scb_types.h @@ -0,0 +1,1213 @@ +/* + * The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + * + * NOTE: comments are copy/paste from cwcemb80.lst + * provided by Tom Woller at Cirrus (my only + * documentation about the SP OS running inside + * the DSP) + */ + +#ifndef __CS46XX_DSP_SCB_TYPES_H__ +#define __CS46XX_DSP_SCB_TYPES_H__ + +#include <asm/byteorder.h> + +#ifndef ___DSP_DUAL_16BIT_ALLOC +#if defined(__LITTLE_ENDIAN) +#define ___DSP_DUAL_16BIT_ALLOC(a,b) u16 a; u16 b; +#elif defined(__BIG_ENDIAN) +#define ___DSP_DUAL_16BIT_ALLOC(a,b) u16 b; u16 a; +#else +#error Not __LITTLE_ENDIAN and not __BIG_ENDIAN, then what ??? +#endif +#endif + +/* This structs are used internally by the SP */ + +struct dsp_basic_dma_req { + /* DMA Requestor Word 0 (DCW) fields: + + 31 [30-28]27 [26:24] 23 22 21 20 [19:18] [17:16] 15 14 13 12 11 10 9 8 7 6 [5:0] + _______________________________________________________________________________________ + |S| SBT |D| DBT |wb|wb| | | LS | SS |Opt|Do|SSG|DSG| | | | | | | Dword | + |H|_____ |H|_________|S_|D |__|__|______|_______|___|ne|__ |__ |__|__|_|_|_|_|_Count -1| + */ + u32 dcw; /* DMA Control Word */ + u32 dmw; /* DMA Mode Word */ + u32 saw; /* Source Address Word */ + u32 daw; /* Destination Address Word */ +}; + +struct dsp_scatter_gather_ext { + u32 npaw; /* Next-Page Address Word */ + + /* DMA Requestor Word 5 (NPCW) fields: + + 31-30 29 28 [27:16] [15:12] [11:3] [2:0] + _________________________________________________________________________________________ + |SV |LE|SE| Sample-end byte offset | | Page-map entry offset for next | | + |page|__|__| ___________________________|_________|__page, if !sample-end___________|____| + */ + u32 npcw; /* Next-Page Control Word */ + u32 lbaw; /* Loop-Begin Address Word */ + u32 nplbaw; /* Next-Page after Loop-Begin Address Word */ + u32 sgaw; /* Scatter/Gather Address Word */ +}; + +struct dsp_volume_control { + ___DSP_DUAL_16BIT_ALLOC( + rightTarg, /* Target volume for left & right channels */ + leftTarg + ) + ___DSP_DUAL_16BIT_ALLOC( + rightVol, /* Current left & right channel volumes */ + leftVol + ) +}; + +/* Generic stream control block (SCB) structure definition */ +struct dsp_generic_scb { + /* For streaming I/O, the DSP should never alter any words in the DMA + requestor or the scatter/gather extension. Only ad hoc DMA request + streams are free to alter the requestor (currently only occur in the + DOS-based MIDI controller and in debugger-inserted code). + + If an SCB does not have any associated DMA requestor, these 9 ints + may be freed for use by other tasks, but the pointer to the SCB must + still be such that the insOrd:nextSCB appear at offset 9 from the + SCB pointer. + + Basic (non scatter/gather) DMA requestor (4 ints) + */ + + /* Initialized by the host, only modified by DMA + R/O for the DSP task */ + struct dsp_basic_dma_req basic_req; /* Optional */ + + /* Scatter/gather DMA requestor extension (5 ints) + Initialized by the host, only modified by DMA + DSP task never needs to even read these. + */ + struct dsp_scatter_gather_ext sg_ext; /* Optional */ + + /* Sublist pointer & next stream control block (SCB) link. + Initialized & modified by the host R/O for the DSP task + */ + ___DSP_DUAL_16BIT_ALLOC( + next_scb, /* REQUIRED */ + sub_list_ptr /* REQUIRED */ + ) + + /* Pointer to this tasks parameter block & stream function pointer + Initialized by the host R/O for the DSP task */ + ___DSP_DUAL_16BIT_ALLOC( + entry_point, /* REQUIRED */ + this_spb /* REQUIRED */ + ) + + /* rsConfig register for stream buffer (rsDMA reg. + is loaded from basicReq.daw for incoming streams, or + basicReq.saw, for outgoing streams) + + 31 30 29 [28:24] [23:16] 15 14 13 12 11 10 9 8 7 6 5 4 [3:0] + ______________________________________________________________________________ + |DMA |D|maxDMAsize| streamNum|dir|p| | | | | | |ds |shr 1|rev Cy | mod | + |prio |_|__________|__________|___|_|__|__|__|__|_|_|___|_____|_______|_______| + 31 30 29 [28:24] [23:16] 15 14 13 12 11 10 9 8 7 6 5 4 [3:0] + + + Initialized by the host R/O for the DSP task + */ + u32 strm_rs_config; /* REQUIRED */ + // + /* On mixer input streams: indicates mixer input stream configuration + On Tees, this is copied from the stream being snooped + + Stream sample pointer & MAC-unit mode for this stream + + Initialized by the host Updated by the DSP task + */ + u32 strm_buf_ptr; /* REQUIRED */ + + /* On mixer input streams: points to next mixer input and is updated by the + mixer subroutine in the "parent" DSP task + (least-significant 16 bits are preserved, unused) + + On Tees, the pointer is copied from the stream being snooped on + initialization, and, subsequently, it is copied into the + stream being snooped. + + On wavetable/3D voices: the strmBufPtr will use all 32 bits to allow for + fractional phase accumulation + + Fractional increment per output sample in the input sample buffer + + (Not used on mixer input streams & redefined on Tees) + On wavetable/3D voices: this 32-bit word specifies the integer.fractional + increment per output sample. + */ + u32 strmPhiIncr; + + + /* Standard stereo volume control + Initialized by the host (host updates target volumes) + + Current volumes update by the DSP task + On mixer input streams: required & updated by the mixer subroutine in the + "parent" DSP task + + On Tees, both current & target volumes are copied up on initialization, + and, subsequently, the target volume is copied up while the current + volume is copied down. + + These two 32-bit words are redefined for wavetable & 3-D voices. + */ + struct dsp_volume_control vol_ctrl_t; /* Optional */ +}; + + +struct dsp_spos_control_block { + /* WARNING: Certain items in this structure are modified by the host + Any dword that can be modified by the host, must not be + modified by the SP as the host can only do atomic dword + writes, and to do otherwise, even a read modify write, + may lead to corrupted data on the SP. + + This rule does not apply to one off boot time initialisation prior to starting the SP + */ + + + ___DSP_DUAL_16BIT_ALLOC( + /* First element on the Hyper forground task tree */ + hfg_tree_root_ptr, /* HOST */ + /* First 3 dwords are written by the host and read-only on the DSP */ + hfg_stack_base /* HOST */ + ) + + ___DSP_DUAL_16BIT_ALLOC( + /* Point to this data structure to enable easy access */ + spos_cb_ptr, /* SP */ + prev_task_tree_ptr /* SP && HOST */ + ) + + ___DSP_DUAL_16BIT_ALLOC( + /* Currently Unused */ + xxinterval_timer_period, + /* Enable extension of SPOS data structure */ + HFGSPB_ptr + ) + + + ___DSP_DUAL_16BIT_ALLOC( + xxnum_HFG_ticks_thisInterval, + /* Modified by the DSP */ + xxnum_tntervals + ) + + + /* Set by DSP upon encountering a trap (breakpoint) or a spurious + interrupt. The host must clear this dword after reading it + upon receiving spInt1. */ + ___DSP_DUAL_16BIT_ALLOC( + spurious_int_flag, /* (Host & SP) Nature of the spurious interrupt */ + trap_flag /* (Host & SP) Nature of detected Trap */ + ) + + ___DSP_DUAL_16BIT_ALLOC( + unused2, + invalid_IP_flag /* (Host & SP ) Indicate detection of invalid instruction pointer */ + ) + + ___DSP_DUAL_16BIT_ALLOC( + /* pointer to forground task tree header for use in next task search */ + fg_task_tree_hdr_ptr, /* HOST */ + /* Data structure for controlling synchronous link update */ + hfg_sync_update_ptr /* HOST */ + ) + + ___DSP_DUAL_16BIT_ALLOC( + begin_foreground_FCNT, /* SP */ + /* Place holder for holding sleep timing */ + last_FCNT_before_sleep /* SP */ + ) + + ___DSP_DUAL_16BIT_ALLOC( + unused7, /* SP */ + next_task_treePtr /* SP */ + ) + + u32 unused5; + + ___DSP_DUAL_16BIT_ALLOC( + active_flags, /* SP */ + /* State flags, used to assist control of execution of Hyper Forground */ + HFG_flags /* SP */ + ) + + ___DSP_DUAL_16BIT_ALLOC( + unused9, + unused8 + ) + + /* Space for saving enough context so that we can set up enough + to save some more context. + */ + u32 rFE_save_for_invalid_IP; + u32 r32_save_for_spurious_int; + u32 r32_save_for_trap; + u32 r32_save_for_HFG; +}; + +/* SPB for MIX_TO_OSTREAM algorithm family */ +struct dsp_mix2_ostream_spb +{ + /* 16b.16b integer.frac approximation to the + number of 3 sample triplets to output each + frame. (approximation must be floor, to + insure that the fractional error is always + positive) + */ + u32 outTripletsPerFrame; + + /* 16b.16b integer.frac accumulated number of + output triplets since the start of group + */ + u32 accumOutTriplets; +}; + +/* SCB for Timing master algorithm */ +struct dsp_timing_master_scb { + /* First 12 dwords from generic_scb_t */ + struct dsp_basic_dma_req basic_req; /* Optional */ + struct dsp_scatter_gather_ext sg_ext; /* Optional */ + ___DSP_DUAL_16BIT_ALLOC( + next_scb, /* REQUIRED */ + sub_list_ptr /* REQUIRED */ + ) + + ___DSP_DUAL_16BIT_ALLOC( + entry_point, /* REQUIRED */ + this_spb /* REQUIRED */ + ) + + ___DSP_DUAL_16BIT_ALLOC( + /* Initial values are 0000:xxxx */ + reserved, + extra_sample_accum + ) + + + /* Initial values are xxxx:0000 + hi: Current CODEC output FIFO pointer + (0 to 0x0f) + lo: Flag indicating that the CODEC + FIFO is sync'd (host clears to + resynchronize the FIFO pointer + upon start/restart) + */ + ___DSP_DUAL_16BIT_ALLOC( + codec_FIFO_syncd, + codec_FIFO_ptr + ) + + /* Init. 8000:0005 for 44.1k + 8000:0001 for 48k + hi: Fractional sample accumulator 0.16b + lo: Number of frames remaining to be + processed in the current group of + frames + */ + ___DSP_DUAL_16BIT_ALLOC( + frac_samp_accum_qm1, + TM_frms_left_in_group + ) + + /* Init. 0001:0005 for 44.1k + 0000:0001 for 48k + hi: Fractional sample correction factor 0.16b + to be added every frameGroupLength frames + to correct for truncation error in + nsamp_per_frm_q15 + lo: Number of frames in the group + */ + ___DSP_DUAL_16BIT_ALLOC( + frac_samp_correction_qm1, + TM_frm_group_length + ) + + /* Init. 44.1k*65536/8k = 0x00058333 for 44.1k + 48k*65536/8k = 0x00060000 for 48k + 16b.16b integer.frac approximation to the + number of samples to output each frame. + (approximation must be floor, to insure */ + u32 nsamp_per_frm_q15; +}; + +/* SCB for CODEC output algorithm */ +struct dsp_codec_output_scb { + /* First 13 dwords from generic_scb_t */ + struct dsp_basic_dma_req basic_req; /* Optional */ + struct dsp_scatter_gather_ext sg_ext; /* Optional */ + ___DSP_DUAL_16BIT_ALLOC( + next_scb, /* REQUIRED */ + sub_list_ptr /* REQUIRED */ + ) + + ___DSP_DUAL_16BIT_ALLOC( + entry_point, /* REQUIRED */ + this_spb /* REQUIRED */ + ) + + u32 strm_rs_config; /* REQUIRED */ + + u32 strm_buf_ptr; /* REQUIRED */ + + /* NOTE: The CODEC output task reads samples from the first task on its + sublist at the stream buffer pointer (init. to lag DMA destination + address word). After the required number of samples is transferred, + the CODEC output task advances sub_list_ptr->strm_buf_ptr past the samples + consumed. + */ + + /* Init. 0000:0010 for SDout + 0060:0010 for SDout2 + 0080:0010 for SDout3 + hi: Base IO address of FIFO to which + the left-channel samples are to + be written. + lo: Displacement for the base IO + address for left-channel to obtain + the base IO address for the FIFO + to which the right-channel samples + are to be written. + */ + ___DSP_DUAL_16BIT_ALLOC( + left_chan_base_IO_addr, + right_chan_IO_disp + ) + + + /* Init: 0x0080:0004 for non-AC-97 + Init: 0x0080:0000 for AC-97 + hi: Exponential volume change rate + for input stream + lo: Positive shift count to shift the + 16-bit input sample to obtain the + 32-bit output word + */ + ___DSP_DUAL_16BIT_ALLOC( + CO_scale_shift_count, + CO_exp_vol_change_rate + ) + + /* Pointer to SCB at end of input chain */ + ___DSP_DUAL_16BIT_ALLOC( + reserved, + last_sub_ptr + ) +}; + +/* SCB for CODEC input algorithm */ +struct dsp_codec_input_scb { + /* First 13 dwords from generic_scb_t */ + struct dsp_basic_dma_req basic_req; /* Optional */ + struct dsp_scatter_gather_ext sg_ext; /* Optional */ + ___DSP_DUAL_16BIT_ALLOC( + next_scb, /* REQUIRED */ + sub_list_ptr /* REQUIRED */ + ) + + ___DSP_DUAL_16BIT_ALLOC( + entry_point, /* REQUIRED */ + this_spb /* REQUIRED */ + ) + + u32 strm_rs_config; /* REQUIRED */ + u32 strm_buf_ptr; /* REQUIRED */ + + /* NOTE: The CODEC input task reads samples from the hardware FIFO + sublist at the DMA source address word (sub_list_ptr->basic_req.saw). + After the required number of samples is transferred, the CODEC + output task advances sub_list_ptr->basic_req.saw past the samples + consumed. SPuD must initialize the sub_list_ptr->basic_req.saw + to point half-way around from the initial sub_list_ptr->strm_nuf_ptr + to allow for lag/lead. + */ + + /* Init. 0000:0010 for SDout + 0060:0010 for SDout2 + 0080:0010 for SDout3 + hi: Base IO address of FIFO to which + the left-channel samples are to + be written. + lo: Displacement for the base IO + address for left-channel to obtain + the base IO address for the FIFO + to which the right-channel samples + are to be written. + */ + ___DSP_DUAL_16BIT_ALLOC( + rightChanINdisp, + left_chan_base_IN_addr + ) + /* Init. ?:fffc + lo: Negative shift count to shift the + 32-bit input dword to obtain the + 16-bit sample msb-aligned (count + is negative to shift left) + */ + ___DSP_DUAL_16BIT_ALLOC( + scaleShiftCount, + reserver1 + ) + + u32 reserved2; +}; + + +struct dsp_pcm_serial_input_scb { + /* First 13 dwords from generic_scb_t */ + struct dsp_basic_dma_req basic_req; /* Optional */ + struct dsp_scatter_gather_ext sg_ext; /* Optional */ + ___DSP_DUAL_16BIT_ALLOC( + next_scb, /* REQUIRED */ + sub_list_ptr /* REQUIRED */ + ) + + ___DSP_DUAL_16BIT_ALLOC( + entry_point, /* REQUIRED */ + this_spb /* REQUIRED */ + ) + + u32 strm_buf_ptr; /* REQUIRED */ + u32 strm_rs_config; /* REQUIRED */ + + /* Init. Ptr to CODEC input SCB + hi: Pointer to the SCB containing the + input buffer to which CODEC input + samples are written + lo: Flag indicating the link to the CODEC + input task is to be initialized + */ + ___DSP_DUAL_16BIT_ALLOC( + init_codec_input_link, + codec_input_buf_scb + ) + + /* Initialized by the host (host updates target volumes) */ + struct dsp_volume_control psi_vol_ctrl; + +}; + +struct dsp_src_task_scb { + ___DSP_DUAL_16BIT_ALLOC( + frames_left_in_gof, + gofs_left_in_sec + ) + + ___DSP_DUAL_16BIT_ALLOC( + const2_thirds, + num_extra_tnput_samples + ) + + ___DSP_DUAL_16BIT_ALLOC( + cor_per_gof, + correction_per_sec + ) + + ___DSP_DUAL_16BIT_ALLOC( + output_buf_producer_ptr, + junk_DMA_MID + ) + + ___DSP_DUAL_16BIT_ALLOC( + gof_length, + gofs_per_sec + ) + + u32 input_buf_strm_config; + + ___DSP_DUAL_16BIT_ALLOC( + reserved_for_SRC_use, + input_buf_consumer_ptr + ) + + u32 accum_phi; + + ___DSP_DUAL_16BIT_ALLOC( + exp_src_vol_change_rate, + input_buf_producer_ptr + ) + + ___DSP_DUAL_16BIT_ALLOC( + src_next_scb, + src_sub_list_ptr + ) + + ___DSP_DUAL_16BIT_ALLOC( + src_entry_point, + src_this_sbp + ) + + u32 src_strm_rs_config; + u32 src_strm_buf_ptr; + + u32 phiIncr6int_26frac; + + struct dsp_volume_control src_vol_ctrl; +}; + +struct dsp_decimate_by_pow2_scb { + /* decimationFactor = 2, 4, or 8 (larger factors waste too much memory + when compared to cascading decimators) + */ + ___DSP_DUAL_16BIT_ALLOC( + dec2_coef_base_ptr, + dec2_coef_increment + ) + + /* coefIncrement = 128 / decimationFactor (for our ROM filter) + coefBasePtr = 0x8000 (for our ROM filter) + */ + ___DSP_DUAL_16BIT_ALLOC( + dec2_in_samples_per_out_triplet, + dec2_extra_in_samples + ) + /* extraInSamples: # of accumulated, unused input samples (init. to 0) + inSamplesPerOutTriplet = 3 * decimationFactor + */ + + ___DSP_DUAL_16BIT_ALLOC( + dec2_const2_thirds, + dec2_half_num_taps_mp5 + ) + /* halfNumTapsM5: (1/2 number of taps in decimation filter) minus 5 + const2thirds: constant 2/3 in 16Q0 format (sign.15) + */ + + ___DSP_DUAL_16BIT_ALLOC( + dec2_output_buf_producer_ptr, + dec2_junkdma_mid + ) + + u32 dec2_reserved2; + + u32 dec2_input_nuf_strm_config; + /* inputBufStrmConfig: rsConfig for the input buffer to the decimator + (buffer size = decimationFactor * 32 dwords) + */ + + ___DSP_DUAL_16BIT_ALLOC( + dec2_phi_incr, + dec2_input_buf_consumer_ptr + ) + /* inputBufConsumerPtr: Input buffer read pointer (into SRC filter) + phiIncr = decimationFactor * 4 + */ + + u32 dec2_reserved3; + + ___DSP_DUAL_16BIT_ALLOC( + dec2_exp_vol_change_rate, + dec2_input_buf_producer_ptr + ) + /* inputBufProducerPtr: Input buffer write pointer + expVolChangeRate: Exponential volume change rate for possible + future mixer on input streams + */ + + ___DSP_DUAL_16BIT_ALLOC( + dec2_next_scb, + dec2_sub_list_ptr + ) + + ___DSP_DUAL_16BIT_ALLOC( + dec2_entry_point, + dec2_this_spb + ) + + u32 dec2_strm_rs_config; + u32 dec2_strm_buf_ptr; + + u32 dec2_reserved4; + + struct dsp_volume_control dec2_vol_ctrl; /* Not used! */ +}; + +struct dsp_vari_decimate_scb { + ___DSP_DUAL_16BIT_ALLOC( + vdec_frames_left_in_gof, + vdec_gofs_left_in_sec + ) + + ___DSP_DUAL_16BIT_ALLOC( + vdec_const2_thirds, + vdec_extra_in_samples + ) + /* extraInSamples: # of accumulated, unused input samples (init. to 0) + const2thirds: constant 2/3 in 16Q0 format (sign.15) */ + + ___DSP_DUAL_16BIT_ALLOC( + vdec_cor_per_gof, + vdec_correction_per_sec + ) + + ___DSP_DUAL_16BIT_ALLOC( + vdec_output_buf_producer_ptr, + vdec_input_buf_consumer_ptr + ) + /* inputBufConsumerPtr: Input buffer read pointer (into SRC filter) */ + ___DSP_DUAL_16BIT_ALLOC( + vdec_gof_length, + vdec_gofs_per_sec + ) + + u32 vdec_input_buf_strm_config; + /* inputBufStrmConfig: rsConfig for the input buffer to the decimator + (buffer size = 64 dwords) */ + u32 vdec_coef_increment; + /* coefIncrement = - 128.0 / decimationFactor (as a 32Q15 number) */ + + u32 vdec_accumphi; + /* accumPhi: accumulated fractional phase increment (6.26) */ + + ___DSP_DUAL_16BIT_ALLOC( + vdec_exp_vol_change_rate, + vdec_input_buf_producer_ptr + ) + /* inputBufProducerPtr: Input buffer write pointer + expVolChangeRate: Exponential volume change rate for possible + future mixer on input streams */ + + ___DSP_DUAL_16BIT_ALLOC( + vdec_next_scb, + vdec_sub_list_ptr + ) + + ___DSP_DUAL_16BIT_ALLOC( + vdec_entry_point, + vdec_this_spb + ) + + u32 vdec_strm_rs_config; + u32 vdec_strm_buf_ptr; + + u32 vdec_phi_incr_6int_26frac; + + struct dsp_volume_control vdec_vol_ctrl; +}; + + +/* SCB for MIX_TO_OSTREAM algorithm family */ +struct dsp_mix2_ostream_scb { + /* First 13 dwords from generic_scb_t */ + struct dsp_basic_dma_req basic_req; /* Optional */ + struct dsp_scatter_gather_ext sg_ext; /* Optional */ + ___DSP_DUAL_16BIT_ALLOC( + next_scb, /* REQUIRED */ + sub_list_ptr /* REQUIRED */ + ) + + ___DSP_DUAL_16BIT_ALLOC( + entry_point, /* REQUIRED */ + this_spb /* REQUIRED */ + ) + + u32 strm_rs_config; /* REQUIRED */ + u32 strm_buf_ptr; /* REQUIRED */ + + + /* hi: Number of mixed-down input triplets + computed since start of group + lo: Number of frames remaining to be + processed in the current group of + frames + */ + ___DSP_DUAL_16BIT_ALLOC( + frames_left_in_group, + accum_input_triplets + ) + + /* hi: Exponential volume change rate + for mixer on input streams + lo: Number of frames in the group + */ + ___DSP_DUAL_16BIT_ALLOC( + frame_group_length, + exp_vol_change_rate + ) + + ___DSP_DUAL_16BIT_ALLOC( + const_FFFF, + const_zero + ) +}; + + +/* SCB for S16_MIX algorithm */ +struct dsp_mix_only_scb { + /* First 13 dwords from generic_scb_t */ + struct dsp_basic_dma_req basic_req; /* Optional */ + struct dsp_scatter_gather_ext sg_ext; /* Optional */ + ___DSP_DUAL_16BIT_ALLOC( + next_scb, /* REQUIRED */ + sub_list_ptr /* REQUIRED */ + ) + + ___DSP_DUAL_16BIT_ALLOC( + entry_point, /* REQUIRED */ + this_spb /* REQUIRED */ + ) + + u32 strm_rs_config; /* REQUIRED */ + u32 strm_buf_ptr; /* REQUIRED */ + + u32 reserved; + struct dsp_volume_control vol_ctrl; +}; + +/* SCB for the async. CODEC input algorithm */ +struct dsp_async_codec_input_scb { + u32 io_free2; + + u32 io_current_total; + u32 io_previous_total; + + u16 io_count; + u16 io_count_limit; + + u16 o_fifo_base_addr; + u16 ost_mo_format; + /* 1 = stereo; 0 = mono + xxx for ASER 1 (not allowed); 118 for ASER2 */ + + u32 ostrm_rs_config; + u32 ostrm_buf_ptr; + + ___DSP_DUAL_16BIT_ALLOC( + io_sclks_per_lr_clk, + io_io_enable + ) + + u32 io_free4; + + ___DSP_DUAL_16BIT_ALLOC( + io_next_scb, + io_sub_list_ptr + ) + + ___DSP_DUAL_16BIT_ALLOC( + io_entry_point, + io_this_spb + ) + + u32 istrm_rs_config; + u32 istrm_buf_ptr; + + /* Init. 0000:8042: for ASER1 + 0000:8044: for ASER2 */ + ___DSP_DUAL_16BIT_ALLOC( + io_stat_reg_addr, + iofifo_pointer + ) + + /* Init 1 stero:100 ASER1 + Init 0 mono:110 ASER2 + */ + ___DSP_DUAL_16BIT_ALLOC( + ififo_base_addr, + ist_mo_format + ) + + u32 i_free; +}; + + +/* SCB for the SP/DIF CODEC input and output */ +struct dsp_spdifiscb { + ___DSP_DUAL_16BIT_ALLOC( + status_ptr, + status_start_ptr + ) + + u32 current_total; + u32 previous_total; + + ___DSP_DUAL_16BIT_ALLOC( + count, + count_limit + ) + + u32 status_data; + + ___DSP_DUAL_16BIT_ALLOC( + status, + free4 + ) + + u32 free3; + + ___DSP_DUAL_16BIT_ALLOC( + free2, + bit_count + ) + + u32 temp_status; + + ___DSP_DUAL_16BIT_ALLOC( + next_SCB, + sub_list_ptr + ) + + ___DSP_DUAL_16BIT_ALLOC( + entry_point, + this_spb + ) + + u32 strm_rs_config; + u32 strm_buf_ptr; + + ___DSP_DUAL_16BIT_ALLOC( + stat_reg_addr, + fifo_pointer + ) + + ___DSP_DUAL_16BIT_ALLOC( + fifo_base_addr, + st_mo_format + ) + + u32 free1; +}; + + +/* SCB for the SP/DIF CODEC input and output */ +struct dsp_spdifoscb { + + u32 free2; + + u32 free3[4]; + + /* Need to be here for compatibility with AsynchFGTxCode */ + u32 strm_rs_config; + + u32 strm_buf_ptr; + + ___DSP_DUAL_16BIT_ALLOC( + status, + free5 + ) + + u32 free4; + + ___DSP_DUAL_16BIT_ALLOC( + next_scb, + sub_list_ptr + ) + + ___DSP_DUAL_16BIT_ALLOC( + entry_point, + this_spb + ) + + u32 free6[2]; + + ___DSP_DUAL_16BIT_ALLOC( + stat_reg_addr, + fifo_pointer + ) + + ___DSP_DUAL_16BIT_ALLOC( + fifo_base_addr, + st_mo_format + ) + + u32 free1; +}; + + +struct dsp_asynch_fg_rx_scb { + ___DSP_DUAL_16BIT_ALLOC( + bot_buf_mask, + buf_Mask + ) + + ___DSP_DUAL_16BIT_ALLOC( + max, + min + ) + + ___DSP_DUAL_16BIT_ALLOC( + old_producer_pointer, + hfg_scb_ptr + ) + + ___DSP_DUAL_16BIT_ALLOC( + delta, + adjust_count + ) + + u32 unused2[5]; + + ___DSP_DUAL_16BIT_ALLOC( + sibling_ptr, + child_ptr + ) + + ___DSP_DUAL_16BIT_ALLOC( + code_ptr, + this_ptr + ) + + u32 strm_rs_config; + + u32 strm_buf_ptr; + + u32 unused_phi_incr; + + ___DSP_DUAL_16BIT_ALLOC( + right_targ, + left_targ + ) + + ___DSP_DUAL_16BIT_ALLOC( + right_vol, + left_vol + ) +}; + + +struct dsp_asynch_fg_tx_scb { + ___DSP_DUAL_16BIT_ALLOC( + not_buf_mask, + buf_mask + ) + + ___DSP_DUAL_16BIT_ALLOC( + max, + min + ) + + ___DSP_DUAL_16BIT_ALLOC( + unused1, + hfg_scb_ptr + ) + + ___DSP_DUAL_16BIT_ALLOC( + delta, + adjust_count + ) + + u32 accum_phi; + + ___DSP_DUAL_16BIT_ALLOC( + unused2, + const_one_third + ) + + u32 unused3[3]; + + ___DSP_DUAL_16BIT_ALLOC( + sibling_ptr, + child_ptr + ) + + ___DSP_DUAL_16BIT_ALLOC( + codePtr, + this_ptr + ) + + u32 strm_rs_config; + + u32 strm_buf_ptr; + + u32 phi_incr; + + ___DSP_DUAL_16BIT_ALLOC( + unused_right_targ, + unused_left_targ + ) + + ___DSP_DUAL_16BIT_ALLOC( + unused_right_vol, + unused_left_vol + ) +}; + + +struct dsp_output_snoop_scb { + /* First 13 dwords from generic_scb_t */ + struct dsp_basic_dma_req basic_req; /* Optional */ + struct dsp_scatter_gather_ext sg_ext; /* Optional */ + ___DSP_DUAL_16BIT_ALLOC( + next_scb, /* REQUIRED */ + sub_list_ptr /* REQUIRED */ + ) + + ___DSP_DUAL_16BIT_ALLOC( + entry_point, /* REQUIRED */ + this_spb /* REQUIRED */ + ) + + u32 strm_rs_config; /* REQUIRED */ + u32 strm_buf_ptr; /* REQUIRED */ + + ___DSP_DUAL_16BIT_ALLOC( + init_snoop_input_link, + snoop_child_input_scb + ) + + u32 snoop_input_buf_ptr; + + ___DSP_DUAL_16BIT_ALLOC( + reserved, + input_scb + ) +}; + +struct dsp_spio_write_scb { + ___DSP_DUAL_16BIT_ALLOC( + address1, + address2 + ) + + u32 data1; + + u32 data2; + + ___DSP_DUAL_16BIT_ALLOC( + address3, + address4 + ) + + u32 data3; + + u32 data4; + + ___DSP_DUAL_16BIT_ALLOC( + unused1, + data_ptr + ) + + u32 unused2[2]; + + ___DSP_DUAL_16BIT_ALLOC( + sibling_ptr, + child_ptr + ) + + ___DSP_DUAL_16BIT_ALLOC( + entry_point, + this_ptr + ) + + u32 unused3[5]; +}; + +struct dsp_magic_snoop_task { + u32 i0; + u32 i1; + + u32 strm_buf_ptr1; + + u16 i2; + u16 snoop_scb; + + u32 i3; + u32 i4; + u32 i5; + u32 i6; + + u32 i7; + + ___DSP_DUAL_16BIT_ALLOC( + next_scb, + sub_list_ptr + ) + + ___DSP_DUAL_16BIT_ALLOC( + entry_point, + this_ptr + ) + + u32 strm_buf_config; + u32 strm_buf_ptr2; + + u32 i8; + + struct dsp_volume_control vdec_vol_ctrl; +}; + + +struct dsp_filter_scb { + ___DSP_DUAL_16BIT_ALLOC( + a0_right, /* 0x00 */ + a0_left + ) + ___DSP_DUAL_16BIT_ALLOC( + a1_right, /* 0x01 */ + a1_left + ) + ___DSP_DUAL_16BIT_ALLOC( + a2_right, /* 0x02 */ + a2_left + ) + ___DSP_DUAL_16BIT_ALLOC( + output_buf_ptr, /* 0x03 */ + init + ) + + ___DSP_DUAL_16BIT_ALLOC( + filter_unused3, /* 0x04 */ + filter_unused2 + ) + + u32 prev_sample_output1; /* 0x05 */ + u32 prev_sample_output2; /* 0x06 */ + u32 prev_sample_input1; /* 0x07 */ + u32 prev_sample_input2; /* 0x08 */ + + ___DSP_DUAL_16BIT_ALLOC( + next_scb_ptr, /* 0x09 */ + sub_list_ptr + ) + + ___DSP_DUAL_16BIT_ALLOC( + entry_point, /* 0x0A */ + spb_ptr + ) + + u32 strm_rs_config; /* 0x0B */ + u32 strm_buf_ptr; /* 0x0C */ + + ___DSP_DUAL_16BIT_ALLOC( + b0_right, /* 0x0D */ + b0_left + ) + ___DSP_DUAL_16BIT_ALLOC( + b1_right, /* 0x0E */ + b1_left + ) + ___DSP_DUAL_16BIT_ALLOC( + b2_right, /* 0x0F */ + b2_left + ) +}; +#endif /* __DSP_SCB_TYPES_H__ */ diff --git a/sound/pci/cs46xx/cs46xx_dsp_spos.h b/sound/pci/cs46xx/cs46xx_dsp_spos.h new file mode 100644 index 000000000000..8008c59288a6 --- /dev/null +++ b/sound/pci/cs46xx/cs46xx_dsp_spos.h @@ -0,0 +1,234 @@ +/* + * The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#ifndef __CS46XX_DSP_SPOS_H__ +#define __CS46XX_DSP_SPOS_H__ + +#include "cs46xx_dsp_scb_types.h" +#include "cs46xx_dsp_task_types.h" + +#define SYMBOL_CONSTANT 0x0 +#define SYMBOL_SAMPLE 0x1 +#define SYMBOL_PARAMETER 0x2 +#define SYMBOL_CODE 0x3 + +#define SEGTYPE_SP_PROGRAM 0x00000001 +#define SEGTYPE_SP_PARAMETER 0x00000002 +#define SEGTYPE_SP_SAMPLE 0x00000003 +#define SEGTYPE_SP_COEFFICIENT 0x00000004 + +#define DSP_SPOS_UU 0x0deadul /* unused */ +#define DSP_SPOS_DC 0x0badul /* don't care */ +#define DSP_SPOS_DC_DC 0x0bad0badul /* don't care */ +#define DSP_SPOS_UUUU 0xdeadc0edul /* unused */ +#define DSP_SPOS_UUHI 0xdeadul +#define DSP_SPOS_UULO 0xc0edul +#define DSP_SPOS_DCDC 0x0badf1d0ul /* don't care */ +#define DSP_SPOS_DCDCHI 0x0badul +#define DSP_SPOS_DCDCLO 0xf1d0ul + +#define DSP_MAX_TASK_NAME 60 +#define DSP_MAX_SYMBOL_NAME 100 +#define DSP_MAX_SCB_NAME 60 +#define DSP_MAX_SCB_DESC 200 +#define DSP_MAX_TASK_DESC 50 + +#define DSP_MAX_PCM_CHANNELS 32 +#define DSP_MAX_SRC_NR 14 + +#define DSP_PCM_MAIN_CHANNEL 1 +#define DSP_PCM_REAR_CHANNEL 2 +#define DSP_PCM_CENTER_LFE_CHANNEL 3 +#define DSP_PCM_S71_CHANNEL 4 /* surround 7.1 */ +#define DSP_IEC958_CHANNEL 5 + +#define DSP_SPDIF_STATUS_OUTPUT_ENABLED 1 +#define DSP_SPDIF_STATUS_PLAYBACK_OPEN 2 +#define DSP_SPDIF_STATUS_HW_ENABLED 4 +#define DSP_SPDIF_STATUS_INPUT_CTRL_ENABLED 8 + +struct dsp_symbol_entry { + u32 address; + char symbol_name[DSP_MAX_SYMBOL_NAME]; + int symbol_type; + + /* initialized by driver */ + struct dsp_module_desc * module; + int deleted; +}; + +struct dsp_symbol_desc { + int nsymbols; + + struct dsp_symbol_entry *symbols; + + /* initialized by driver */ + int highest_frag_index; +}; + +struct dsp_segment_desc { + int segment_type; + u32 offset; + u32 size; + u32 * data; +}; + +struct dsp_module_desc { + char * module_name; + struct dsp_symbol_desc symbol_table; + int nsegments; + struct dsp_segment_desc * segments; + + /* initialized by driver */ + u32 overlay_begin_address; + u32 load_address; + int nfixups; +}; + +struct dsp_scb_descriptor { + char scb_name[DSP_MAX_SCB_NAME]; + u32 address; + int index; + u32 *data; + + struct dsp_scb_descriptor * sub_list_ptr; + struct dsp_scb_descriptor * next_scb_ptr; + struct dsp_scb_descriptor * parent_scb_ptr; + + struct dsp_symbol_entry * task_entry; + struct dsp_symbol_entry * scb_symbol; + + struct snd_info_entry *proc_info; + int ref_count; + + u16 volume[2]; + unsigned int deleted :1; + unsigned int updated :1; + unsigned int volume_set :1; +}; + +struct dsp_task_descriptor { + char task_name[DSP_MAX_TASK_NAME]; + int size; + u32 address; + int index; + u32 *data; +}; + +struct dsp_pcm_channel_descriptor { + int active; + int src_slot; + int pcm_slot; + u32 sample_rate; + u32 unlinked; + struct dsp_scb_descriptor * pcm_reader_scb; + struct dsp_scb_descriptor * src_scb; + struct dsp_scb_descriptor * mixer_scb; + + void * private_data; +}; + +struct dsp_spos_instance { + struct dsp_symbol_desc symbol_table; /* currently available loaded symbols in SP */ + + int nmodules; + struct dsp_module_desc * modules; /* modules loaded into SP */ + + struct dsp_segment_desc code; + + /* Main PCM playback mixer */ + struct dsp_scb_descriptor * master_mix_scb; + u16 dac_volume_right; + u16 dac_volume_left; + + /* Rear/surround PCM playback mixer */ + struct dsp_scb_descriptor * rear_mix_scb; + + /* Center/LFE mixer */ + struct dsp_scb_descriptor * center_lfe_mix_scb; + + int npcm_channels; + int nsrc_scb; + struct dsp_pcm_channel_descriptor pcm_channels[DSP_MAX_PCM_CHANNELS]; + int src_scb_slots[DSP_MAX_SRC_NR]; + + /* cache this symbols */ + struct dsp_symbol_entry * null_algorithm; /* used by PCMreaderSCB's */ + struct dsp_symbol_entry * s16_up; /* used by SRCtaskSCB's */ + + /* proc fs */ + struct snd_card *snd_card; + struct snd_info_entry * proc_dsp_dir; + struct snd_info_entry * proc_sym_info_entry; + struct snd_info_entry * proc_modules_info_entry; + struct snd_info_entry * proc_parameter_dump_info_entry; + struct snd_info_entry * proc_sample_dump_info_entry; + + /* SCB's descriptors */ + int nscb; + int scb_highest_frag_index; + struct dsp_scb_descriptor scbs[DSP_MAX_SCB_DESC]; + struct snd_info_entry * proc_scb_info_entry; + struct dsp_scb_descriptor * the_null_scb; + + /* Task's descriptors */ + int ntask; + struct dsp_task_descriptor tasks[DSP_MAX_TASK_DESC]; + struct snd_info_entry * proc_task_info_entry; + + /* SPDIF status */ + int spdif_status_out; + int spdif_status_in; + u16 spdif_input_volume_right; + u16 spdif_input_volume_left; + /* spdif channel status, + left right and user validity bits */ + unsigned int spdif_csuv_default; + unsigned int spdif_csuv_stream; + + /* SPDIF input sample rate converter */ + struct dsp_scb_descriptor * spdif_in_src; + /* SPDIF input asynch. receiver */ + struct dsp_scb_descriptor * asynch_rx_scb; + + /* Capture record mixer SCB */ + struct dsp_scb_descriptor * record_mixer_scb; + + /* CODEC input SCB */ + struct dsp_scb_descriptor * codec_in_scb; + + /* reference snooper */ + struct dsp_scb_descriptor * ref_snoop_scb; + + /* SPDIF output PCM reference */ + struct dsp_scb_descriptor * spdif_pcm_input_scb; + + /* asynch TX task */ + struct dsp_scb_descriptor * asynch_tx_scb; + + /* record sources */ + struct dsp_scb_descriptor * pcm_input; + struct dsp_scb_descriptor * adc_input; + + int spdif_in_sample_rate; +}; + +#endif /* __DSP_SPOS_H__ */ diff --git a/sound/pci/cs46xx/cs46xx_dsp_task_types.h b/sound/pci/cs46xx/cs46xx_dsp_task_types.h new file mode 100644 index 000000000000..5cf920bfda27 --- /dev/null +++ b/sound/pci/cs46xx/cs46xx_dsp_task_types.h @@ -0,0 +1,252 @@ +/* + * The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + * + * NOTE: comments are copy/paste from cwcemb80.lst + * provided by Tom Woller at Cirrus (my only + * documentation about the SP OS running inside + * the DSP) + */ + +#ifndef __CS46XX_DSP_TASK_TYPES_H__ +#define __CS46XX_DSP_TASK_TYPES_H__ + +#include "cs46xx_dsp_scb_types.h" + +/********************************************************************************************* +Example hierarchy of stream control blocks in the SP + +hfgTree +Ptr____Call (c) + \ + -------+------ ------------- ------------- ------------- ----- +| SBlaster IF |______\| Foreground |___\| Middlegr'nd |___\| Background |___\| Nul | +| |Goto /| tree header |g /| tree header |g /| tree header |g /| SCB |r + -------------- (g) ------------- ------------- ------------- ----- + |c |c |c |c + | | | | + \/ ------------- ------------- ------------- + | Foreground |_\ | Middlegr'nd |_\ | Background |_\ + | tree |g/ | tree |g/ | tree |g/ + ------------- ------------- ------------- + |c |c |c + | | | + \/ \/ \/ + +*********************************************************************************************/ + +#define HFG_FIRST_EXECUTE_MODE 0x0001 +#define HFG_FIRST_EXECUTE_MODE_BIT 0 +#define HFG_CONTEXT_SWITCH_MODE 0x0002 +#define HFG_CONTEXT_SWITCH_MODE_BIT 1 + +#define MAX_FG_STACK_SIZE 32 /* THESE NEED TO BE COMPUTED PROPERLY */ +#define MAX_MG_STACK_SIZE 16 +#define MAX_BG_STACK_SIZE 9 +#define MAX_HFG_STACK_SIZE 4 + +#define SLEEP_ACTIVE_INCREMENT 0 /* Enable task tree thread to go to sleep + This should only ever be used on the Background thread */ +#define STANDARD_ACTIVE_INCREMENT 1 /* Task tree thread normal operation */ +#define SUSPEND_ACTIVE_INCREMENT 2 /* Cause execution to suspend in the task tree thread + This should only ever be used on the Background thread */ + +#define HOSTFLAGS_DISABLE_BG_SLEEP 0 /* Host-controlled flag that determines whether we go to sleep + at the end of BG */ + +/* Minimal context save area for Hyper Forground */ +struct dsp_hf_save_area { + u32 r10_save; + u32 r54_save; + u32 r98_save; + + ___DSP_DUAL_16BIT_ALLOC( + status_save, + ind_save + ) + + ___DSP_DUAL_16BIT_ALLOC( + rci1_save, + rci0_save + ) + + u32 r32_save; + u32 r76_save; + u32 rsd2_save; + + ___DSP_DUAL_16BIT_ALLOC( + rsi2_save, /* See TaskTreeParameterBlock for + remainder of registers */ + rsa2Save + ) + /* saved as part of HFG context */ +}; + + +/* Task link data structure */ +struct dsp_tree_link { + ___DSP_DUAL_16BIT_ALLOC( + /* Pointer to sibling task control block */ + next_scb, + /* Pointer to child task control block */ + sub_ptr + ) + + ___DSP_DUAL_16BIT_ALLOC( + /* Pointer to code entry point */ + entry_point, + /* Pointer to local data */ + this_spb + ) +}; + + +struct dsp_task_tree_data { + ___DSP_DUAL_16BIT_ALLOC( + /* Initial tock count; controls task tree execution rate */ + tock_count_limit, + /* Tock down counter */ + tock_count + ) + + /* Add to ActiveCount when TockCountLimit reached: + Subtract on task tree termination */ + ___DSP_DUAL_16BIT_ALLOC( + active_tncrement, + /* Number of pending activations for task tree */ + active_count + ) + + ___DSP_DUAL_16BIT_ALLOC( + /* BitNumber to enable modification of correct bit in ActiveTaskFlags */ + active_bit, + /* Pointer to OS location for indicating current activity on task level */ + active_task_flags_ptr + ) + + /* Data structure for controlling movement of memory blocks:- + currently unused */ + ___DSP_DUAL_16BIT_ALLOC( + mem_upd_ptr, + /* Data structure for controlling synchronous link update */ + link_upd_ptr + ) + + ___DSP_DUAL_16BIT_ALLOC( + /* Save area for remainder of full context. */ + save_area, + /* Address of start of local stack for data storage */ + data_stack_base_ptr + ) + +}; + + +struct dsp_interval_timer_data +{ + /* These data items have the same relative locations to those */ + ___DSP_DUAL_16BIT_ALLOC( + interval_timer_period, + itd_unused + ) + + /* used for this data in the SPOS control block for SPOS 1.0 */ + ___DSP_DUAL_16BIT_ALLOC( + num_FG_ticks_this_interval, + num_intervals + ) +}; + + +/* This structure contains extra storage for the task tree + Currently, this additional data is related only to a full context save */ +struct dsp_task_tree_context_block { + /* Up to 10 values are saved onto the stack. 8 for the task tree, 1 for + The access to the context switch (call or interrupt), and 1 spare that + users should never use. This last may be required by the system */ + ___DSP_DUAL_16BIT_ALLOC( + stack1, + stack0 + ) + ___DSP_DUAL_16BIT_ALLOC( + stack3, + stack2 + ) + ___DSP_DUAL_16BIT_ALLOC( + stack5, + stack4 + ) + ___DSP_DUAL_16BIT_ALLOC( + stack7, + stack6 + ) + ___DSP_DUAL_16BIT_ALLOC( + stack9, + stack8 + ) + + u32 saverfe; + + /* Value may be overwriten by stack save algorithm. + Retain the size of the stack data saved here if used */ + ___DSP_DUAL_16BIT_ALLOC( + reserved1, + stack_size + ) + u32 saverba; /* (HFG) */ + u32 saverdc; + u32 savers_config_23; /* (HFG) */ + u32 savers_DMA23; /* (HFG) */ + u32 saversa0; + u32 saversi0; + u32 saversa1; + u32 saversi1; + u32 saversa3; + u32 saversd0; + u32 saversd1; + u32 saversd3; + u32 savers_config01; + u32 savers_DMA01; + u32 saveacc0hl; + u32 saveacc1hl; + u32 saveacc0xacc1x; + u32 saveacc2hl; + u32 saveacc3hl; + u32 saveacc2xacc3x; + u32 saveaux0hl; + u32 saveaux1hl; + u32 saveaux0xaux1x; + u32 saveaux2hl; + u32 saveaux3hl; + u32 saveaux2xaux3x; + u32 savershouthl; + u32 savershoutxmacmode; +}; + + +struct dsp_task_tree_control_block { + struct dsp_hf_save_area context; + struct dsp_tree_link links; + struct dsp_task_tree_data data; + struct dsp_task_tree_context_block context_blk; + struct dsp_interval_timer_data int_timer; +}; + + +#endif /* __DSP_TASK_TYPES_H__ */ diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 4fa53161b094..f75f5ffdfdfb 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -61,7 +61,7 @@ #include <sound/info.h> #include <sound/pcm.h> #include <sound/pcm_params.h> -#include <sound/cs46xx.h> +#include "cs46xx.h" #include <asm/io.h> @@ -3599,9 +3599,10 @@ static unsigned int saved_regs[] = { BA1_CVOL, }; -int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_cs46xx_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_cs46xx *chip = card->private_data; int i, amp_saved; @@ -3628,13 +3629,14 @@ int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -int snd_cs46xx_resume(struct pci_dev *pci) +static int snd_cs46xx_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_cs46xx *chip = card->private_data; int amp_saved; #ifdef CONFIG_SND_CS46XX_NEW_DSP @@ -3707,6 +3709,8 @@ int snd_cs46xx_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +SIMPLE_DEV_PM_OPS(snd_cs46xx_pm, snd_cs46xx_suspend, snd_cs46xx_resume); #endif /* CONFIG_PM */ diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index e377287192aa..56fec0bc0efb 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -32,7 +32,7 @@ #include <sound/control.h> #include <sound/info.h> #include <sound/asoundef.h> -#include <sound/cs46xx.h> +#include "cs46xx.h" #include "cs46xx_lib.h" #include "dsp_spos.h" diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 00b148a10239..c2c695b07f8c 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -31,7 +31,7 @@ #include <sound/core.h> #include <sound/control.h> #include <sound/info.h> -#include <sound/cs46xx.h> +#include "cs46xx.h" #include "cs46xx_lib.h" #include "dsp_spos.h" diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index c47cabff2bfa..f1e4229993af 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -291,23 +291,11 @@ static int __devinit snd_cs5530_probe(struct pci_dev *pci, return 0; } -static struct pci_driver driver = { +static struct pci_driver cs5530_driver = { .name = KBUILD_MODNAME, .id_table = snd_cs5530_ids, .probe = snd_cs5530_probe, .remove = __devexit_p(snd_cs5530_remove), }; -static int __init alsa_card_cs5530_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_cs5530_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_cs5530_init) -module_exit(alsa_card_cs5530_exit) - +module_pci_driver(cs5530_driver); diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index a2fb2173e980..51f64ba5facf 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -394,29 +394,19 @@ static void __devexit snd_cs5535audio_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver cs5535audio_driver = { .name = KBUILD_MODNAME, .id_table = snd_cs5535audio_ids, .probe = snd_cs5535audio_probe, .remove = __devexit_p(snd_cs5535audio_remove), #ifdef CONFIG_PM - .suspend = snd_cs5535audio_suspend, - .resume = snd_cs5535audio_resume, + .driver = { + .pm = &snd_cs5535audio_pm, + }, #endif }; -static int __init alsa_card_cs5535audio_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_cs5535audio_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_cs5535audio_init) -module_exit(alsa_card_cs5535audio_exit) +module_pci_driver(cs5535audio_driver); MODULE_AUTHOR("Jaya Kumar"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h index 51966d782a3c..bb3cc641130c 100644 --- a/sound/pci/cs5535audio/cs5535audio.h +++ b/sound/pci/cs5535audio/cs5535audio.h @@ -94,10 +94,7 @@ struct cs5535audio { struct cs5535audio_dma dmas[NUM_CS5535AUDIO_DMAS]; }; -#ifdef CONFIG_PM -int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state); -int snd_cs5535audio_resume(struct pci_dev *pci); -#endif +extern const struct dev_pm_ops snd_cs5535audio_pm; #ifdef CONFIG_OLPC void __devinit olpc_prequirks(struct snd_card *card, diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c index 185b00088320..6c34def5986d 100644 --- a/sound/pci/cs5535audio/cs5535audio_pm.c +++ b/sound/pci/cs5535audio/cs5535audio_pm.c @@ -55,9 +55,10 @@ static void snd_cs5535audio_stop_hardware(struct cs5535audio *cs5535au) } -int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_cs5535audio_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct cs5535audio *cs5535au = card->private_data; int i; @@ -77,13 +78,14 @@ int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state) return -EIO; } pci_disable_device(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -int snd_cs5535audio_resume(struct pci_dev *pci) +static int snd_cs5535audio_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct cs5535audio *cs5535au = card->private_data; u32 tmp; int timeout; @@ -129,3 +131,4 @@ int snd_cs5535audio_resume(struct pci_dev *pci) return 0; } +SIMPLE_DEV_PM_OPS(snd_cs5535audio_pm, snd_cs5535audio_suspend, snd_cs5535audio_resume); diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index d8a4423539ce..8e40262d4117 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1537,7 +1537,7 @@ static void atc_connect_resources(struct ct_atc *atc) } #ifdef CONFIG_PM -static int atc_suspend(struct ct_atc *atc, pm_message_t state) +static int atc_suspend(struct ct_atc *atc) { int i; struct hw *hw = atc->hw; @@ -1553,7 +1553,7 @@ static int atc_suspend(struct ct_atc *atc, pm_message_t state) atc_release_resources(atc); - hw->suspend(hw, state); + hw->suspend(hw); return 0; } diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h index 3a0def656af0..653e813ad142 100644 --- a/sound/pci/ctxfi/ctatc.h +++ b/sound/pci/ctxfi/ctatc.h @@ -144,7 +144,7 @@ struct ct_atc { struct ct_timer *timer; #ifdef CONFIG_PM - int (*suspend)(struct ct_atc *atc, pm_message_t state); + int (*suspend)(struct ct_atc *atc); int (*resume)(struct ct_atc *atc); #define NUM_PCMS (NUM_CTALSADEVS - 1) struct snd_pcm *pcms[NUM_PCMS]; diff --git a/sound/pci/ctxfi/cthardware.h b/sound/pci/ctxfi/cthardware.h index 908315bec3b4..c56fe533b3f3 100644 --- a/sound/pci/ctxfi/cthardware.h +++ b/sound/pci/ctxfi/cthardware.h @@ -73,7 +73,7 @@ struct hw { int (*card_stop)(struct hw *hw); int (*pll_init)(struct hw *hw, unsigned int rsr); #ifdef CONFIG_PM - int (*suspend)(struct hw *hw, pm_message_t state); + int (*suspend)(struct hw *hw); int (*resume)(struct hw *hw, struct card_conf *info); #endif int (*is_adc_source_selected)(struct hw *hw, enum ADCSRC source); diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index a7df19791f5a..dc1969bc67d4 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -2086,7 +2086,7 @@ static int hw_card_init(struct hw *hw, struct card_conf *info) } #ifdef CONFIG_PM -static int hw_suspend(struct hw *hw, pm_message_t state) +static int hw_suspend(struct hw *hw) { struct pci_dev *pci = hw->pci; @@ -2099,7 +2099,7 @@ static int hw_suspend(struct hw *hw, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index d6c54b524bfa..9d1231dc4ae2 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -2202,7 +2202,7 @@ static int hw_card_init(struct hw *hw, struct card_conf *info) } #ifdef CONFIG_PM -static int hw_suspend(struct hw *hw, pm_message_t state) +static int hw_suspend(struct hw *hw) { struct pci_dev *pci = hw->pci; @@ -2210,7 +2210,7 @@ static int hw_suspend(struct hw *hw, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index 15d95d2bacee..e002183ef8b2 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -126,21 +126,26 @@ static void __devexit ct_card_remove(struct pci_dev *pci) } #ifdef CONFIG_PM -static int ct_card_suspend(struct pci_dev *pci, pm_message_t state) +static int ct_card_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct snd_card *card = dev_get_drvdata(dev); struct ct_atc *atc = card->private_data; - return atc->suspend(atc, state); + return atc->suspend(atc); } -static int ct_card_resume(struct pci_dev *pci) +static int ct_card_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct snd_card *card = dev_get_drvdata(dev); struct ct_atc *atc = card->private_data; return atc->resume(atc); } + +static SIMPLE_DEV_PM_OPS(ct_card_pm, ct_card_suspend, ct_card_resume); +#define CT_CARD_PM_OPS &ct_card_pm +#else +#define CT_CARD_PM_OPS NULL #endif static struct pci_driver ct_driver = { @@ -148,21 +153,9 @@ static struct pci_driver ct_driver = { .id_table = ct_pci_dev_ids, .probe = ct_card_probe, .remove = __devexit_p(ct_card_remove), -#ifdef CONFIG_PM - .suspend = ct_card_suspend, - .resume = ct_card_resume, -#endif + .driver = { + .pm = CT_CARD_PM_OPS, + }, }; -static int __init ct_card_init(void) -{ - return pci_register_driver(&ct_driver); -} - -static void __exit ct_card_exit(void) -{ - pci_unregister_driver(&ct_driver); -} - -module_init(ct_card_init) -module_exit(ct_card_exit) +module_pci_driver(ct_driver); diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 595c11f904bb..0ff754f180d0 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2205,9 +2205,10 @@ ctl_error: #if defined(CONFIG_PM) -static int snd_echo_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_echo_suspend(struct device *dev) { - struct echoaudio *chip = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct echoaudio *chip = dev_get_drvdata(dev); DE_INIT(("suspend start\n")); snd_pcm_suspend_all(chip->analog_pcm); @@ -2242,9 +2243,10 @@ static int snd_echo_suspend(struct pci_dev *pci, pm_message_t state) -static int snd_echo_resume(struct pci_dev *pci) +static int snd_echo_resume(struct device *dev) { - struct echoaudio *chip = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct echoaudio *chip = dev_get_drvdata(dev); struct comm_page *commpage, *commpage_bak; u32 pipe_alloc_mask; int err; @@ -2307,10 +2309,13 @@ static int snd_echo_resume(struct pci_dev *pci) return 0; } +static SIMPLE_DEV_PM_OPS(snd_echo_pm, snd_echo_suspend, snd_echo_resume); +#define SND_ECHO_PM_OPS &snd_echo_pm +#else +#define SND_ECHO_PM_OPS NULL #endif /* CONFIG_PM */ - static void __devexit snd_echo_remove(struct pci_dev *pci) { struct echoaudio *chip; @@ -2328,33 +2333,14 @@ static void __devexit snd_echo_remove(struct pci_dev *pci) ******************************************************************************/ /* pci_driver definition */ -static struct pci_driver driver = { +static struct pci_driver echo_driver = { .name = KBUILD_MODNAME, .id_table = snd_echo_ids, .probe = snd_echo_probe, .remove = __devexit_p(snd_echo_remove), -#ifdef CONFIG_PM - .suspend = snd_echo_suspend, - .resume = snd_echo_resume, -#endif /* CONFIG_PM */ + .driver = { + .pm = SND_ECHO_PM_OPS, + }, }; - - -/* initialization of the module */ -static int __init alsa_card_echo_init(void) -{ - return pci_register_driver(&driver); -} - - - -/* clean up the module */ -static void __exit alsa_card_echo_exit(void) -{ - pci_unregister_driver(&driver); -} - - -module_init(alsa_card_echo_init) -module_exit(alsa_card_echo_exit) +module_pci_driver(echo_driver); diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 790c65d980c8..ddac4e6d660d 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -207,9 +207,10 @@ static void __devexit snd_card_emu10k1_remove(struct pci_dev *pci) #ifdef CONFIG_PM -static int snd_emu10k1_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_emu10k1_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_emu10k1 *emu = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -231,13 +232,14 @@ static int snd_emu10k1_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_emu10k1_resume(struct pci_dev *pci) +static int snd_emu10k1_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_emu10k1 *emu = card->private_data; pci_set_power_state(pci, PCI_D0); @@ -261,28 +263,21 @@ static int snd_emu10k1_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -#endif -static struct pci_driver driver = { +static SIMPLE_DEV_PM_OPS(snd_emu10k1_pm, snd_emu10k1_suspend, snd_emu10k1_resume); +#define SND_EMU10K1_PM_OPS &snd_emu10k1_pm +#else +#define SND_EMU10K1_PM_OPS NULL +#endif /* CONFIG_PM */ + +static struct pci_driver emu10k1_driver = { .name = KBUILD_MODNAME, .id_table = snd_emu10k1_ids, .probe = snd_card_emu10k1_probe, .remove = __devexit_p(snd_card_emu10k1_remove), -#ifdef CONFIG_PM - .suspend = snd_emu10k1_suspend, - .resume = snd_emu10k1_resume, -#endif + .driver = { + .pm = SND_EMU10K1_PM_OPS, + }, }; -static int __init alsa_card_emu10k1_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_emu10k1_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_emu10k1_init) -module_exit(alsa_card_emu10k1_exit) +module_pci_driver(emu10k1_driver); diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 47a651cb6e84..5c8978b2c4d9 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1612,24 +1612,11 @@ static DEFINE_PCI_DEVICE_TABLE(snd_emu10k1x_ids) = { MODULE_DEVICE_TABLE(pci, snd_emu10k1x_ids); // pci_driver definition -static struct pci_driver driver = { +static struct pci_driver emu10k1x_driver = { .name = KBUILD_MODNAME, .id_table = snd_emu10k1x_ids, .probe = snd_emu10k1x_probe, .remove = __devexit_p(snd_emu10k1x_remove), }; -// initialization of the module -static int __init alsa_card_emu10k1x_init(void) -{ - return pci_register_driver(&driver); -} - -// clean up the module -static void __exit alsa_card_emu10k1x_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_emu10k1x_init) -module_exit(alsa_card_emu10k1x_exit) +module_pci_driver(emu10k1x_driver); diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 47a245e84190..f7e6f73186e1 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -2033,9 +2033,10 @@ static void snd_ensoniq_chip_init(struct ensoniq *ensoniq) } #ifdef CONFIG_PM -static int snd_ensoniq_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_ensoniq_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct ensoniq *ensoniq = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -2058,13 +2059,14 @@ static int snd_ensoniq_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_ensoniq_resume(struct pci_dev *pci) +static int snd_ensoniq_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct ensoniq *ensoniq = card->private_data; pci_set_power_state(pci, PCI_D0); @@ -2087,8 +2089,12 @@ static int snd_ensoniq_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -#endif /* CONFIG_PM */ +static SIMPLE_DEV_PM_OPS(snd_ensoniq_pm, snd_ensoniq_suspend, snd_ensoniq_resume); +#define SND_ENSONIQ_PM_OPS &snd_ensoniq_pm +#else +#define SND_ENSONIQ_PM_OPS NULL +#endif /* CONFIG_PM */ static int __devinit snd_ensoniq_create(struct snd_card *card, struct pci_dev *pci, @@ -2488,26 +2494,14 @@ static void __devexit snd_audiopci_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver ens137x_driver = { .name = KBUILD_MODNAME, .id_table = snd_audiopci_ids, .probe = snd_audiopci_probe, .remove = __devexit_p(snd_audiopci_remove), -#ifdef CONFIG_PM - .suspend = snd_ensoniq_suspend, - .resume = snd_ensoniq_resume, -#endif + .driver = { + .pm = SND_ENSONIQ_PM_OPS, + }, }; -static int __init alsa_card_ens137x_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_ens137x_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_ens137x_init) -module_exit(alsa_card_ens137x_exit) +module_pci_driver(ens137x_driver); diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 53eb76b41108..dbb81807bc1a 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1321,35 +1321,30 @@ static int snd_es1938_put_double(struct snd_kcontrol *kcontrol, return change; } -static unsigned int db_scale_master[] = { - TLV_DB_RANGE_HEAD(2), +static const DECLARE_TLV_DB_RANGE(db_scale_master, 0, 54, TLV_DB_SCALE_ITEM(-3600, 50, 1), 54, 63, TLV_DB_SCALE_ITEM(-900, 100, 0), -}; +); -static unsigned int db_scale_audio1[] = { - TLV_DB_RANGE_HEAD(2), +static const DECLARE_TLV_DB_RANGE(db_scale_audio1, 0, 8, TLV_DB_SCALE_ITEM(-3300, 300, 1), 8, 15, TLV_DB_SCALE_ITEM(-900, 150, 0), -}; +); -static unsigned int db_scale_audio2[] = { - TLV_DB_RANGE_HEAD(2), +static const DECLARE_TLV_DB_RANGE(db_scale_audio2, 0, 8, TLV_DB_SCALE_ITEM(-3450, 300, 1), 8, 15, TLV_DB_SCALE_ITEM(-1050, 150, 0), -}; +); -static unsigned int db_scale_mic[] = { - TLV_DB_RANGE_HEAD(2), +static const DECLARE_TLV_DB_RANGE(db_scale_mic, 0, 8, TLV_DB_SCALE_ITEM(-2400, 300, 1), 8, 15, TLV_DB_SCALE_ITEM(0, 150, 0), -}; +); -static unsigned int db_scale_line[] = { - TLV_DB_RANGE_HEAD(2), +static const DECLARE_TLV_DB_RANGE(db_scale_line, 0, 8, TLV_DB_SCALE_ITEM(-3150, 300, 1), 8, 15, TLV_DB_SCALE_ITEM(-750, 150, 0), -}; +); static const DECLARE_TLV_DB_SCALE(db_scale_capture, 0, 150, 0); @@ -1474,9 +1469,10 @@ static unsigned char saved_regs[SAVED_REG_SIZE+1] = { }; -static int es1938_suspend(struct pci_dev *pci, pm_message_t state) +static int es1938_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct es1938 *chip = card->private_data; unsigned char *s, *d; @@ -1494,13 +1490,14 @@ static int es1938_suspend(struct pci_dev *pci, pm_message_t state) } pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int es1938_resume(struct pci_dev *pci) +static int es1938_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct es1938 *chip = card->private_data; unsigned char *s, *d; @@ -1534,6 +1531,11 @@ static int es1938_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(es1938_pm, es1938_suspend, es1938_resume); +#define ES1938_PM_OPS &es1938_pm +#else +#define ES1938_PM_OPS NULL #endif /* CONFIG_PM */ #ifdef SUPPORT_JOYSTICK @@ -1882,26 +1884,14 @@ static void __devexit snd_es1938_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver es1938_driver = { .name = KBUILD_MODNAME, .id_table = snd_es1938_ids, .probe = snd_es1938_probe, .remove = __devexit_p(snd_es1938_remove), -#ifdef CONFIG_PM - .suspend = es1938_suspend, - .resume = es1938_resume, -#endif + .driver = { + .pm = ES1938_PM_OPS, + }, }; -static int __init alsa_card_es1938_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_es1938_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_es1938_init) -module_exit(alsa_card_es1938_exit) +module_pci_driver(es1938_driver); diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 0f2811eeeebd..fb4c90b99c00 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2381,9 +2381,10 @@ static void snd_es1968_start_irq(struct es1968 *chip) /* * PM support */ -static int es1968_suspend(struct pci_dev *pci, pm_message_t state) +static int es1968_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct es1968 *chip = card->private_data; if (! chip->do_pm) @@ -2398,13 +2399,14 @@ static int es1968_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int es1968_resume(struct pci_dev *pci) +static int es1968_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct es1968 *chip = card->private_data; struct esschan *es; @@ -2454,6 +2456,11 @@ static int es1968_resume(struct pci_dev *pci) chip->in_suspend = 0; return 0; } + +static SIMPLE_DEV_PM_OPS(es1968_pm, es1968_suspend, es1968_resume); +#define ES1968_PM_OPS &es1968_pm +#else +#define ES1968_PM_OPS NULL #endif /* CONFIG_PM */ #ifdef SUPPORT_JOYSTICK @@ -2898,26 +2905,14 @@ static void __devexit snd_es1968_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver es1968_driver = { .name = KBUILD_MODNAME, .id_table = snd_es1968_ids, .probe = snd_es1968_probe, .remove = __devexit_p(snd_es1968_remove), -#ifdef CONFIG_PM - .suspend = es1968_suspend, - .resume = es1968_resume, -#endif + .driver = { + .pm = ES1968_PM_OPS, + }, }; -static int __init alsa_card_es1968_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_es1968_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_es1968_init) -module_exit(alsa_card_es1968_exit) +module_pci_driver(es1968_driver); diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 5265c576a26a..522c8706f244 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1369,9 +1369,10 @@ static unsigned char saved_regs[] = { FM801_CODEC_CTRL, FM801_I2S_MODE, FM801_VOLUME, FM801_GEN_CTRL, }; -static int snd_fm801_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_fm801_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct fm801 *chip = card->private_data; int i; @@ -1385,13 +1386,14 @@ static int snd_fm801_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_fm801_resume(struct pci_dev *pci) +static int snd_fm801_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct fm801 *chip = card->private_data; int i; @@ -1414,28 +1416,21 @@ static int snd_fm801_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -#endif -static struct pci_driver driver = { +static SIMPLE_DEV_PM_OPS(snd_fm801_pm, snd_fm801_suspend, snd_fm801_resume); +#define SND_FM801_PM_OPS &snd_fm801_pm +#else +#define SND_FM801_PM_OPS NULL +#endif /* CONFIG_PM */ + +static struct pci_driver fm801_driver = { .name = KBUILD_MODNAME, .id_table = snd_fm801_ids, .probe = snd_card_fm801_probe, .remove = __devexit_p(snd_card_fm801_remove), -#ifdef CONFIG_PM - .suspend = snd_fm801_suspend, - .resume = snd_fm801_resume, -#endif + .driver = { + .pm = SND_FM801_PM_OPS, + }, }; -static int __init alsa_card_fm801_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_fm801_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_fm801_init) -module_exit(alsa_card_fm801_exit) +module_pci_driver(fm801_driver); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 163b6b5de3eb..194d625c1f83 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -53,15 +53,14 @@ config SND_HDA_INPUT_BEEP driver. This interface is used to generate digital beeps. config SND_HDA_INPUT_BEEP_MODE - int "Digital beep registration mode (0=off, 1=on, 2=mute sw on/off)" + int "Digital beep registration mode (0=off, 1=on)" depends on SND_HDA_INPUT_BEEP=y default "1" - range 0 2 + range 0 1 help Set 0 to disable the digital beep interface for HD-audio by default. Set 1 to always enable the digital beep interface for HD-audio by - default. Set 2 to control the beep device registration to input - layer using a "Beep Switch" in mixer applications. + default. config SND_HDA_INPUT_JACK bool "Support jack plugging notification via input layer" @@ -97,19 +96,6 @@ config SND_HDA_CODEC_REALTEK snd-hda-codec-realtek. This module is automatically loaded at probing. -config SND_HDA_ENABLE_REALTEK_QUIRKS - bool "Build static quirks for Realtek codecs" - depends on SND_HDA_CODEC_REALTEK - default y - help - Say Y here to build the static quirks codes for Realtek codecs. - If you need the "model" preset that the default BIOS auto-parser - can't handle, turn this option on. - - If your device works with model=auto option, basically you don't - need the quirk code. By turning this off, you can reduce the - module size quite a lot. - config SND_HDA_CODEC_ANALOG bool "Build Analog Device HD-audio codec support" default y diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index ace157cc3d15..bd4149f1aaf4 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,6 +1,6 @@ snd-hda-intel-objs := hda_intel.o -snd-hda-codec-y := hda_codec.o hda_jack.o +snd-hda-codec-y := hda_codec.o hda_jack.o hda_auto_parser.o snd-hda-codec-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o snd-hda-codec-$(CONFIG_PROC_FS) += hda_proc.o snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c new file mode 100644 index 000000000000..647218d69f68 --- /dev/null +++ b/sound/pci/hda/hda_auto_parser.c @@ -0,0 +1,759 @@ +/* + * BIOS auto-parser helper functions for HD-audio + * + * Copyright (c) 2012 Takashi Iwai <tiwai@suse.de> + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#include <linux/slab.h> +#include <linux/export.h> +#include <sound/core.h> +#include "hda_codec.h" +#include "hda_local.h" +#include "hda_auto_parser.h" + +#define SFX "hda_codec: " + +/* + * Helper for automatic pin configuration + */ + +static int is_in_nid_list(hda_nid_t nid, const hda_nid_t *list) +{ + for (; *list; list++) + if (*list == nid) + return 1; + return 0; +} + + +/* + * Sort an associated group of pins according to their sequence numbers. + */ +static void sort_pins_by_sequence(hda_nid_t *pins, short *sequences, + int num_pins) +{ + int i, j; + short seq; + hda_nid_t nid; + + for (i = 0; i < num_pins; i++) { + for (j = i + 1; j < num_pins; j++) { + if (sequences[i] > sequences[j]) { + seq = sequences[i]; + sequences[i] = sequences[j]; + sequences[j] = seq; + nid = pins[i]; + pins[i] = pins[j]; + pins[j] = nid; + } + } + } +} + + +/* add the found input-pin to the cfg->inputs[] table */ +static void add_auto_cfg_input_pin(struct auto_pin_cfg *cfg, hda_nid_t nid, + int type) +{ + if (cfg->num_inputs < AUTO_CFG_MAX_INS) { + cfg->inputs[cfg->num_inputs].pin = nid; + cfg->inputs[cfg->num_inputs].type = type; + cfg->num_inputs++; + } +} + +/* sort inputs in the order of AUTO_PIN_* type */ +static void sort_autocfg_input_pins(struct auto_pin_cfg *cfg) +{ + int i, j; + + for (i = 0; i < cfg->num_inputs; i++) { + for (j = i + 1; j < cfg->num_inputs; j++) { + if (cfg->inputs[i].type > cfg->inputs[j].type) { + struct auto_pin_cfg_item tmp; + tmp = cfg->inputs[i]; + cfg->inputs[i] = cfg->inputs[j]; + cfg->inputs[j] = tmp; + } + } + } +} + +/* Reorder the surround channels + * ALSA sequence is front/surr/clfe/side + * HDA sequence is: + * 4-ch: front/surr => OK as it is + * 6-ch: front/clfe/surr + * 8-ch: front/clfe/rear/side|fc + */ +static void reorder_outputs(unsigned int nums, hda_nid_t *pins) +{ + hda_nid_t nid; + + switch (nums) { + case 3: + case 4: + nid = pins[1]; + pins[1] = pins[2]; + pins[2] = nid; + break; + } +} + +/* + * Parse all pin widgets and store the useful pin nids to cfg + * + * The number of line-outs or any primary output is stored in line_outs, + * and the corresponding output pins are assigned to line_out_pins[], + * in the order of front, rear, CLFE, side, ... + * + * If more extra outputs (speaker and headphone) are found, the pins are + * assisnged to hp_pins[] and speaker_pins[], respectively. If no line-out jack + * is detected, one of speaker of HP pins is assigned as the primary + * output, i.e. to line_out_pins[0]. So, line_outs is always positive + * if any analog output exists. + * + * The analog input pins are assigned to inputs array. + * The digital input/output pins are assigned to dig_in_pin and dig_out_pin, + * respectively. + */ +int snd_hda_parse_pin_defcfg(struct hda_codec *codec, + struct auto_pin_cfg *cfg, + const hda_nid_t *ignore_nids, + unsigned int cond_flags) +{ + hda_nid_t nid, end_nid; + short seq, assoc_line_out; + short sequences_line_out[ARRAY_SIZE(cfg->line_out_pins)]; + short sequences_speaker[ARRAY_SIZE(cfg->speaker_pins)]; + short sequences_hp[ARRAY_SIZE(cfg->hp_pins)]; + int i; + + memset(cfg, 0, sizeof(*cfg)); + + memset(sequences_line_out, 0, sizeof(sequences_line_out)); + memset(sequences_speaker, 0, sizeof(sequences_speaker)); + memset(sequences_hp, 0, sizeof(sequences_hp)); + assoc_line_out = 0; + + codec->ignore_misc_bit = true; + end_nid = codec->start_nid + codec->num_nodes; + for (nid = codec->start_nid; nid < end_nid; nid++) { + unsigned int wid_caps = get_wcaps(codec, nid); + unsigned int wid_type = get_wcaps_type(wid_caps); + unsigned int def_conf; + short assoc, loc, conn, dev; + + /* read all default configuration for pin complex */ + if (wid_type != AC_WID_PIN) + continue; + /* ignore the given nids (e.g. pc-beep returns error) */ + if (ignore_nids && is_in_nid_list(nid, ignore_nids)) + continue; + + def_conf = snd_hda_codec_get_pincfg(codec, nid); + if (!(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & + AC_DEFCFG_MISC_NO_PRESENCE)) + codec->ignore_misc_bit = false; + conn = get_defcfg_connect(def_conf); + if (conn == AC_JACK_PORT_NONE) + continue; + loc = get_defcfg_location(def_conf); + dev = get_defcfg_device(def_conf); + + /* workaround for buggy BIOS setups */ + if (dev == AC_JACK_LINE_OUT) { + if (conn == AC_JACK_PORT_FIXED) + dev = AC_JACK_SPEAKER; + } + + switch (dev) { + case AC_JACK_LINE_OUT: + seq = get_defcfg_sequence(def_conf); + assoc = get_defcfg_association(def_conf); + + if (!(wid_caps & AC_WCAP_STEREO)) + if (!cfg->mono_out_pin) + cfg->mono_out_pin = nid; + if (!assoc) + continue; + if (!assoc_line_out) + assoc_line_out = assoc; + else if (assoc_line_out != assoc) + continue; + if (cfg->line_outs >= ARRAY_SIZE(cfg->line_out_pins)) + continue; + cfg->line_out_pins[cfg->line_outs] = nid; + sequences_line_out[cfg->line_outs] = seq; + cfg->line_outs++; + break; + case AC_JACK_SPEAKER: + seq = get_defcfg_sequence(def_conf); + assoc = get_defcfg_association(def_conf); + if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins)) + continue; + cfg->speaker_pins[cfg->speaker_outs] = nid; + sequences_speaker[cfg->speaker_outs] = (assoc << 4) | seq; + cfg->speaker_outs++; + break; + case AC_JACK_HP_OUT: + seq = get_defcfg_sequence(def_conf); + assoc = get_defcfg_association(def_conf); + if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins)) + continue; + cfg->hp_pins[cfg->hp_outs] = nid; + sequences_hp[cfg->hp_outs] = (assoc << 4) | seq; + cfg->hp_outs++; + break; + case AC_JACK_MIC_IN: + add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_MIC); + break; + case AC_JACK_LINE_IN: + add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_LINE_IN); + break; + case AC_JACK_CD: + add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_CD); + break; + case AC_JACK_AUX: + add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_AUX); + break; + case AC_JACK_SPDIF_OUT: + case AC_JACK_DIG_OTHER_OUT: + if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins)) + continue; + cfg->dig_out_pins[cfg->dig_outs] = nid; + cfg->dig_out_type[cfg->dig_outs] = + (loc == AC_JACK_LOC_HDMI) ? + HDA_PCM_TYPE_HDMI : HDA_PCM_TYPE_SPDIF; + cfg->dig_outs++; + break; + case AC_JACK_SPDIF_IN: + case AC_JACK_DIG_OTHER_IN: + cfg->dig_in_pin = nid; + if (loc == AC_JACK_LOC_HDMI) + cfg->dig_in_type = HDA_PCM_TYPE_HDMI; + else + cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; + break; + } + } + + /* FIX-UP: + * If no line-out is defined but multiple HPs are found, + * some of them might be the real line-outs. + */ + if (!cfg->line_outs && cfg->hp_outs > 1 && + !(cond_flags & HDA_PINCFG_NO_HP_FIXUP)) { + int i = 0; + while (i < cfg->hp_outs) { + /* The real HPs should have the sequence 0x0f */ + if ((sequences_hp[i] & 0x0f) == 0x0f) { + i++; + continue; + } + /* Move it to the line-out table */ + cfg->line_out_pins[cfg->line_outs] = cfg->hp_pins[i]; + sequences_line_out[cfg->line_outs] = sequences_hp[i]; + cfg->line_outs++; + cfg->hp_outs--; + memmove(cfg->hp_pins + i, cfg->hp_pins + i + 1, + sizeof(cfg->hp_pins[0]) * (cfg->hp_outs - i)); + memmove(sequences_hp + i, sequences_hp + i + 1, + sizeof(sequences_hp[0]) * (cfg->hp_outs - i)); + } + memset(cfg->hp_pins + cfg->hp_outs, 0, + sizeof(hda_nid_t) * (AUTO_CFG_MAX_OUTS - cfg->hp_outs)); + if (!cfg->hp_outs) + cfg->line_out_type = AUTO_PIN_HP_OUT; + + } + + /* sort by sequence */ + sort_pins_by_sequence(cfg->line_out_pins, sequences_line_out, + cfg->line_outs); + sort_pins_by_sequence(cfg->speaker_pins, sequences_speaker, + cfg->speaker_outs); + sort_pins_by_sequence(cfg->hp_pins, sequences_hp, + cfg->hp_outs); + + /* + * FIX-UP: if no line-outs are detected, try to use speaker or HP pin + * as a primary output + */ + if (!cfg->line_outs && + !(cond_flags & HDA_PINCFG_NO_LO_FIXUP)) { + if (cfg->speaker_outs) { + cfg->line_outs = cfg->speaker_outs; + memcpy(cfg->line_out_pins, cfg->speaker_pins, + sizeof(cfg->speaker_pins)); + cfg->speaker_outs = 0; + memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins)); + cfg->line_out_type = AUTO_PIN_SPEAKER_OUT; + } else if (cfg->hp_outs) { + cfg->line_outs = cfg->hp_outs; + memcpy(cfg->line_out_pins, cfg->hp_pins, + sizeof(cfg->hp_pins)); + cfg->hp_outs = 0; + memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins)); + cfg->line_out_type = AUTO_PIN_HP_OUT; + } + } + + reorder_outputs(cfg->line_outs, cfg->line_out_pins); + reorder_outputs(cfg->hp_outs, cfg->hp_pins); + reorder_outputs(cfg->speaker_outs, cfg->speaker_pins); + + sort_autocfg_input_pins(cfg); + + /* + * debug prints of the parsed results + */ + snd_printd("autoconfig: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x) type:%s\n", + cfg->line_outs, cfg->line_out_pins[0], cfg->line_out_pins[1], + cfg->line_out_pins[2], cfg->line_out_pins[3], + cfg->line_out_pins[4], + cfg->line_out_type == AUTO_PIN_HP_OUT ? "hp" : + (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT ? + "speaker" : "line")); + snd_printd(" speaker_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", + cfg->speaker_outs, cfg->speaker_pins[0], + cfg->speaker_pins[1], cfg->speaker_pins[2], + cfg->speaker_pins[3], cfg->speaker_pins[4]); + snd_printd(" hp_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", + cfg->hp_outs, cfg->hp_pins[0], + cfg->hp_pins[1], cfg->hp_pins[2], + cfg->hp_pins[3], cfg->hp_pins[4]); + snd_printd(" mono: mono_out=0x%x\n", cfg->mono_out_pin); + if (cfg->dig_outs) + snd_printd(" dig-out=0x%x/0x%x\n", + cfg->dig_out_pins[0], cfg->dig_out_pins[1]); + snd_printd(" inputs:"); + for (i = 0; i < cfg->num_inputs; i++) { + snd_printd(" %s=0x%x", + hda_get_autocfg_input_label(codec, cfg, i), + cfg->inputs[i].pin); + } + snd_printd("\n"); + if (cfg->dig_in_pin) + snd_printd(" dig-in=0x%x\n", cfg->dig_in_pin); + + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_parse_pin_defcfg); + +int snd_hda_get_input_pin_attr(unsigned int def_conf) +{ + unsigned int loc = get_defcfg_location(def_conf); + unsigned int conn = get_defcfg_connect(def_conf); + if (conn == AC_JACK_PORT_NONE) + return INPUT_PIN_ATTR_UNUSED; + /* Windows may claim the internal mic to be BOTH, too */ + if (conn == AC_JACK_PORT_FIXED || conn == AC_JACK_PORT_BOTH) + return INPUT_PIN_ATTR_INT; + if ((loc & 0x30) == AC_JACK_LOC_INTERNAL) + return INPUT_PIN_ATTR_INT; + if ((loc & 0x30) == AC_JACK_LOC_SEPARATE) + return INPUT_PIN_ATTR_DOCK; + if (loc == AC_JACK_LOC_REAR) + return INPUT_PIN_ATTR_REAR; + if (loc == AC_JACK_LOC_FRONT) + return INPUT_PIN_ATTR_FRONT; + return INPUT_PIN_ATTR_NORMAL; +} +EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_attr); + +/** + * hda_get_input_pin_label - Give a label for the given input pin + * + * When check_location is true, the function checks the pin location + * for mic and line-in pins, and set an appropriate prefix like "Front", + * "Rear", "Internal". + */ + +static const char *hda_get_input_pin_label(struct hda_codec *codec, + hda_nid_t pin, bool check_location) +{ + unsigned int def_conf; + static const char * const mic_names[] = { + "Internal Mic", "Dock Mic", "Mic", "Front Mic", "Rear Mic", + }; + int attr; + + def_conf = snd_hda_codec_get_pincfg(codec, pin); + + switch (get_defcfg_device(def_conf)) { + case AC_JACK_MIC_IN: + if (!check_location) + return "Mic"; + attr = snd_hda_get_input_pin_attr(def_conf); + if (!attr) + return "None"; + return mic_names[attr - 1]; + case AC_JACK_LINE_IN: + if (!check_location) + return "Line"; + attr = snd_hda_get_input_pin_attr(def_conf); + if (!attr) + return "None"; + if (attr == INPUT_PIN_ATTR_DOCK) + return "Dock Line"; + return "Line"; + case AC_JACK_AUX: + return "Aux"; + case AC_JACK_CD: + return "CD"; + case AC_JACK_SPDIF_IN: + return "SPDIF In"; + case AC_JACK_DIG_OTHER_IN: + return "Digital In"; + default: + return "Misc"; + } +} + +/* Check whether the location prefix needs to be added to the label. + * If all mic-jacks are in the same location (e.g. rear panel), we don't + * have to put "Front" prefix to each label. In such a case, returns false. + */ +static int check_mic_location_need(struct hda_codec *codec, + const struct auto_pin_cfg *cfg, + int input) +{ + unsigned int defc; + int i, attr, attr2; + + defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[input].pin); + attr = snd_hda_get_input_pin_attr(defc); + /* for internal or docking mics, we need locations */ + if (attr <= INPUT_PIN_ATTR_NORMAL) + return 1; + + attr = 0; + for (i = 0; i < cfg->num_inputs; i++) { + defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[i].pin); + attr2 = snd_hda_get_input_pin_attr(defc); + if (attr2 >= INPUT_PIN_ATTR_NORMAL) { + if (attr && attr != attr2) + return 1; /* different locations found */ + attr = attr2; + } + } + return 0; +} + +/** + * hda_get_autocfg_input_label - Get a label for the given input + * + * Get a label for the given input pin defined by the autocfg item. + * Unlike hda_get_input_pin_label(), this function checks all inputs + * defined in autocfg and avoids the redundant mic/line prefix as much as + * possible. + */ +const char *hda_get_autocfg_input_label(struct hda_codec *codec, + const struct auto_pin_cfg *cfg, + int input) +{ + int type = cfg->inputs[input].type; + int has_multiple_pins = 0; + + if ((input > 0 && cfg->inputs[input - 1].type == type) || + (input < cfg->num_inputs - 1 && cfg->inputs[input + 1].type == type)) + has_multiple_pins = 1; + if (has_multiple_pins && type == AUTO_PIN_MIC) + has_multiple_pins &= check_mic_location_need(codec, cfg, input); + return hda_get_input_pin_label(codec, cfg->inputs[input].pin, + has_multiple_pins); +} +EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label); + +/* return the position of NID in the list, or -1 if not found */ +static int find_idx_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums) +{ + int i; + for (i = 0; i < nums; i++) + if (list[i] == nid) + return i; + return -1; +} + +/* get a unique suffix or an index number */ +static const char *check_output_sfx(hda_nid_t nid, const hda_nid_t *pins, + int num_pins, int *indexp) +{ + static const char * const channel_sfx[] = { + " Front", " Surround", " CLFE", " Side" + }; + int i; + + i = find_idx_in_nid_list(nid, pins, num_pins); + if (i < 0) + return NULL; + if (num_pins == 1) + return ""; + if (num_pins > ARRAY_SIZE(channel_sfx)) { + if (indexp) + *indexp = i; + return ""; + } + return channel_sfx[i]; +} + +static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, + const struct auto_pin_cfg *cfg, + const char *name, char *label, int maxlen, + int *indexp) +{ + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); + int attr = snd_hda_get_input_pin_attr(def_conf); + const char *pfx = "", *sfx = ""; + + /* handle as a speaker if it's a fixed line-out */ + if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT) + name = "Speaker"; + /* check the location */ + switch (attr) { + case INPUT_PIN_ATTR_DOCK: + pfx = "Dock "; + break; + case INPUT_PIN_ATTR_FRONT: + pfx = "Front "; + break; + } + if (cfg) { + /* try to give a unique suffix if needed */ + sfx = check_output_sfx(nid, cfg->line_out_pins, cfg->line_outs, + indexp); + if (!sfx) + sfx = check_output_sfx(nid, cfg->speaker_pins, cfg->speaker_outs, + indexp); + if (!sfx) { + /* don't add channel suffix for Headphone controls */ + int idx = find_idx_in_nid_list(nid, cfg->hp_pins, + cfg->hp_outs); + if (idx >= 0) + *indexp = idx; + sfx = ""; + } + } + snprintf(label, maxlen, "%s%s%s", pfx, name, sfx); + return 1; +} + +/** + * snd_hda_get_pin_label - Get a label for the given I/O pin + * + * Get a label for the given pin. This function works for both input and + * output pins. When @cfg is given as non-NULL, the function tries to get + * an optimized label using hda_get_autocfg_input_label(). + * + * This function tries to give a unique label string for the pin as much as + * possible. For example, when the multiple line-outs are present, it adds + * the channel suffix like "Front", "Surround", etc (only when @cfg is given). + * If no unique name with a suffix is available and @indexp is non-NULL, the + * index number is stored in the pointer. + */ +int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, + const struct auto_pin_cfg *cfg, + char *label, int maxlen, int *indexp) +{ + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); + const char *name = NULL; + int i; + + if (indexp) + *indexp = 0; + if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) + return 0; + + switch (get_defcfg_device(def_conf)) { + case AC_JACK_LINE_OUT: + return fill_audio_out_name(codec, nid, cfg, "Line Out", + label, maxlen, indexp); + case AC_JACK_SPEAKER: + return fill_audio_out_name(codec, nid, cfg, "Speaker", + label, maxlen, indexp); + case AC_JACK_HP_OUT: + return fill_audio_out_name(codec, nid, cfg, "Headphone", + label, maxlen, indexp); + case AC_JACK_SPDIF_OUT: + case AC_JACK_DIG_OTHER_OUT: + if (get_defcfg_location(def_conf) == AC_JACK_LOC_HDMI) + name = "HDMI"; + else + name = "SPDIF"; + if (cfg && indexp) { + i = find_idx_in_nid_list(nid, cfg->dig_out_pins, + cfg->dig_outs); + if (i >= 0) + *indexp = i; + } + break; + default: + if (cfg) { + for (i = 0; i < cfg->num_inputs; i++) { + if (cfg->inputs[i].pin != nid) + continue; + name = hda_get_autocfg_input_label(codec, cfg, i); + if (name) + break; + } + } + if (!name) + name = hda_get_input_pin_label(codec, nid, true); + break; + } + if (!name) + return 0; + strlcpy(label, name, maxlen); + return 1; +} +EXPORT_SYMBOL_HDA(snd_hda_get_pin_label); + +int snd_hda_gen_add_verbs(struct hda_gen_spec *spec, + const struct hda_verb *list) +{ + const struct hda_verb **v; + v = snd_array_new(&spec->verbs); + if (!v) + return -ENOMEM; + *v = list; + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_gen_add_verbs); + +void snd_hda_gen_apply_verbs(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + int i; + for (i = 0; i < spec->verbs.used; i++) { + struct hda_verb **v = snd_array_elem(&spec->verbs, i); + snd_hda_sequence_write(codec, *v); + } +} +EXPORT_SYMBOL_HDA(snd_hda_gen_apply_verbs); + +void snd_hda_apply_pincfgs(struct hda_codec *codec, + const struct hda_pintbl *cfg) +{ + for (; cfg->nid; cfg++) + snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); +} +EXPORT_SYMBOL_HDA(snd_hda_apply_pincfgs); + +void snd_hda_apply_fixup(struct hda_codec *codec, int action) +{ + struct hda_gen_spec *spec = codec->spec; + int id = spec->fixup_id; +#ifdef CONFIG_SND_DEBUG_VERBOSE + const char *modelname = spec->fixup_name; +#endif + int depth = 0; + + if (!spec->fixup_list) + return; + + while (id >= 0) { + const struct hda_fixup *fix = spec->fixup_list + id; + + switch (fix->type) { + case HDA_FIXUP_PINS: + if (action != HDA_FIXUP_ACT_PRE_PROBE || !fix->v.pins) + break; + snd_printdd(KERN_INFO SFX + "%s: Apply pincfg for %s\n", + codec->chip_name, modelname); + snd_hda_apply_pincfgs(codec, fix->v.pins); + break; + case HDA_FIXUP_VERBS: + if (action != HDA_FIXUP_ACT_PROBE || !fix->v.verbs) + break; + snd_printdd(KERN_INFO SFX + "%s: Apply fix-verbs for %s\n", + codec->chip_name, modelname); + snd_hda_gen_add_verbs(codec->spec, fix->v.verbs); + break; + case HDA_FIXUP_FUNC: + if (!fix->v.func) + break; + snd_printdd(KERN_INFO SFX + "%s: Apply fix-func for %s\n", + codec->chip_name, modelname); + fix->v.func(codec, fix, action); + break; + default: + snd_printk(KERN_ERR SFX + "%s: Invalid fixup type %d\n", + codec->chip_name, fix->type); + break; + } + if (!fix->chained) + break; + if (++depth > 10) + break; + id = fix->chain_id; + } +} +EXPORT_SYMBOL_HDA(snd_hda_apply_fixup); + +void snd_hda_pick_fixup(struct hda_codec *codec, + const struct hda_model_fixup *models, + const struct snd_pci_quirk *quirk, + const struct hda_fixup *fixlist) +{ + struct hda_gen_spec *spec = codec->spec; + const struct snd_pci_quirk *q; + int id = -1; + const char *name = NULL; + + /* when model=nofixup is given, don't pick up any fixups */ + if (codec->modelname && !strcmp(codec->modelname, "nofixup")) { + spec->fixup_list = NULL; + spec->fixup_id = -1; + return; + } + + if (codec->modelname && models) { + while (models->name) { + if (!strcmp(codec->modelname, models->name)) { + id = models->id; + name = models->name; + break; + } + models++; + } + } + if (id < 0 && quirk) { + q = snd_pci_quirk_lookup(codec->bus->pci, quirk); + if (q) { + id = q->value; +#ifdef CONFIG_SND_DEBUG_VERBOSE + name = q->name; +#endif + } + } + if (id < 0 && quirk) { + for (q = quirk; q->subvendor; q++) { + unsigned int vendorid = + q->subdevice | (q->subvendor << 16); + if (vendorid == codec->subsystem_id) { + id = q->value; +#ifdef CONFIG_SND_DEBUG_VERBOSE + name = q->name; +#endif + break; + } + } + } + + spec->fixup_id = id; + if (id >= 0) { + spec->fixup_list = fixlist; + spec->fixup_name = name; + } +} +EXPORT_SYMBOL_HDA(snd_hda_pick_fixup); diff --git a/sound/pci/hda/hda_auto_parser.h b/sound/pci/hda/hda_auto_parser.h new file mode 100644 index 000000000000..632ad0ad3007 --- /dev/null +++ b/sound/pci/hda/hda_auto_parser.h @@ -0,0 +1,170 @@ +/* + * BIOS auto-parser helper functions for HD-audio + * + * Copyright (c) 2012 Takashi Iwai <tiwai@suse.de> + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#ifndef __SOUND_HDA_AUTO_PARSER_H +#define __SOUND_HDA_AUTO_PARSER_H + +/* + * Helper for automatic pin configuration + */ + +enum { + AUTO_PIN_MIC, + AUTO_PIN_LINE_IN, + AUTO_PIN_CD, + AUTO_PIN_AUX, + AUTO_PIN_LAST +}; + +enum { + AUTO_PIN_LINE_OUT, + AUTO_PIN_SPEAKER_OUT, + AUTO_PIN_HP_OUT +}; + +#define AUTO_CFG_MAX_OUTS HDA_MAX_OUTS +#define AUTO_CFG_MAX_INS 8 + +struct auto_pin_cfg_item { + hda_nid_t pin; + int type; +}; + +struct auto_pin_cfg; +const char *hda_get_autocfg_input_label(struct hda_codec *codec, + const struct auto_pin_cfg *cfg, + int input); +int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, + const struct auto_pin_cfg *cfg, + char *label, int maxlen, int *indexp); + +enum { + INPUT_PIN_ATTR_UNUSED, /* pin not connected */ + INPUT_PIN_ATTR_INT, /* internal mic/line-in */ + INPUT_PIN_ATTR_DOCK, /* docking mic/line-in */ + INPUT_PIN_ATTR_NORMAL, /* mic/line-in jack */ + INPUT_PIN_ATTR_FRONT, /* mic/line-in jack in front */ + INPUT_PIN_ATTR_REAR, /* mic/line-in jack in rear */ +}; + +int snd_hda_get_input_pin_attr(unsigned int def_conf); + +struct auto_pin_cfg { + int line_outs; + /* sorted in the order of Front/Surr/CLFE/Side */ + hda_nid_t line_out_pins[AUTO_CFG_MAX_OUTS]; + int speaker_outs; + hda_nid_t speaker_pins[AUTO_CFG_MAX_OUTS]; + int hp_outs; + int line_out_type; /* AUTO_PIN_XXX_OUT */ + hda_nid_t hp_pins[AUTO_CFG_MAX_OUTS]; + int num_inputs; + struct auto_pin_cfg_item inputs[AUTO_CFG_MAX_INS]; + int dig_outs; + hda_nid_t dig_out_pins[2]; + hda_nid_t dig_in_pin; + hda_nid_t mono_out_pin; + int dig_out_type[2]; /* HDA_PCM_TYPE_XXX */ + int dig_in_type; /* HDA_PCM_TYPE_XXX */ +}; + +/* bit-flags for snd_hda_parse_pin_def_config() behavior */ +#define HDA_PINCFG_NO_HP_FIXUP (1 << 0) /* no HP-split */ +#define HDA_PINCFG_NO_LO_FIXUP (1 << 1) /* don't take other outs as LO */ + +int snd_hda_parse_pin_defcfg(struct hda_codec *codec, + struct auto_pin_cfg *cfg, + const hda_nid_t *ignore_nids, + unsigned int cond_flags); + +/* older function */ +#define snd_hda_parse_pin_def_config(codec, cfg, ignore) \ + snd_hda_parse_pin_defcfg(codec, cfg, ignore, 0) + +/* + */ + +struct hda_gen_spec { + /* fix-up list */ + int fixup_id; + const struct hda_fixup *fixup_list; + const char *fixup_name; + + /* additional init verbs */ + struct snd_array verbs; +}; + + +/* + * Fix-up pin default configurations and add default verbs + */ + +struct hda_pintbl { + hda_nid_t nid; + u32 val; +}; + +struct hda_model_fixup { + const int id; + const char *name; +}; + +struct hda_fixup { + int type; + bool chained; + int chain_id; + union { + const struct hda_pintbl *pins; + const struct hda_verb *verbs; + void (*func)(struct hda_codec *codec, + const struct hda_fixup *fix, + int action); + } v; +}; + +/* fixup types */ +enum { + HDA_FIXUP_INVALID, + HDA_FIXUP_PINS, + HDA_FIXUP_VERBS, + HDA_FIXUP_FUNC, +}; + +/* fixup action definitions */ +enum { + HDA_FIXUP_ACT_PRE_PROBE, + HDA_FIXUP_ACT_PROBE, + HDA_FIXUP_ACT_INIT, + HDA_FIXUP_ACT_BUILD, +}; + +int snd_hda_gen_add_verbs(struct hda_gen_spec *spec, + const struct hda_verb *list); +void snd_hda_gen_apply_verbs(struct hda_codec *codec); +void snd_hda_apply_pincfgs(struct hda_codec *codec, + const struct hda_pintbl *cfg); +void snd_hda_apply_fixup(struct hda_codec *codec, int action); +void snd_hda_pick_fixup(struct hda_codec *codec, + const struct hda_model_fixup *models, + const struct snd_pci_quirk *quirk, + const struct hda_fixup *fixlist); + +static inline void snd_hda_gen_init(struct hda_gen_spec *spec) +{ + snd_array_init(&spec->verbs, sizeof(struct hda_verb *), 8); +} + +static inline void snd_hda_gen_free(struct hda_gen_spec *spec) +{ + snd_array_free(&spec->verbs); +} + +#endif /* __SOUND_HDA_AUTO_PARSER_H */ diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 60738e52b8f9..0bc2315b181d 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -162,50 +162,20 @@ static int snd_hda_do_attach(struct hda_beep *beep) return 0; } -static void snd_hda_do_register(struct work_struct *work) -{ - struct hda_beep *beep = - container_of(work, struct hda_beep, register_work); - - mutex_lock(&beep->mutex); - if (beep->enabled && !beep->dev) - snd_hda_do_attach(beep); - mutex_unlock(&beep->mutex); -} - -static void snd_hda_do_unregister(struct work_struct *work) -{ - struct hda_beep *beep = - container_of(work, struct hda_beep, unregister_work.work); - - mutex_lock(&beep->mutex); - if (!beep->enabled && beep->dev) - snd_hda_do_detach(beep); - mutex_unlock(&beep->mutex); -} - int snd_hda_enable_beep_device(struct hda_codec *codec, int enable) { struct hda_beep *beep = codec->beep; - enable = !!enable; - if (beep == NULL) + if (!beep) return 0; + enable = !!enable; if (beep->enabled != enable) { beep->enabled = enable; if (!enable) { + cancel_work_sync(&beep->beep_work); /* turn off beep */ snd_hda_codec_write(beep->codec, beep->nid, 0, AC_VERB_SET_BEEP_CONTROL, 0); } - if (beep->mode == HDA_BEEP_MODE_SWREG) { - if (enable) { - cancel_delayed_work(&beep->unregister_work); - schedule_work(&beep->register_work); - } else { - schedule_delayed_work(&beep->unregister_work, - HZ); - } - } return 1; } return 0; @@ -215,6 +185,7 @@ EXPORT_SYMBOL_HDA(snd_hda_enable_beep_device); int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) { struct hda_beep *beep; + int err; if (!snd_hda_get_bool_hint(codec, "beep")) return 0; /* disabled explicitly by hints */ @@ -232,21 +203,16 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) beep->nid = nid; beep->codec = codec; - beep->mode = codec->beep_mode; codec->beep = beep; - INIT_WORK(&beep->register_work, &snd_hda_do_register); - INIT_DELAYED_WORK(&beep->unregister_work, &snd_hda_do_unregister); INIT_WORK(&beep->beep_work, &snd_hda_generate_beep); mutex_init(&beep->mutex); - if (beep->mode == HDA_BEEP_MODE_ON) { - int err = snd_hda_do_attach(beep); - if (err < 0) { - kfree(beep); - codec->beep = NULL; - return err; - } + err = snd_hda_do_attach(beep); + if (err < 0) { + kfree(beep); + codec->beep = NULL; + return err; } return 0; @@ -257,8 +223,6 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) { struct hda_beep *beep = codec->beep; if (beep) { - cancel_work_sync(&beep->register_work); - cancel_delayed_work(&beep->unregister_work); if (beep->dev) snd_hda_do_detach(beep); codec->beep = NULL; @@ -266,3 +230,31 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) } } EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device); + +/* get/put callbacks for beep mute mixer switches */ +int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hda_beep *beep = codec->beep; + if (beep) { + ucontrol->value.integer.value[0] = + ucontrol->value.integer.value[1] = + beep->enabled; + return 0; + } + return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); +} +EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_get_beep); + +int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hda_beep *beep = codec->beep; + if (beep) + snd_hda_enable_beep_device(codec, + *ucontrol->value.integer.value); + return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); +} +EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep); diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index 55f0647458c7..4dc6933bc655 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -26,21 +26,16 @@ #define HDA_BEEP_MODE_OFF 0 #define HDA_BEEP_MODE_ON 1 -#define HDA_BEEP_MODE_SWREG 2 /* beep information */ struct hda_beep { struct input_dev *dev; struct hda_codec *codec; - unsigned int mode; char phys[32]; int tone; hda_nid_t nid; unsigned int enabled:1; - unsigned int request_enable:1; unsigned int linear_tone:1; /* linear tone for IDT/STAC codec */ - struct work_struct register_work; /* registration work */ - struct delayed_work unregister_work; /* unregistration work */ struct work_struct beep_work; /* scheduled task for beep event */ struct mutex mutex; }; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 841475cc13b6..88a9c20eb7a2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -334,78 +334,67 @@ static hda_nid_t *lookup_conn_list(struct snd_array *array, hda_nid_t nid) return NULL; } +/* read the connection and add to the cache */ +static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid) +{ + hda_nid_t list[HDA_MAX_CONNECTIONS]; + int len; + + len = snd_hda_get_raw_connections(codec, nid, list, ARRAY_SIZE(list)); + if (len < 0) + return len; + return snd_hda_override_conn_list(codec, nid, len, list); +} + /** - * snd_hda_get_conn_list - get connection list + * snd_hda_get_connections - copy connection list * @codec: the HDA codec * @nid: NID to parse - * @listp: the pointer to store NID list + * @conn_list: connection list array; when NULL, checks only the size + * @max_conns: max. number of connections to store * * Parses the connection list of the given widget and stores the list * of NIDs. * * Returns the number of connections, or a negative error code. */ -int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, - const hda_nid_t **listp) +int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, + hda_nid_t *conn_list, int max_conns) { struct snd_array *array = &codec->conn_lists; - int len, err; - hda_nid_t list[HDA_MAX_CONNECTIONS]; + int len; hda_nid_t *p; bool added = false; again: + mutex_lock(&codec->hash_mutex); + len = -1; /* if the connection-list is already cached, read it */ p = lookup_conn_list(array, nid); if (p) { - if (listp) - *listp = p + 2; - return p[1]; + len = p[1]; + if (conn_list && len > max_conns) { + snd_printk(KERN_ERR "hda_codec: " + "Too many connections %d for NID 0x%x\n", + len, nid); + mutex_unlock(&codec->hash_mutex); + return -EINVAL; + } + if (conn_list && len) + memcpy(conn_list, p + 2, len * sizeof(hda_nid_t)); } + mutex_unlock(&codec->hash_mutex); + if (len >= 0) + return len; if (snd_BUG_ON(added)) return -EINVAL; - /* read the connection and add to the cache */ - len = snd_hda_get_raw_connections(codec, nid, list, HDA_MAX_CONNECTIONS); + len = read_and_add_raw_conns(codec, nid); if (len < 0) return len; - err = snd_hda_override_conn_list(codec, nid, len, list); - if (err < 0) - return err; added = true; goto again; } -EXPORT_SYMBOL_HDA(snd_hda_get_conn_list); - -/** - * snd_hda_get_connections - copy connection list - * @codec: the HDA codec - * @nid: NID to parse - * @conn_list: connection list array - * @max_conns: max. number of connections to store - * - * Parses the connection list of the given widget and stores the list - * of NIDs. - * - * Returns the number of connections, or a negative error code. - */ -int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, - hda_nid_t *conn_list, int max_conns) -{ - const hda_nid_t *list; - int len = snd_hda_get_conn_list(codec, nid, &list); - - if (len <= 0) - return len; - if (len > max_conns) { - snd_printk(KERN_ERR "hda_codec: " - "Too many connections %d for NID 0x%x\n", - len, nid); - return -EINVAL; - } - memcpy(conn_list, list, len * sizeof(hda_nid_t)); - return len; -} EXPORT_SYMBOL_HDA(snd_hda_get_connections); /** @@ -543,6 +532,7 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int len, hda_nid_t *p; int i, old_used; + mutex_lock(&codec->hash_mutex); p = lookup_conn_list(array, nid); if (p) *p = -1; /* invalidate the old entry */ @@ -553,10 +543,12 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int len, for (i = 0; i < len; i++) if (!add_conn_list(array, list[i])) goto error_add; + mutex_unlock(&codec->hash_mutex); return 0; error_add: array->used = old_used; + mutex_unlock(&codec->hash_mutex); return -ENOMEM; } EXPORT_SYMBOL_HDA(snd_hda_override_conn_list); @@ -1192,6 +1184,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) { if (!codec) return; + snd_hda_jack_tbl_clear(codec); restore_init_pincfgs(codec); #ifdef CONFIG_SND_HDA_POWER_SAVE cancel_delayed_work(&codec->power_work); @@ -1200,6 +1193,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) list_del(&codec->list); snd_array_free(&codec->mixers); snd_array_free(&codec->nids); + snd_array_free(&codec->cvt_setups); snd_array_free(&codec->conn_lists); snd_array_free(&codec->spdif_out); codec->bus->caddr_tbl[codec->addr] = NULL; @@ -1255,6 +1249,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, codec->addr = codec_addr; mutex_init(&codec->spdif_mutex); mutex_init(&codec->control_mutex); + mutex_init(&codec->hash_mutex); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); @@ -1264,15 +1259,9 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); snd_array_init(&codec->conn_lists, sizeof(hda_nid_t), 64); snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16); - if (codec->bus->modelname) { - codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL); - if (!codec->modelname) { - snd_hda_codec_free(codec); - return -ENODEV; - } - } #ifdef CONFIG_SND_HDA_POWER_SAVE + spin_lock_init(&codec->power_lock); INIT_DELAYED_WORK(&codec->power_work, hda_power_work); /* snd_hda_codec_new() marks the codec as power-up, and leave it as is. * the caller has to power down appropriatley after initialization @@ -1281,6 +1270,14 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, hda_keep_power_on(codec); #endif + if (codec->bus->modelname) { + codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL); + if (!codec->modelname) { + snd_hda_codec_free(codec); + return -ENODEV; + } + } + list_add_tail(&codec->list, &bus->codec_list); bus->caddr_tbl[codec_addr] = codec; @@ -1603,6 +1600,60 @@ get_alloc_amp_hash(struct hda_codec *codec, u32 key) return (struct hda_amp_info *)get_alloc_hash(&codec->amp_cache, key); } +/* overwrite the value with the key in the caps hash */ +static int write_caps_hash(struct hda_codec *codec, u32 key, unsigned int val) +{ + struct hda_amp_info *info; + + mutex_lock(&codec->hash_mutex); + info = get_alloc_amp_hash(codec, key); + if (!info) { + mutex_unlock(&codec->hash_mutex); + return -EINVAL; + } + info->amp_caps = val; + info->head.val |= INFO_AMP_CAPS; + mutex_unlock(&codec->hash_mutex); + return 0; +} + +/* query the value from the caps hash; if not found, fetch the current + * value from the given function and store in the hash + */ +static unsigned int +query_caps_hash(struct hda_codec *codec, hda_nid_t nid, int dir, u32 key, + unsigned int (*func)(struct hda_codec *, hda_nid_t, int)) +{ + struct hda_amp_info *info; + unsigned int val; + + mutex_lock(&codec->hash_mutex); + info = get_alloc_amp_hash(codec, key); + if (!info) { + mutex_unlock(&codec->hash_mutex); + return 0; + } + if (!(info->head.val & INFO_AMP_CAPS)) { + mutex_unlock(&codec->hash_mutex); /* for reentrance */ + val = func(codec, nid, dir); + write_caps_hash(codec, key, val); + } else { + val = info->amp_caps; + mutex_unlock(&codec->hash_mutex); + } + return val; +} + +static unsigned int read_amp_cap(struct hda_codec *codec, hda_nid_t nid, + int direction) +{ + if (!(get_wcaps(codec, nid) & AC_WCAP_AMP_OVRD)) + nid = codec->afg; + return snd_hda_param_read(codec, nid, + direction == HDA_OUTPUT ? + AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP); +} + /** * query_amp_caps - query AMP capabilities * @codec: the HD-auio codec @@ -1617,22 +1668,9 @@ get_alloc_amp_hash(struct hda_codec *codec, u32 key) */ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) { - struct hda_amp_info *info; - - info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, 0)); - if (!info) - return 0; - if (!(info->head.val & INFO_AMP_CAPS)) { - if (!(get_wcaps(codec, nid) & AC_WCAP_AMP_OVRD)) - nid = codec->afg; - info->amp_caps = snd_hda_param_read(codec, nid, - direction == HDA_OUTPUT ? - AC_PAR_AMP_OUT_CAP : - AC_PAR_AMP_IN_CAP); - if (info->amp_caps) - info->head.val |= INFO_AMP_CAPS; - } - return info->amp_caps; + return query_caps_hash(codec, nid, direction, + HDA_HASH_KEY(nid, direction, 0), + read_amp_cap); } EXPORT_SYMBOL_HDA(query_amp_caps); @@ -1652,34 +1690,12 @@ EXPORT_SYMBOL_HDA(query_amp_caps); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps) { - struct hda_amp_info *info; - - info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, dir, 0)); - if (!info) - return -EINVAL; - info->amp_caps = caps; - info->head.val |= INFO_AMP_CAPS; - return 0; + return write_caps_hash(codec, HDA_HASH_KEY(nid, dir, 0), caps); } EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps); -static unsigned int -query_caps_hash(struct hda_codec *codec, hda_nid_t nid, u32 key, - unsigned int (*func)(struct hda_codec *, hda_nid_t)) -{ - struct hda_amp_info *info; - - info = get_alloc_amp_hash(codec, key); - if (!info) - return 0; - if (!info->head.val) { - info->head.val |= INFO_AMP_CAPS; - info->amp_caps = func(codec, nid); - } - return info->amp_caps; -} - -static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid) +static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid, + int dir) { return snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); } @@ -1697,7 +1713,7 @@ static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid) */ u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) { - return query_caps_hash(codec, nid, HDA_HASH_PINCAP_KEY(nid), + return query_caps_hash(codec, nid, 0, HDA_HASH_PINCAP_KEY(nid), read_pin_cap); } EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps); @@ -1715,41 +1731,47 @@ EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps); int snd_hda_override_pin_caps(struct hda_codec *codec, hda_nid_t nid, unsigned int caps) { - struct hda_amp_info *info; - info = get_alloc_amp_hash(codec, HDA_HASH_PINCAP_KEY(nid)); - if (!info) - return -ENOMEM; - info->amp_caps = caps; - info->head.val |= INFO_AMP_CAPS; - return 0; + return write_caps_hash(codec, HDA_HASH_PINCAP_KEY(nid), caps); } EXPORT_SYMBOL_HDA(snd_hda_override_pin_caps); -/* - * read the current volume to info - * if the cache exists, read the cache value. +/* read or sync the hash value with the current value; + * call within hash_mutex */ -static unsigned int get_vol_mute(struct hda_codec *codec, - struct hda_amp_info *info, hda_nid_t nid, - int ch, int direction, int index) +static struct hda_amp_info * +update_amp_hash(struct hda_codec *codec, hda_nid_t nid, int ch, + int direction, int index) { - u32 val, parm; - - if (info->head.val & INFO_AMP_VOL(ch)) - return info->vol[ch]; + struct hda_amp_info *info; + unsigned int parm, val = 0; + bool val_read = false; - parm = ch ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT; - parm |= direction == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT; - parm |= index; - val = snd_hda_codec_read(codec, nid, 0, + retry: + info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, index)); + if (!info) + return NULL; + if (!(info->head.val & INFO_AMP_VOL(ch))) { + if (!val_read) { + mutex_unlock(&codec->hash_mutex); + parm = ch ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT; + parm |= direction == HDA_OUTPUT ? + AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT; + parm |= index; + val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_AMP_GAIN_MUTE, parm); - info->vol[ch] = val & 0xff; - info->head.val |= INFO_AMP_VOL(ch); - return info->vol[ch]; + val &= 0xff; + val_read = true; + mutex_lock(&codec->hash_mutex); + goto retry; + } + info->vol[ch] = val; + info->head.val |= INFO_AMP_VOL(ch); + } + return info; } /* - * write the current volume in info to the h/w and update the cache + * write the current volume in info to the h/w */ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info, hda_nid_t nid, int ch, int direction, int index, @@ -1766,7 +1788,6 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info, else parm |= val; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm); - info->vol[ch] = val; } /** @@ -1783,10 +1804,14 @@ int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int index) { struct hda_amp_info *info; - info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, index)); - if (!info) - return 0; - return get_vol_mute(codec, info, nid, ch, direction, index); + unsigned int val = 0; + + mutex_lock(&codec->hash_mutex); + info = update_amp_hash(codec, nid, ch, direction, index); + if (info) + val = info->vol[ch]; + mutex_unlock(&codec->hash_mutex); + return val; } EXPORT_SYMBOL_HDA(snd_hda_codec_amp_read); @@ -1808,15 +1833,23 @@ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, { struct hda_amp_info *info; - info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, idx)); - if (!info) - return 0; if (snd_BUG_ON(mask & ~0xff)) mask &= 0xff; val &= mask; - val |= get_vol_mute(codec, info, nid, ch, direction, idx) & ~mask; - if (info->vol[ch] == val) + + mutex_lock(&codec->hash_mutex); + info = update_amp_hash(codec, nid, ch, direction, idx); + if (!info) { + mutex_unlock(&codec->hash_mutex); + return 0; + } + val |= info->vol[ch] & ~mask; + if (info->vol[ch] == val) { + mutex_unlock(&codec->hash_mutex); return 0; + } + info->vol[ch] = val; + mutex_unlock(&codec->hash_mutex); put_vol_mute(codec, info, nid, ch, direction, idx, val); return 1; } @@ -2208,24 +2241,50 @@ void snd_hda_ctls_clear(struct hda_codec *codec) /* pseudo device locking * toggle card->shutdown to allow/disallow the device access (as a hack) */ -static int hda_lock_devices(struct snd_card *card) +int snd_hda_lock_devices(struct hda_bus *bus) { + struct snd_card *card = bus->card; + struct hda_codec *codec; + spin_lock(&card->files_lock); - if (card->shutdown) { - spin_unlock(&card->files_lock); - return -EINVAL; - } + if (card->shutdown) + goto err_unlock; card->shutdown = 1; + if (!list_empty(&card->ctl_files)) + goto err_clear; + + list_for_each_entry(codec, &bus->codec_list, list) { + int pcm; + for (pcm = 0; pcm < codec->num_pcms; pcm++) { + struct hda_pcm *cpcm = &codec->pcm_info[pcm]; + if (!cpcm->pcm) + continue; + if (cpcm->pcm->streams[0].substream_opened || + cpcm->pcm->streams[1].substream_opened) + goto err_clear; + } + } spin_unlock(&card->files_lock); return 0; + + err_clear: + card->shutdown = 0; + err_unlock: + spin_unlock(&card->files_lock); + return -EINVAL; } +EXPORT_SYMBOL_HDA(snd_hda_lock_devices); -static void hda_unlock_devices(struct snd_card *card) +void snd_hda_unlock_devices(struct hda_bus *bus) { + struct snd_card *card = bus->card; + + card = bus->card; spin_lock(&card->files_lock); card->shutdown = 0; spin_unlock(&card->files_lock); } +EXPORT_SYMBOL_HDA(snd_hda_unlock_devices); /** * snd_hda_codec_reset - Clear all objects assigned to the codec @@ -2239,32 +2298,21 @@ static void hda_unlock_devices(struct snd_card *card) */ int snd_hda_codec_reset(struct hda_codec *codec) { - struct snd_card *card = codec->bus->card; - int i, pcm; + struct hda_bus *bus = codec->bus; + struct snd_card *card = bus->card; + int i; - if (hda_lock_devices(card) < 0) + if (snd_hda_lock_devices(bus) < 0) return -EBUSY; - /* check whether the codec isn't used by any mixer or PCM streams */ - if (!list_empty(&card->ctl_files)) { - hda_unlock_devices(card); - return -EBUSY; - } - for (pcm = 0; pcm < codec->num_pcms; pcm++) { - struct hda_pcm *cpcm = &codec->pcm_info[pcm]; - if (!cpcm->pcm) - continue; - if (cpcm->pcm->streams[0].substream_opened || - cpcm->pcm->streams[1].substream_opened) { - hda_unlock_devices(card); - return -EBUSY; - } - } /* OK, let it free */ #ifdef CONFIG_SND_HDA_POWER_SAVE - cancel_delayed_work(&codec->power_work); - flush_workqueue(codec->bus->workq); + cancel_delayed_work_sync(&codec->power_work); + codec->power_on = 0; + codec->power_transition = 0; + codec->power_jiffies = jiffies; + flush_workqueue(bus->workq); #endif snd_hda_ctls_clear(codec); /* relase PCMs */ @@ -2272,7 +2320,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) if (codec->pcm_info[i].pcm) { snd_device_free(card, codec->pcm_info[i].pcm); clear_bit(codec->pcm_info[i].device, - codec->bus->pcm_dev_bits); + bus->pcm_dev_bits); } } if (codec->patch_ops.free) @@ -2287,6 +2335,8 @@ int snd_hda_codec_reset(struct hda_codec *codec) /* free only driver_pins so that init_pins + user_pins are restored */ snd_array_free(&codec->driver_pins); restore_pincfgs(codec); + snd_array_free(&codec->cvt_setups); + snd_array_free(&codec->spdif_out); codec->num_pcms = 0; codec->pcm_info = NULL; codec->preset = NULL; @@ -2297,7 +2347,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) codec->owner = NULL; /* allow device access again */ - hda_unlock_devices(card); + snd_hda_unlock_devices(bus); return 0; } @@ -2626,25 +2676,6 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put); -#ifdef CONFIG_SND_HDA_INPUT_BEEP -/** - * snd_hda_mixer_amp_switch_put_beep - Put callback for a beep AMP switch - * - * This function calls snd_hda_enable_beep_device(), which behaves differently - * depending on beep_mode option. - */ -int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - long *valp = ucontrol->value.integer.value; - - snd_hda_enable_beep_device(codec, *valp); - return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); -} -EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep); -#endif /* CONFIG_SND_HDA_INPUT_BEEP */ - /* * bound volume controls * @@ -2859,12 +2890,15 @@ static int snd_hda_spdif_default_get(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); int idx = kcontrol->private_value; - struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx); + struct hda_spdif_out *spdif; + mutex_lock(&codec->spdif_mutex); + spdif = snd_array_elem(&codec->spdif_out, idx); ucontrol->value.iec958.status[0] = spdif->status & 0xff; ucontrol->value.iec958.status[1] = (spdif->status >> 8) & 0xff; ucontrol->value.iec958.status[2] = (spdif->status >> 16) & 0xff; ucontrol->value.iec958.status[3] = (spdif->status >> 24) & 0xff; + mutex_unlock(&codec->spdif_mutex); return 0; } @@ -2950,12 +2984,14 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); int idx = kcontrol->private_value; - struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx); - hda_nid_t nid = spdif->nid; + struct hda_spdif_out *spdif; + hda_nid_t nid; unsigned short val; int change; mutex_lock(&codec->spdif_mutex); + spdif = snd_array_elem(&codec->spdif_out, idx); + nid = spdif->nid; spdif->status = ucontrol->value.iec958.status[0] | ((unsigned int)ucontrol->value.iec958.status[1] << 8) | ((unsigned int)ucontrol->value.iec958.status[2] << 16) | @@ -2977,9 +3013,12 @@ static int snd_hda_spdif_out_switch_get(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); int idx = kcontrol->private_value; - struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx); + struct hda_spdif_out *spdif; + mutex_lock(&codec->spdif_mutex); + spdif = snd_array_elem(&codec->spdif_out, idx); ucontrol->value.integer.value[0] = spdif->ctls & AC_DIG1_ENABLE; + mutex_unlock(&codec->spdif_mutex); return 0; } @@ -2999,12 +3038,14 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); int idx = kcontrol->private_value; - struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx); - hda_nid_t nid = spdif->nid; + struct hda_spdif_out *spdif; + hda_nid_t nid; unsigned short val; int change; mutex_lock(&codec->spdif_mutex); + spdif = snd_array_elem(&codec->spdif_out, idx); + nid = spdif->nid; val = spdif->ctls & ~AC_DIG1_ENABLE; if (ucontrol->value.integer.value[0]) val |= AC_DIG1_ENABLE; @@ -3092,6 +3133,9 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_create_spdif_out_ctls); +/* get the hda_spdif_out entry from the given NID + * call within spdif_mutex lock + */ struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec, hda_nid_t nid) { @@ -3108,9 +3152,10 @@ EXPORT_SYMBOL_HDA(snd_hda_spdif_out_of_nid); void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx) { - struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx); + struct hda_spdif_out *spdif; mutex_lock(&codec->spdif_mutex); + spdif = snd_array_elem(&codec->spdif_out, idx); spdif->nid = (u16)-1; mutex_unlock(&codec->spdif_mutex); } @@ -3118,10 +3163,11 @@ EXPORT_SYMBOL_HDA(snd_hda_spdif_ctls_unassign); void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid) { - struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx); + struct hda_spdif_out *spdif; unsigned short val; mutex_lock(&codec->spdif_mutex); + spdif = snd_array_elem(&codec->spdif_out, idx); if (spdif->nid != nid) { spdif->nid = nid; val = spdif->ctls; @@ -3444,22 +3490,52 @@ void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, EXPORT_SYMBOL_HDA(snd_hda_codec_set_power_to_all); /* + * supported power states check + */ +static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE); + + if (sup < 0) + return false; + if (sup & power_state) + return true; + else + return false; +} + +/* * set power state of the codec */ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state) { + int count; + unsigned int state; + if (codec->patch_ops.set_power_state) { codec->patch_ops.set_power_state(codec, fg, power_state); return; } /* this delay seems necessary to avoid click noise at power-down */ - if (power_state == AC_PWRST_D3) - msleep(100); - snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, - power_state); - snd_hda_codec_set_power_to_all(codec, fg, power_state, true); + if (power_state == AC_PWRST_D3) { + /* transition time less than 10ms for power down */ + bool epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS); + msleep(epss ? 10 : 100); + } + + /* repeat power states setting at most 10 times*/ + for (count = 0; count < 10; count++) { + snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, + power_state); + snd_hda_codec_set_power_to_all(codec, fg, power_state, true); + state = snd_hda_codec_read(codec, fg, 0, + AC_VERB_GET_POWER_STATE, 0); + if (!(state & AC_PWRST_ERROR)) + break; + } } #ifdef CONFIG_SND_HDA_HWDEP @@ -3480,17 +3556,20 @@ static inline void hda_exec_init_verbs(struct hda_codec *codec) {} static void hda_call_codec_suspend(struct hda_codec *codec) { if (codec->patch_ops.suspend) - codec->patch_ops.suspend(codec, PMSG_SUSPEND); + codec->patch_ops.suspend(codec); hda_cleanup_all_streams(codec); hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_D3); #ifdef CONFIG_SND_HDA_POWER_SAVE - snd_hda_update_power_acct(codec); cancel_delayed_work(&codec->power_work); + spin_lock(&codec->power_lock); + snd_hda_update_power_acct(codec); + trace_hda_power_down(codec); codec->power_on = 0; codec->power_transition = 0; codec->power_jiffies = jiffies; + spin_unlock(&codec->power_lock); #endif } @@ -3499,6 +3578,10 @@ static void hda_call_codec_suspend(struct hda_codec *codec) */ static void hda_call_codec_resume(struct hda_codec *codec) { + /* set as if powered on for avoiding re-entering the resume + * in the resume / power-save sequence + */ + hda_keep_power_on(codec); hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_D0); @@ -3514,6 +3597,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); } + snd_hda_power_down(codec); /* flag down before returning */ } #endif /* CONFIG_PM */ @@ -3665,7 +3749,8 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, } EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format); -static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid) +static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid, + int dir) { unsigned int val = 0; if (nid != codec->afg && @@ -3680,11 +3765,12 @@ static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid) static unsigned int query_pcm_param(struct hda_codec *codec, hda_nid_t nid) { - return query_caps_hash(codec, nid, HDA_HASH_PARPCM_KEY(nid), + return query_caps_hash(codec, nid, 0, HDA_HASH_PARPCM_KEY(nid), get_pcm_param); } -static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid) +static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid, + int dir) { unsigned int streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM); if (!streams || streams == -1) @@ -3696,7 +3782,7 @@ static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid) static unsigned int query_stream_param(struct hda_codec *codec, hda_nid_t nid) { - return query_caps_hash(codec, nid, HDA_HASH_PARSTR_KEY(nid), + return query_caps_hash(codec, nid, 0, HDA_HASH_PARSTR_KEY(nid), get_stream_param); } @@ -3775,11 +3861,13 @@ int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, bps = 20; } } +#if 0 /* FIXME: CS4206 doesn't work, which is the only codec supporting float */ if (streams & AC_SUPFMT_FLOAT32) { formats |= SNDRV_PCM_FMTBIT_FLOAT_LE; if (!bps) bps = 32; } +#endif if (streams == AC_SUPFMT_AC3) { /* should be exclusive */ /* temporary hack: we have still no proper support @@ -4283,12 +4371,18 @@ static void hda_power_work(struct work_struct *work) container_of(work, struct hda_codec, power_work.work); struct hda_bus *bus = codec->bus; + spin_lock(&codec->power_lock); + if (codec->power_transition > 0) { /* during power-up sequence? */ + spin_unlock(&codec->power_lock); + return; + } if (!codec->power_on || codec->power_count) { codec->power_transition = 0; + spin_unlock(&codec->power_lock); return; } + spin_unlock(&codec->power_lock); - trace_hda_power_down(codec); hda_call_codec_suspend(codec); if (bus->ops.pm_notify) bus->ops.pm_notify(bus); @@ -4296,9 +4390,11 @@ static void hda_power_work(struct work_struct *work) static void hda_keep_power_on(struct hda_codec *codec) { + spin_lock(&codec->power_lock); codec->power_count++; codec->power_on = 1; codec->power_jiffies = jiffies; + spin_unlock(&codec->power_lock); } /* update the power on/off account with the current jiffies */ @@ -4312,33 +4408,80 @@ void snd_hda_update_power_acct(struct hda_codec *codec) codec->power_jiffies += delta; } -/** - * snd_hda_power_up - Power-up the codec - * @codec: HD-audio codec - * - * Increment the power-up counter and power up the hardware really when - * not turned on yet. - */ -void snd_hda_power_up(struct hda_codec *codec) +/* Transition to powered up, if wait_power_down then wait for a pending + * transition to D3 to complete. A pending D3 transition is indicated + * with power_transition == -1. */ +static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) { struct hda_bus *bus = codec->bus; + spin_lock(&codec->power_lock); codec->power_count++; - if (codec->power_on || codec->power_transition) + /* Return if power_on or transitioning to power_on, unless currently + * powering down. */ + if ((codec->power_on || codec->power_transition > 0) && + !(wait_power_down && codec->power_transition < 0)) { + spin_unlock(&codec->power_lock); return; + } + spin_unlock(&codec->power_lock); + + cancel_delayed_work_sync(&codec->power_work); + spin_lock(&codec->power_lock); + /* If the power down delayed work was cancelled above before starting, + * then there is no need to go through power up here. + */ + if (codec->power_on) { + spin_unlock(&codec->power_lock); + return; + } trace_hda_power_up(codec); snd_hda_update_power_acct(codec); codec->power_on = 1; codec->power_jiffies = jiffies; + codec->power_transition = 1; /* avoid reentrance */ + spin_unlock(&codec->power_lock); + if (bus->ops.pm_notify) bus->ops.pm_notify(bus); hda_call_codec_resume(codec); - cancel_delayed_work(&codec->power_work); + + spin_lock(&codec->power_lock); codec->power_transition = 0; + spin_unlock(&codec->power_lock); +} + +/** + * snd_hda_power_up - Power-up the codec + * @codec: HD-audio codec + * + * Increment the power-up counter and power up the hardware really when + * not turned on yet. + */ +void snd_hda_power_up(struct hda_codec *codec) +{ + __snd_hda_power_up(codec, false); } EXPORT_SYMBOL_HDA(snd_hda_power_up); +/** + * snd_hda_power_up_d3wait - Power-up the codec after waiting for any pending + * D3 transition to complete. This differs from snd_hda_power_up() when + * power_transition == -1. snd_hda_power_up sees this case as a nop, + * snd_hda_power_up_d3wait waits for the D3 transition to complete then powers + * back up. + * @codec: HD-audio codec + * + * Cancel any power down operation hapenning on the work queue, then power up. + */ +void snd_hda_power_up_d3wait(struct hda_codec *codec) +{ + /* This will cancel and wait for pending power_work to complete. */ + __snd_hda_power_up(codec, true); +} +EXPORT_SYMBOL_HDA(snd_hda_power_up_d3wait); + #define power_save(codec) \ ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) @@ -4351,14 +4494,18 @@ EXPORT_SYMBOL_HDA(snd_hda_power_up); */ void snd_hda_power_down(struct hda_codec *codec) { + spin_lock(&codec->power_lock); --codec->power_count; - if (!codec->power_on || codec->power_count || codec->power_transition) + if (!codec->power_on || codec->power_count || codec->power_transition) { + spin_unlock(&codec->power_lock); return; + } if (power_save(codec)) { - codec->power_transition = 1; /* avoid reentrance */ + codec->power_transition = -1; /* avoid reentrance */ queue_delayed_work(codec->bus->workq, &codec->power_work, msecs_to_jiffies(power_save(codec) * 1000)); } + spin_unlock(&codec->power_lock); } EXPORT_SYMBOL_HDA(snd_hda_power_down); @@ -4710,11 +4857,11 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, { const hda_nid_t *nids = mout->dac_nids; int chs = substream->runtime->channels; - struct hda_spdif_out *spdif = - snd_hda_spdif_out_of_nid(codec, mout->dig_out_nid); + struct hda_spdif_out *spdif; int i; mutex_lock(&codec->spdif_mutex); + spdif = snd_hda_spdif_out_of_nid(codec, mout->dig_out_nid); if (mout->dig_out_nid && mout->share_spdif && mout->dig_out_used != HDA_DIG_EXCLUSIVE) { if (chs == 2 && @@ -4795,601 +4942,58 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_cleanup); -/* - * Helper for automatic pin configuration - */ - -static int is_in_nid_list(hda_nid_t nid, const hda_nid_t *list) -{ - for (; *list; list++) - if (*list == nid) - return 1; - return 0; -} - - -/* - * Sort an associated group of pins according to their sequence numbers. - */ -static void sort_pins_by_sequence(hda_nid_t *pins, short *sequences, - int num_pins) -{ - int i, j; - short seq; - hda_nid_t nid; - - for (i = 0; i < num_pins; i++) { - for (j = i + 1; j < num_pins; j++) { - if (sequences[i] > sequences[j]) { - seq = sequences[i]; - sequences[i] = sequences[j]; - sequences[j] = seq; - nid = pins[i]; - pins[i] = pins[j]; - pins[j] = nid; - } - } - } -} - - -/* add the found input-pin to the cfg->inputs[] table */ -static void add_auto_cfg_input_pin(struct auto_pin_cfg *cfg, hda_nid_t nid, - int type) -{ - if (cfg->num_inputs < AUTO_CFG_MAX_INS) { - cfg->inputs[cfg->num_inputs].pin = nid; - cfg->inputs[cfg->num_inputs].type = type; - cfg->num_inputs++; - } -} - -/* sort inputs in the order of AUTO_PIN_* type */ -static void sort_autocfg_input_pins(struct auto_pin_cfg *cfg) -{ - int i, j; - - for (i = 0; i < cfg->num_inputs; i++) { - for (j = i + 1; j < cfg->num_inputs; j++) { - if (cfg->inputs[i].type > cfg->inputs[j].type) { - struct auto_pin_cfg_item tmp; - tmp = cfg->inputs[i]; - cfg->inputs[i] = cfg->inputs[j]; - cfg->inputs[j] = tmp; - } - } - } -} - -/* Reorder the surround channels - * ALSA sequence is front/surr/clfe/side - * HDA sequence is: - * 4-ch: front/surr => OK as it is - * 6-ch: front/clfe/surr - * 8-ch: front/clfe/rear/side|fc - */ -static void reorder_outputs(unsigned int nums, hda_nid_t *pins) -{ - hda_nid_t nid; - - switch (nums) { - case 3: - case 4: - nid = pins[1]; - pins[1] = pins[2]; - pins[2] = nid; - break; - } -} - -/* - * Parse all pin widgets and store the useful pin nids to cfg - * - * The number of line-outs or any primary output is stored in line_outs, - * and the corresponding output pins are assigned to line_out_pins[], - * in the order of front, rear, CLFE, side, ... - * - * If more extra outputs (speaker and headphone) are found, the pins are - * assisnged to hp_pins[] and speaker_pins[], respectively. If no line-out jack - * is detected, one of speaker of HP pins is assigned as the primary - * output, i.e. to line_out_pins[0]. So, line_outs is always positive - * if any analog output exists. - * - * The analog input pins are assigned to inputs array. - * The digital input/output pins are assigned to dig_in_pin and dig_out_pin, - * respectively. - */ -int snd_hda_parse_pin_defcfg(struct hda_codec *codec, - struct auto_pin_cfg *cfg, - const hda_nid_t *ignore_nids, - unsigned int cond_flags) -{ - hda_nid_t nid, end_nid; - short seq, assoc_line_out; - short sequences_line_out[ARRAY_SIZE(cfg->line_out_pins)]; - short sequences_speaker[ARRAY_SIZE(cfg->speaker_pins)]; - short sequences_hp[ARRAY_SIZE(cfg->hp_pins)]; - int i; - - memset(cfg, 0, sizeof(*cfg)); - - memset(sequences_line_out, 0, sizeof(sequences_line_out)); - memset(sequences_speaker, 0, sizeof(sequences_speaker)); - memset(sequences_hp, 0, sizeof(sequences_hp)); - assoc_line_out = 0; - - codec->ignore_misc_bit = true; - end_nid = codec->start_nid + codec->num_nodes; - for (nid = codec->start_nid; nid < end_nid; nid++) { - unsigned int wid_caps = get_wcaps(codec, nid); - unsigned int wid_type = get_wcaps_type(wid_caps); - unsigned int def_conf; - short assoc, loc, conn, dev; - - /* read all default configuration for pin complex */ - if (wid_type != AC_WID_PIN) - continue; - /* ignore the given nids (e.g. pc-beep returns error) */ - if (ignore_nids && is_in_nid_list(nid, ignore_nids)) - continue; - - def_conf = snd_hda_codec_get_pincfg(codec, nid); - if (!(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & - AC_DEFCFG_MISC_NO_PRESENCE)) - codec->ignore_misc_bit = false; - conn = get_defcfg_connect(def_conf); - if (conn == AC_JACK_PORT_NONE) - continue; - loc = get_defcfg_location(def_conf); - dev = get_defcfg_device(def_conf); - - /* workaround for buggy BIOS setups */ - if (dev == AC_JACK_LINE_OUT) { - if (conn == AC_JACK_PORT_FIXED) - dev = AC_JACK_SPEAKER; - } - - switch (dev) { - case AC_JACK_LINE_OUT: - seq = get_defcfg_sequence(def_conf); - assoc = get_defcfg_association(def_conf); - - if (!(wid_caps & AC_WCAP_STEREO)) - if (!cfg->mono_out_pin) - cfg->mono_out_pin = nid; - if (!assoc) - continue; - if (!assoc_line_out) - assoc_line_out = assoc; - else if (assoc_line_out != assoc) - continue; - if (cfg->line_outs >= ARRAY_SIZE(cfg->line_out_pins)) - continue; - cfg->line_out_pins[cfg->line_outs] = nid; - sequences_line_out[cfg->line_outs] = seq; - cfg->line_outs++; - break; - case AC_JACK_SPEAKER: - seq = get_defcfg_sequence(def_conf); - assoc = get_defcfg_association(def_conf); - if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins)) - continue; - cfg->speaker_pins[cfg->speaker_outs] = nid; - sequences_speaker[cfg->speaker_outs] = (assoc << 4) | seq; - cfg->speaker_outs++; - break; - case AC_JACK_HP_OUT: - seq = get_defcfg_sequence(def_conf); - assoc = get_defcfg_association(def_conf); - if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins)) - continue; - cfg->hp_pins[cfg->hp_outs] = nid; - sequences_hp[cfg->hp_outs] = (assoc << 4) | seq; - cfg->hp_outs++; - break; - case AC_JACK_MIC_IN: - add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_MIC); - break; - case AC_JACK_LINE_IN: - add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_LINE_IN); - break; - case AC_JACK_CD: - add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_CD); - break; - case AC_JACK_AUX: - add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_AUX); - break; - case AC_JACK_SPDIF_OUT: - case AC_JACK_DIG_OTHER_OUT: - if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins)) - continue; - cfg->dig_out_pins[cfg->dig_outs] = nid; - cfg->dig_out_type[cfg->dig_outs] = - (loc == AC_JACK_LOC_HDMI) ? - HDA_PCM_TYPE_HDMI : HDA_PCM_TYPE_SPDIF; - cfg->dig_outs++; - break; - case AC_JACK_SPDIF_IN: - case AC_JACK_DIG_OTHER_IN: - cfg->dig_in_pin = nid; - if (loc == AC_JACK_LOC_HDMI) - cfg->dig_in_type = HDA_PCM_TYPE_HDMI; - else - cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; - break; - } - } - - /* FIX-UP: - * If no line-out is defined but multiple HPs are found, - * some of them might be the real line-outs. - */ - if (!cfg->line_outs && cfg->hp_outs > 1 && - !(cond_flags & HDA_PINCFG_NO_HP_FIXUP)) { - int i = 0; - while (i < cfg->hp_outs) { - /* The real HPs should have the sequence 0x0f */ - if ((sequences_hp[i] & 0x0f) == 0x0f) { - i++; - continue; - } - /* Move it to the line-out table */ - cfg->line_out_pins[cfg->line_outs] = cfg->hp_pins[i]; - sequences_line_out[cfg->line_outs] = sequences_hp[i]; - cfg->line_outs++; - cfg->hp_outs--; - memmove(cfg->hp_pins + i, cfg->hp_pins + i + 1, - sizeof(cfg->hp_pins[0]) * (cfg->hp_outs - i)); - memmove(sequences_hp + i, sequences_hp + i + 1, - sizeof(sequences_hp[0]) * (cfg->hp_outs - i)); - } - memset(cfg->hp_pins + cfg->hp_outs, 0, - sizeof(hda_nid_t) * (AUTO_CFG_MAX_OUTS - cfg->hp_outs)); - if (!cfg->hp_outs) - cfg->line_out_type = AUTO_PIN_HP_OUT; - - } - - /* sort by sequence */ - sort_pins_by_sequence(cfg->line_out_pins, sequences_line_out, - cfg->line_outs); - sort_pins_by_sequence(cfg->speaker_pins, sequences_speaker, - cfg->speaker_outs); - sort_pins_by_sequence(cfg->hp_pins, sequences_hp, - cfg->hp_outs); - - /* - * FIX-UP: if no line-outs are detected, try to use speaker or HP pin - * as a primary output - */ - if (!cfg->line_outs && - !(cond_flags & HDA_PINCFG_NO_LO_FIXUP)) { - if (cfg->speaker_outs) { - cfg->line_outs = cfg->speaker_outs; - memcpy(cfg->line_out_pins, cfg->speaker_pins, - sizeof(cfg->speaker_pins)); - cfg->speaker_outs = 0; - memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins)); - cfg->line_out_type = AUTO_PIN_SPEAKER_OUT; - } else if (cfg->hp_outs) { - cfg->line_outs = cfg->hp_outs; - memcpy(cfg->line_out_pins, cfg->hp_pins, - sizeof(cfg->hp_pins)); - cfg->hp_outs = 0; - memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins)); - cfg->line_out_type = AUTO_PIN_HP_OUT; - } - } - - reorder_outputs(cfg->line_outs, cfg->line_out_pins); - reorder_outputs(cfg->hp_outs, cfg->hp_pins); - reorder_outputs(cfg->speaker_outs, cfg->speaker_pins); - - sort_autocfg_input_pins(cfg); - - /* - * debug prints of the parsed results - */ - snd_printd("autoconfig: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x) type:%s\n", - cfg->line_outs, cfg->line_out_pins[0], cfg->line_out_pins[1], - cfg->line_out_pins[2], cfg->line_out_pins[3], - cfg->line_out_pins[4], - cfg->line_out_type == AUTO_PIN_HP_OUT ? "hp" : - (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT ? - "speaker" : "line")); - snd_printd(" speaker_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", - cfg->speaker_outs, cfg->speaker_pins[0], - cfg->speaker_pins[1], cfg->speaker_pins[2], - cfg->speaker_pins[3], cfg->speaker_pins[4]); - snd_printd(" hp_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", - cfg->hp_outs, cfg->hp_pins[0], - cfg->hp_pins[1], cfg->hp_pins[2], - cfg->hp_pins[3], cfg->hp_pins[4]); - snd_printd(" mono: mono_out=0x%x\n", cfg->mono_out_pin); - if (cfg->dig_outs) - snd_printd(" dig-out=0x%x/0x%x\n", - cfg->dig_out_pins[0], cfg->dig_out_pins[1]); - snd_printd(" inputs:"); - for (i = 0; i < cfg->num_inputs; i++) { - snd_printd(" %s=0x%x", - hda_get_autocfg_input_label(codec, cfg, i), - cfg->inputs[i].pin); - } - snd_printd("\n"); - if (cfg->dig_in_pin) - snd_printd(" dig-in=0x%x\n", cfg->dig_in_pin); - - return 0; -} -EXPORT_SYMBOL_HDA(snd_hda_parse_pin_defcfg); - -int snd_hda_get_input_pin_attr(unsigned int def_conf) -{ - unsigned int loc = get_defcfg_location(def_conf); - unsigned int conn = get_defcfg_connect(def_conf); - if (conn == AC_JACK_PORT_NONE) - return INPUT_PIN_ATTR_UNUSED; - /* Windows may claim the internal mic to be BOTH, too */ - if (conn == AC_JACK_PORT_FIXED || conn == AC_JACK_PORT_BOTH) - return INPUT_PIN_ATTR_INT; - if ((loc & 0x30) == AC_JACK_LOC_INTERNAL) - return INPUT_PIN_ATTR_INT; - if ((loc & 0x30) == AC_JACK_LOC_SEPARATE) - return INPUT_PIN_ATTR_DOCK; - if (loc == AC_JACK_LOC_REAR) - return INPUT_PIN_ATTR_REAR; - if (loc == AC_JACK_LOC_FRONT) - return INPUT_PIN_ATTR_FRONT; - return INPUT_PIN_ATTR_NORMAL; -} -EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_attr); - -/** - * hda_get_input_pin_label - Give a label for the given input pin - * - * When check_location is true, the function checks the pin location - * for mic and line-in pins, and set an appropriate prefix like "Front", - * "Rear", "Internal". - */ - -static const char *hda_get_input_pin_label(struct hda_codec *codec, - hda_nid_t pin, bool check_location) -{ - unsigned int def_conf; - static const char * const mic_names[] = { - "Internal Mic", "Dock Mic", "Mic", "Front Mic", "Rear Mic", - }; - int attr; - - def_conf = snd_hda_codec_get_pincfg(codec, pin); - - switch (get_defcfg_device(def_conf)) { - case AC_JACK_MIC_IN: - if (!check_location) - return "Mic"; - attr = snd_hda_get_input_pin_attr(def_conf); - if (!attr) - return "None"; - return mic_names[attr - 1]; - case AC_JACK_LINE_IN: - if (!check_location) - return "Line"; - attr = snd_hda_get_input_pin_attr(def_conf); - if (!attr) - return "None"; - if (attr == INPUT_PIN_ATTR_DOCK) - return "Dock Line"; - return "Line"; - case AC_JACK_AUX: - return "Aux"; - case AC_JACK_CD: - return "CD"; - case AC_JACK_SPDIF_IN: - return "SPDIF In"; - case AC_JACK_DIG_OTHER_IN: - return "Digital In"; - default: - return "Misc"; - } -} - -/* Check whether the location prefix needs to be added to the label. - * If all mic-jacks are in the same location (e.g. rear panel), we don't - * have to put "Front" prefix to each label. In such a case, returns false. - */ -static int check_mic_location_need(struct hda_codec *codec, - const struct auto_pin_cfg *cfg, - int input) -{ - unsigned int defc; - int i, attr, attr2; - - defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[input].pin); - attr = snd_hda_get_input_pin_attr(defc); - /* for internal or docking mics, we need locations */ - if (attr <= INPUT_PIN_ATTR_NORMAL) - return 1; - - attr = 0; - for (i = 0; i < cfg->num_inputs; i++) { - defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[i].pin); - attr2 = snd_hda_get_input_pin_attr(defc); - if (attr2 >= INPUT_PIN_ATTR_NORMAL) { - if (attr && attr != attr2) - return 1; /* different locations found */ - attr = attr2; - } - } - return 0; -} - -/** - * hda_get_autocfg_input_label - Get a label for the given input - * - * Get a label for the given input pin defined by the autocfg item. - * Unlike hda_get_input_pin_label(), this function checks all inputs - * defined in autocfg and avoids the redundant mic/line prefix as much as - * possible. - */ -const char *hda_get_autocfg_input_label(struct hda_codec *codec, - const struct auto_pin_cfg *cfg, - int input) -{ - int type = cfg->inputs[input].type; - int has_multiple_pins = 0; - - if ((input > 0 && cfg->inputs[input - 1].type == type) || - (input < cfg->num_inputs - 1 && cfg->inputs[input + 1].type == type)) - has_multiple_pins = 1; - if (has_multiple_pins && type == AUTO_PIN_MIC) - has_multiple_pins &= check_mic_location_need(codec, cfg, input); - return hda_get_input_pin_label(codec, cfg->inputs[input].pin, - has_multiple_pins); -} -EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label); - -/* return the position of NID in the list, or -1 if not found */ -static int find_idx_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums) -{ - int i; - for (i = 0; i < nums; i++) - if (list[i] == nid) - return i; - return -1; -} - -/* get a unique suffix or an index number */ -static const char *check_output_sfx(hda_nid_t nid, const hda_nid_t *pins, - int num_pins, int *indexp) -{ - static const char * const channel_sfx[] = { - " Front", " Surround", " CLFE", " Side" - }; - int i; - - i = find_idx_in_nid_list(nid, pins, num_pins); - if (i < 0) - return NULL; - if (num_pins == 1) - return ""; - if (num_pins > ARRAY_SIZE(channel_sfx)) { - if (indexp) - *indexp = i; - return ""; - } - return channel_sfx[i]; -} - -static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, - const struct auto_pin_cfg *cfg, - const char *name, char *label, int maxlen, - int *indexp) -{ - unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); - int attr = snd_hda_get_input_pin_attr(def_conf); - const char *pfx = "", *sfx = ""; - - /* handle as a speaker if it's a fixed line-out */ - if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT) - name = "Speaker"; - /* check the location */ - switch (attr) { - case INPUT_PIN_ATTR_DOCK: - pfx = "Dock "; - break; - case INPUT_PIN_ATTR_FRONT: - pfx = "Front "; - break; - } - if (cfg) { - /* try to give a unique suffix if needed */ - sfx = check_output_sfx(nid, cfg->line_out_pins, cfg->line_outs, - indexp); - if (!sfx) - sfx = check_output_sfx(nid, cfg->speaker_pins, cfg->speaker_outs, - indexp); - if (!sfx) { - /* don't add channel suffix for Headphone controls */ - int idx = find_idx_in_nid_list(nid, cfg->hp_pins, - cfg->hp_outs); - if (idx >= 0) - *indexp = idx; - sfx = ""; - } - } - snprintf(label, maxlen, "%s%s%s", pfx, name, sfx); - return 1; -} - /** - * snd_hda_get_pin_label - Get a label for the given I/O pin - * - * Get a label for the given pin. This function works for both input and - * output pins. When @cfg is given as non-NULL, the function tries to get - * an optimized label using hda_get_autocfg_input_label(). + * snd_hda_get_default_vref - Get the default (mic) VREF pin bits * - * This function tries to give a unique label string for the pin as much as - * possible. For example, when the multiple line-outs are present, it adds - * the channel suffix like "Front", "Surround", etc (only when @cfg is given). - * If no unique name with a suffix is available and @indexp is non-NULL, the - * index number is stored in the pointer. - */ -int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, - const struct auto_pin_cfg *cfg, - char *label, int maxlen, int *indexp) -{ - unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); - const char *name = NULL; - int i; - - if (indexp) - *indexp = 0; - if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) - return 0; - - switch (get_defcfg_device(def_conf)) { - case AC_JACK_LINE_OUT: - return fill_audio_out_name(codec, nid, cfg, "Line Out", - label, maxlen, indexp); - case AC_JACK_SPEAKER: - return fill_audio_out_name(codec, nid, cfg, "Speaker", - label, maxlen, indexp); - case AC_JACK_HP_OUT: - return fill_audio_out_name(codec, nid, cfg, "Headphone", - label, maxlen, indexp); - case AC_JACK_SPDIF_OUT: - case AC_JACK_DIG_OTHER_OUT: - if (get_defcfg_location(def_conf) == AC_JACK_LOC_HDMI) - name = "HDMI"; - else - name = "SPDIF"; - if (cfg && indexp) { - i = find_idx_in_nid_list(nid, cfg->dig_out_pins, - cfg->dig_outs); - if (i >= 0) - *indexp = i; + * Guess the suitable VREF pin bits to be set as the pin-control value. + * Note: the function doesn't set the AC_PINCTL_IN_EN bit. + */ +unsigned int snd_hda_get_default_vref(struct hda_codec *codec, hda_nid_t pin) +{ + unsigned int pincap; + unsigned int oldval; + oldval = snd_hda_codec_read(codec, pin, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + pincap = snd_hda_query_pin_caps(codec, pin); + pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; + /* Exception: if the default pin setup is vref50, we give it priority */ + if ((pincap & AC_PINCAP_VREF_80) && oldval != PIN_VREF50) + return AC_PINCTL_VREF_80; + else if (pincap & AC_PINCAP_VREF_50) + return AC_PINCTL_VREF_50; + else if (pincap & AC_PINCAP_VREF_100) + return AC_PINCTL_VREF_100; + else if (pincap & AC_PINCAP_VREF_GRD) + return AC_PINCTL_VREF_GRD; + return AC_PINCTL_VREF_HIZ; +} +EXPORT_SYMBOL_HDA(snd_hda_get_default_vref); + +int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin, + unsigned int val, bool cached) +{ + if (val) { + unsigned int cap = snd_hda_query_pin_caps(codec, pin); + if (cap && (val & AC_PINCTL_OUT_EN)) { + if (!(cap & AC_PINCAP_OUT)) + val &= ~(AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); + else if ((val & AC_PINCTL_HP_EN) && + !(cap & AC_PINCAP_HP_DRV)) + val &= ~AC_PINCTL_HP_EN; } - break; - default: - if (cfg) { - for (i = 0; i < cfg->num_inputs; i++) { - if (cfg->inputs[i].pin != nid) - continue; - name = hda_get_autocfg_input_label(codec, cfg, i); - if (name) - break; - } + if (cap && (val & AC_PINCTL_IN_EN)) { + if (!(cap & AC_PINCAP_IN)) + val &= ~(AC_PINCTL_IN_EN | AC_PINCTL_VREFEN); } - if (!name) - name = hda_get_input_pin_label(codec, nid, true); - break; } - if (!name) - return 0; - strlcpy(label, name, maxlen); - return 1; + if (cached) + return snd_hda_codec_update_cache(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, val); + else + return snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, val); } -EXPORT_SYMBOL_HDA(snd_hda_get_pin_label); +EXPORT_SYMBOL_HDA(_snd_hda_set_pin_ctl); /** * snd_hda_add_imux_item - Add an item to input_mux @@ -5444,8 +5048,6 @@ int snd_hda_suspend(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { if (hda_codec_is_power_on(codec)) hda_call_codec_suspend(codec); - if (codec->patch_ops.post_suspend) - codec->patch_ops.post_suspend(codec); } return 0; } @@ -5465,10 +5067,7 @@ int snd_hda_resume(struct hda_bus *bus) struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { - if (codec->patch_ops.pre_resume) - codec->patch_ops.pre_resume(codec); - if (snd_hda_codec_needs_resume(codec)) - hda_call_codec_resume(codec); + hda_call_codec_resume(codec); } return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 56b4f74c0b13..c422d330ca54 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -323,6 +323,9 @@ enum { #define AC_PWRST_D1 0x01 #define AC_PWRST_D2 0x02 #define AC_PWRST_D3 0x03 +#define AC_PWRST_ERROR (1<<8) +#define AC_PWRST_CLK_STOP_OK (1<<9) +#define AC_PWRST_SETTING_RESET (1<<10) /* Processing capabilies */ #define AC_PCAP_BENIGN (1<<0) @@ -703,9 +706,7 @@ struct hda_codec_ops { void (*set_power_state)(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); #ifdef CONFIG_PM - int (*suspend)(struct hda_codec *codec, pm_message_t state); - int (*post_suspend)(struct hda_codec *codec); - int (*pre_resume)(struct hda_codec *codec); + int (*suspend)(struct hda_codec *codec); int (*resume)(struct hda_codec *codec); #endif #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -829,6 +830,7 @@ struct hda_codec { struct mutex spdif_mutex; struct mutex control_mutex; + struct mutex hash_mutex; struct snd_array spdif_out; unsigned int spdif_in_enable; /* SPDIF input enable? */ const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ @@ -861,12 +863,13 @@ struct hda_codec { unsigned int no_jack_detect:1; /* Machine has no jack-detection */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ - unsigned int power_transition :1; /* power-state in transition */ + int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ unsigned long power_on_acct; unsigned long power_off_acct; unsigned long power_jiffies; + spinlock_t power_lock; #endif /* codec-specific additional proc output */ @@ -911,10 +914,13 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *start_id); int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *conn_list, int max_conns); +static inline int +snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid) +{ + return snd_hda_get_connections(codec, nid, NULL, 0); +} int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *conn_list, int max_conns); -int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, - const hda_nid_t **listp); int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums, const hda_nid_t *list); int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, @@ -1020,6 +1026,9 @@ void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state, bool eapd_workaround); +int snd_hda_lock_devices(struct hda_bus *bus); +void snd_hda_unlock_devices(struct hda_bus *bus); + /* * power management */ @@ -1050,13 +1059,13 @@ const char *snd_hda_get_jack_location(u32 cfg); */ #ifdef CONFIG_SND_HDA_POWER_SAVE void snd_hda_power_up(struct hda_codec *codec); +void snd_hda_power_up_d3wait(struct hda_codec *codec); void snd_hda_power_down(struct hda_codec *codec); -#define snd_hda_codec_needs_resume(codec) codec->power_count void snd_hda_update_power_acct(struct hda_codec *codec); #else static inline void snd_hda_power_up(struct hda_codec *codec) {} +static inline void snd_hda_power_up_d3wait(struct hda_codec *codec) {} static inline void snd_hda_power_down(struct hda_codec *codec) {} -#define snd_hda_codec_needs_resume(codec) 1 #endif #ifdef CONFIG_SND_HDA_PATCH_LOADER diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1f350522bed4..c8aced182fd1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -53,6 +53,8 @@ #endif #include <sound/core.h> #include <sound/initval.h> +#include <linux/vgaarb.h> +#include <linux/vga_switcheroo.h> #include "hda_codec.h" @@ -70,7 +72,7 @@ static int enable_msi = -1; static char *patch[SNDRV_CARDS]; #endif #ifdef CONFIG_SND_HDA_INPUT_BEEP -static int beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = +static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif @@ -101,9 +103,9 @@ module_param_array(patch, charp, NULL, 0444); MODULE_PARM_DESC(patch, "Patch file for Intel HD audio interface."); #endif #ifdef CONFIG_SND_HDA_INPUT_BEEP -module_param_array(beep_mode, int, NULL, 0444); +module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " - "(0=off, 1=on, 2=mute switch on/off) (default=1)."); + "(0=off, 1=on) (default=1)."); #endif #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -149,6 +151,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, CPT}," "{Intel, PPT}," "{Intel, LPT}," + "{Intel, HPT}," "{Intel, PBG}," "{Intel, SCH}," "{ATI, SB450}," @@ -175,6 +178,13 @@ MODULE_DESCRIPTION("Intel HDA driver"); #define SFX "hda-intel: " #endif +#if defined(CONFIG_PM) && defined(CONFIG_VGA_SWITCHEROO) +#ifdef CONFIG_SND_HDA_CODEC_HDMI +#define SUPPORT_VGA_SWITCHEROO +#endif +#endif + + /* * registers */ @@ -472,6 +482,12 @@ struct azx { unsigned int probing :1; /* codec probing phase */ unsigned int snoop:1; unsigned int align_buffer_size:1; + unsigned int region_requested:1; + + /* VGA-switcheroo setup */ + unsigned int use_vga_switcheroo:1; + unsigned int init_failed:1; /* delayed init failed */ + unsigned int disabled:1; /* disabled by VGA-switcher */ /* for debugging */ unsigned int last_cmd[AZX_MAX_CODECS]; @@ -497,6 +513,7 @@ enum { AZX_DRIVER_NVIDIA, AZX_DRIVER_TERA, AZX_DRIVER_CTX, + AZX_DRIVER_CTHDA, AZX_DRIVER_GENERIC, AZX_NUM_DRIVERS, /* keep this as last entry */ }; @@ -518,6 +535,8 @@ enum { #define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ #define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */ #define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */ +#define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */ +#define AZX_DCAPS_POSFIX_COMBO (1 << 24) /* Use COMBO as default */ /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ @@ -533,7 +552,23 @@ enum { (AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\ AZX_DCAPS_ALIGN_BUFSIZE) -static char *driver_short_names[] __devinitdata = { +#define AZX_DCAPS_PRESET_CTHDA \ + (AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_4K_BDLE_BOUNDARY) + +/* + * VGA-switcher support + */ +#ifdef SUPPORT_VGA_SWITCHEROO +#define DELAYED_INIT_MARK +#define DELAYED_INITDATA_MARK +#define use_vga_switcheroo(chip) ((chip)->use_vga_switcheroo) +#else +#define DELAYED_INIT_MARK __devinit +#define DELAYED_INITDATA_MARK __devinitdata +#define use_vga_switcheroo(chip) 0 +#endif + +static char *driver_short_names[] DELAYED_INITDATA_MARK = { [AZX_DRIVER_ICH] = "HDA Intel", [AZX_DRIVER_PCH] = "HDA Intel PCH", [AZX_DRIVER_SCH] = "HDA Intel MID", @@ -546,6 +581,7 @@ static char *driver_short_names[] __devinitdata = { [AZX_DRIVER_NVIDIA] = "HDA NVidia", [AZX_DRIVER_TERA] = "HDA Teradici", [AZX_DRIVER_CTX] = "HDA Creative", + [AZX_DRIVER_CTHDA] = "HDA Creative", [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; @@ -953,6 +989,8 @@ static int azx_send_cmd(struct hda_bus *bus, unsigned int val) { struct azx *chip = bus->private_data; + if (chip->disabled) + return 0; chip->last_cmd[azx_command_addr(val)] = val; if (chip->single_cmd) return azx_single_send_cmd(bus, val); @@ -965,6 +1003,8 @@ static unsigned int azx_get_response(struct hda_bus *bus, unsigned int addr) { struct azx *chip = bus->private_data; + if (chip->disabled) + return 0; if (chip->single_cmd) return azx_single_get_response(bus, addr); else @@ -1230,6 +1270,11 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) spin_lock(&chip->reg_lock); + if (chip->disabled) { + spin_unlock(&chip->reg_lock); + return IRQ_NONE; + } + status = azx_readl(chip, INTSTS); if (status == 0) { spin_unlock(&chip->reg_lock); @@ -1285,7 +1330,8 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) /* * set up a BDL entry */ -static int setup_bdle(struct snd_pcm_substream *substream, +static int setup_bdle(struct azx *chip, + struct snd_pcm_substream *substream, struct azx_dev *azx_dev, u32 **bdlp, int ofs, int size, int with_ioc) { @@ -1304,6 +1350,12 @@ static int setup_bdle(struct snd_pcm_substream *substream, bdl[1] = cpu_to_le32(upper_32_bits(addr)); /* program the size field of the BDL entry */ chunk = snd_pcm_sgbuf_get_chunk_size(substream, ofs, size); + /* one BDLE cannot cross 4K boundary on CTHDA chips */ + if (chip->driver_caps & AZX_DCAPS_4K_BDLE_BOUNDARY) { + u32 remain = 0x1000 - (ofs & 0xfff); + if (chunk > remain) + chunk = remain; + } bdl[2] = cpu_to_le32(chunk); /* program the IOC to enable interrupt * only when the whole fragment is processed @@ -1356,7 +1408,7 @@ static int azx_setup_periods(struct azx *chip, bdl_pos_adj[chip->dev_index]); pos_adj = 0; } else { - ofs = setup_bdle(substream, azx_dev, + ofs = setup_bdle(chip, substream, azx_dev, &bdl, ofs, pos_adj, !substream->runtime->no_period_wakeup); if (ofs < 0) @@ -1366,10 +1418,10 @@ static int azx_setup_periods(struct azx *chip, pos_adj = 0; for (i = 0; i < periods; i++) { if (i == periods - 1 && pos_adj) - ofs = setup_bdle(substream, azx_dev, &bdl, ofs, + ofs = setup_bdle(chip, substream, azx_dev, &bdl, ofs, period_bytes - pos_adj, 0); else - ofs = setup_bdle(substream, azx_dev, &bdl, ofs, + ofs = setup_bdle(chip, substream, azx_dev, &bdl, ofs, period_bytes, !substream->runtime->no_period_wakeup); if (ofs < 0) @@ -1508,12 +1560,12 @@ static void azx_bus_reset(struct hda_bus *bus) */ /* number of codec slots for each chipset: 0 = default slots (i.e. 4) */ -static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] __devinitdata = { +static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] DELAYED_INITDATA_MARK = { [AZX_DRIVER_NVIDIA] = 8, [AZX_DRIVER_TERA] = 1, }; -static int __devinit azx_codec_create(struct azx *chip, const char *model) +static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *model) { struct hda_bus_template bus_temp; int c, codecs, err; @@ -1716,7 +1768,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) buff_step); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, buff_step); - snd_hda_power_up(apcm->codec); + snd_hda_power_up_d3wait(apcm->codec); err = hinfo->ops.open(hinfo, apcm->codec, substream); if (err < 0) { azx_release_device(azx_dev); @@ -2353,20 +2405,10 @@ static void azx_power_notify(struct hda_bus *bus) * power management */ -static int snd_hda_codecs_inuse(struct hda_bus *bus) -{ - struct hda_codec *codec; - - list_for_each_entry(codec, &bus->codec_list, list) { - if (snd_hda_codec_needs_resume(codec)) - return 1; - } - return 0; -} - -static int azx_suspend(struct pci_dev *pci, pm_message_t state) +static int azx_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; struct azx_pcm *p; @@ -2385,13 +2427,14 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_msi(chip->pci); pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int azx_resume(struct pci_dev *pci) +static int azx_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; pci_set_power_state(pci, PCI_D0); @@ -2410,13 +2453,18 @@ static int azx_resume(struct pci_dev *pci) return -EIO; azx_init_pci(chip); - if (snd_hda_codecs_inuse(chip->bus)) - azx_init_chip(chip, 1); + azx_init_chip(chip, 1); snd_hda_resume(chip->bus); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } +static SIMPLE_DEV_PM_OPS(azx_pm, azx_suspend, azx_resume); +#define AZX_PM_OPS &azx_pm +#else +#define azx_suspend(dev) +#define azx_resume(dev) +#define AZX_PM_OPS NULL #endif /* CONFIG_PM */ @@ -2443,6 +2491,106 @@ static void azx_notifier_unregister(struct azx *chip) unregister_reboot_notifier(&chip->reboot_notifier); } +static int DELAYED_INIT_MARK azx_first_init(struct azx *chip); +static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip); + +#ifdef SUPPORT_VGA_SWITCHEROO +static struct pci_dev __devinit *get_bound_vga(struct pci_dev *pci); + +static void azx_vs_set_state(struct pci_dev *pci, + enum vga_switcheroo_state state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct azx *chip = card->private_data; + bool disabled; + + if (chip->init_failed) + return; + + disabled = (state == VGA_SWITCHEROO_OFF); + if (chip->disabled == disabled) + return; + + if (!chip->bus) { + chip->disabled = disabled; + if (!disabled) { + snd_printk(KERN_INFO SFX + "%s: Start delayed initialization\n", + pci_name(chip->pci)); + if (azx_first_init(chip) < 0 || + azx_probe_continue(chip) < 0) { + snd_printk(KERN_ERR SFX + "%s: initialization error\n", + pci_name(chip->pci)); + chip->init_failed = true; + } + } + } else { + snd_printk(KERN_INFO SFX + "%s %s via VGA-switcheroo\n", + disabled ? "Disabling" : "Enabling", + pci_name(chip->pci)); + if (disabled) { + azx_suspend(&pci->dev); + chip->disabled = true; + snd_hda_lock_devices(chip->bus); + } else { + snd_hda_unlock_devices(chip->bus); + chip->disabled = false; + azx_resume(&pci->dev); + } + } +} + +static bool azx_vs_can_switch(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct azx *chip = card->private_data; + + if (chip->init_failed) + return false; + if (chip->disabled || !chip->bus) + return true; + if (snd_hda_lock_devices(chip->bus)) + return false; + snd_hda_unlock_devices(chip->bus); + return true; +} + +static void __devinit init_vga_switcheroo(struct azx *chip) +{ + struct pci_dev *p = get_bound_vga(chip->pci); + if (p) { + snd_printk(KERN_INFO SFX + "%s: Handle VGA-switcheroo audio client\n", + pci_name(chip->pci)); + chip->use_vga_switcheroo = 1; + pci_dev_put(p); + } +} + +static const struct vga_switcheroo_client_ops azx_vs_ops = { + .set_gpu_state = azx_vs_set_state, + .can_switch = azx_vs_can_switch, +}; + +static int __devinit register_vga_switcheroo(struct azx *chip) +{ + if (!chip->use_vga_switcheroo) + return 0; + /* FIXME: currently only handling DIS controller + * is there any machine with two switchable HDMI audio controllers? + */ + return vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, + VGA_SWITCHEROO_DIS, + chip->bus != NULL); +} +#else +#define init_vga_switcheroo(chip) /* NOP */ +#define register_vga_switcheroo(chip) 0 +#define check_hdmi_disabled(pci) false +#endif /* SUPPORT_VGA_SWITCHER */ + /* * destructor */ @@ -2452,6 +2600,12 @@ static int azx_free(struct azx *chip) azx_notifier_unregister(chip); + if (use_vga_switcheroo(chip)) { + if (chip->disabled && chip->bus) + snd_hda_unlock_devices(chip->bus); + vga_switcheroo_unregister_client(chip->pci); + } + if (chip->initialized) { azx_clear_irq_pending(chip); for (i = 0; i < chip->num_streams; i++) @@ -2481,7 +2635,8 @@ static int azx_free(struct azx *chip) mark_pages_wc(chip, &chip->posbuf, false); snd_dma_free_pages(&chip->posbuf); } - pci_release_regions(chip->pci); + if (chip->region_requested) + pci_release_regions(chip->pci); pci_disable_device(chip->pci); kfree(chip->azx_dev); kfree(chip); @@ -2494,6 +2649,47 @@ static int azx_dev_free(struct snd_device *device) return azx_free(device->device_data); } +#ifdef SUPPORT_VGA_SWITCHEROO +/* + * Check of disabled HDMI controller by vga-switcheroo + */ +static struct pci_dev __devinit *get_bound_vga(struct pci_dev *pci) +{ + struct pci_dev *p; + + /* check only discrete GPU */ + switch (pci->vendor) { + case PCI_VENDOR_ID_ATI: + case PCI_VENDOR_ID_AMD: + case PCI_VENDOR_ID_NVIDIA: + if (pci->devfn == 1) { + p = pci_get_domain_bus_and_slot(pci_domain_nr(pci->bus), + pci->bus->number, 0); + if (p) { + if ((p->class >> 8) == PCI_CLASS_DISPLAY_VGA) + return p; + pci_dev_put(p); + } + } + break; + } + return NULL; +} + +static bool __devinit check_hdmi_disabled(struct pci_dev *pci) +{ + bool vga_inactive = false; + struct pci_dev *p = get_bound_vga(pci); + + if (p) { + if (vga_switcheroo_get_client_state(p) == VGA_SWITCHEROO_OFF) + vga_inactive = true; + pci_dev_put(p); + } + return vga_inactive; +} +#endif /* SUPPORT_VGA_SWITCHEROO */ + /* * white/black-listing for position_fix */ @@ -2545,6 +2741,10 @@ static int __devinit check_position_fix(struct azx *chip, int fix) snd_printd(SFX "Using LPIB position fix\n"); return POS_FIX_LPIB; } + if (chip->driver_caps & AZX_DCAPS_POSFIX_COMBO) { + snd_printd(SFX "Using COMBO position fix\n"); + return POS_FIX_COMBO; + } return POS_FIX_AUTO; } @@ -2565,6 +2765,8 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = { /* forced codec slots */ SND_PCI_QUIRK(0x1043, 0x1262, "ASUS W5Fm", 0x103), SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103), + /* WinFast VP200 H (Teradici) user reported broken communication */ + SND_PCI_QUIRK(0x3a21, 0x040d, "WinFast VP200 H", 0x101), {} }; @@ -2669,12 +2871,11 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, int dev, unsigned int driver_caps, struct azx **rchip) { - struct azx *chip; - int i, err; - unsigned short gcap; static struct snd_device_ops ops = { .dev_free = azx_dev_free, }; + struct azx *chip; + int err; *rchip = NULL; @@ -2700,6 +2901,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->dev_index = dev; INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work); INIT_LIST_HEAD(&chip->pcm_list); + init_vga_switcheroo(chip); chip->position_fix[0] = chip->position_fix[1] = check_position_fix(chip, position_fix[dev]); @@ -2727,6 +2929,53 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } } + if (check_hdmi_disabled(pci)) { + snd_printk(KERN_INFO SFX "VGA controller for %s is disabled\n", + pci_name(pci)); + if (use_vga_switcheroo(chip)) { + snd_printk(KERN_INFO SFX "Delaying initialization\n"); + chip->disabled = true; + goto ok; + } + kfree(chip); + pci_disable_device(pci); + return -ENXIO; + } + + err = azx_first_init(chip); + if (err < 0) { + azx_free(chip); + return err; + } + + ok: + err = register_vga_switcheroo(chip); + if (err < 0) { + snd_printk(KERN_ERR SFX + "Error registering VGA-switcheroo client\n"); + azx_free(chip); + return err; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_printk(KERN_ERR SFX "Error creating device [card]!\n"); + azx_free(chip); + return err; + } + + *rchip = chip; + return 0; +} + +static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) +{ + int dev = chip->dev_index; + struct pci_dev *pci = chip->pci; + struct snd_card *card = chip->card; + int i, err; + unsigned short gcap; + #if BITS_PER_LONG != 64 /* Fix up base address on ULI M5461 */ if (chip->driver_type == AZX_DRIVER_ULI) { @@ -2738,28 +2987,23 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, #endif err = pci_request_regions(pci, "ICH HD audio"); - if (err < 0) { - kfree(chip); - pci_disable_device(pci); + if (err < 0) return err; - } + chip->region_requested = 1; chip->addr = pci_resource_start(pci, 0); chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR SFX "ioremap error\n"); - err = -ENXIO; - goto errout; + return -ENXIO; } if (chip->msi) if (pci_enable_msi(pci) < 0) chip->msi = 0; - if (azx_acquire_irq(chip, 0) < 0) { - err = -EBUSY; - goto errout; - } + if (azx_acquire_irq(chip, 0) < 0) + return -EBUSY; pci_set_master(pci); synchronize_irq(chip->irq); @@ -2838,7 +3082,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, GFP_KERNEL); if (!chip->azx_dev) { snd_printk(KERN_ERR SFX "cannot malloc azx_dev\n"); - goto errout; + return -ENOMEM; } for (i = 0; i < chip->num_streams; i++) { @@ -2848,7 +3092,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, BDL_SIZE, &chip->azx_dev[i].bdl); if (err < 0) { snd_printk(KERN_ERR SFX "cannot allocate BDL\n"); - goto errout; + return -ENOMEM; } mark_pages_wc(chip, &chip->azx_dev[i].bdl, true); } @@ -2858,13 +3102,13 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->num_streams * 8, &chip->posbuf); if (err < 0) { snd_printk(KERN_ERR SFX "cannot allocate posbuf\n"); - goto errout; + return -ENOMEM; } mark_pages_wc(chip, &chip->posbuf, true); /* allocate CORB/RIRB */ err = azx_alloc_cmd_io(chip); if (err < 0) - goto errout; + return err; /* initialize streams */ azx_init_stream(chip); @@ -2876,14 +3120,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, /* codec detection */ if (!chip->codec_mask) { snd_printk(KERN_ERR SFX "no codecs found!\n"); - err = -ENODEV; - goto errout; - } - - err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); - if (err <0) { - snd_printk(KERN_ERR SFX "Error creating device [card]!\n"); - goto errout; + return -ENODEV; } strcpy(card->driver, "HDA-Intel"); @@ -2893,12 +3130,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, "%s at 0x%lx irq %i", card->shortname, chip->addr, chip->irq); - *rchip = chip; return 0; - - errout: - azx_free(chip); - return err; } static void power_down_all_codecs(struct azx *chip) @@ -2943,6 +3175,27 @@ static int __devinit azx_probe(struct pci_dev *pci, goto out_free; card->private_data = chip; + if (!chip->disabled) { + err = azx_probe_continue(chip); + if (err < 0) + goto out_free; + } + + pci_set_drvdata(pci, card); + + dev++; + return 0; + +out_free: + snd_card_free(card); + return err; +} + +static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) +{ + int dev = chip->dev_index; + int err; + #ifdef CONFIG_SND_HDA_INPUT_BEEP chip->beep_mode = beep_mode[dev]; #endif @@ -2976,25 +3229,26 @@ static int __devinit azx_probe(struct pci_dev *pci, if (err < 0) goto out_free; - err = snd_card_register(card); + err = snd_card_register(chip->card); if (err < 0) goto out_free; - pci_set_drvdata(pci, card); chip->running = 1; power_down_all_codecs(chip); azx_notifier_register(chip); - dev++; - return err; + return 0; + out_free: - snd_card_free(card); + chip->init_failed = 1; return err; } static void __devexit azx_remove(struct pci_dev *pci) { - snd_card_free(pci_get_drvdata(pci)); + struct snd_card *card = pci_get_drvdata(pci); + if (card) + snd_card_free(card); pci_set_drvdata(pci, NULL); } @@ -3003,7 +3257,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* CPT */ { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE }, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, /* PBG */ { PCI_DEVICE(0x8086, 0x1d20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | @@ -3011,11 +3265,15 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* Panther Point */ { PCI_DEVICE(0x8086, 0x1e20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE}, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, /* Lynx Point */ { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE}, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + /* Haswell */ + { PCI_DEVICE(0x8086, 0x0c0c), + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | @@ -3101,6 +3359,10 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA | AZX_DCAPS_POSFIX_VIA }, + /* VIA GFX VT7122/VX900 */ + { PCI_DEVICE(0x1106, 0x9170), .driver_data = AZX_DRIVER_GENERIC }, + /* VIA GFX VT6122/VX11 */ + { PCI_DEVICE(0x1106, 0x9140), .driver_data = AZX_DRIVER_GENERIC }, /* SIS966 */ { PCI_DEVICE(0x1039, 0x7502), .driver_data = AZX_DRIVER_SIS }, /* ULI M5461 */ @@ -3114,6 +3376,11 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA | AZX_DCAPS_NO_64BIT }, /* Creative X-Fi (CA0110-IBG) */ + /* CTHDA chips */ + { PCI_DEVICE(0x1102, 0x0010), + .driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA }, + { PCI_DEVICE(0x1102, 0x0012), + .driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA }, #if !defined(CONFIG_SND_CTXFI) && !defined(CONFIG_SND_CTXFI_MODULE) /* the following entry conflicts with snd-ctxfi driver, * as ctxfi driver mutates from HD-audio to native mode with @@ -3148,26 +3415,14 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { MODULE_DEVICE_TABLE(pci, azx_ids); /* pci_driver definition */ -static struct pci_driver driver = { +static struct pci_driver azx_driver = { .name = KBUILD_MODNAME, .id_table = azx_ids, .probe = azx_probe, .remove = __devexit_p(azx_remove), -#ifdef CONFIG_PM - .suspend = azx_suspend, - .resume = azx_resume, -#endif + .driver = { + .pm = AZX_PM_OPS, + }, }; -static int __init alsa_card_azx_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_azx_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_azx_init) -module_exit(alsa_card_azx_exit) +module_pci_driver(azx_driver); diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index d68948499fbc..aaccc0236bda 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -17,6 +17,7 @@ #include <sound/jack.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_jack.h" bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) @@ -126,10 +127,15 @@ void snd_hda_jack_tbl_clear(struct hda_codec *codec) static void jack_detect_update(struct hda_codec *codec, struct hda_jack_tbl *jack) { - if (jack->jack_dirty || !jack->jack_detect) { + if (!jack->jack_dirty) + return; + + if (jack->phantom_jack) + jack->pin_sense = AC_PINSENSE_PRESENCE; + else jack->pin_sense = read_pin_sense(codec, jack->nid); - jack->jack_dirty = 0; - } + + jack->jack_dirty = 0; } /** @@ -263,8 +269,8 @@ static void hda_free_jack_priv(struct snd_jack *jack) * This assigns a jack-detection kctl to the given pin. The kcontrol * will have the given name and index. */ -int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, - const char *name, int idx) +static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, + const char *name, int idx, bool phantom_jack) { struct hda_jack_tbl *jack; struct snd_kcontrol *kctl; @@ -282,47 +288,81 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, if (err < 0) return err; jack->kctl = kctl; + jack->phantom_jack = !!phantom_jack; + state = snd_hda_jack_detect(codec, nid); snd_kctl_jack_report(codec->bus->card, kctl, state); #ifdef CONFIG_SND_HDA_INPUT_JACK - jack->type = get_input_jack_type(codec, nid); - err = snd_jack_new(codec->bus->card, name, jack->type, &jack->jack); - if (err < 0) - return err; - jack->jack->private_data = jack; - jack->jack->private_free = hda_free_jack_priv; - snd_jack_report(jack->jack, state ? jack->type : 0); + if (!phantom_jack) { + jack->type = get_input_jack_type(codec, nid); + err = snd_jack_new(codec->bus->card, name, jack->type, + &jack->jack); + if (err < 0) + return err; + jack->jack->private_data = jack; + jack->jack->private_free = hda_free_jack_priv; + snd_jack_report(jack->jack, state ? jack->type : 0); + } #endif return 0; } + +int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, + const char *name, int idx) +{ + return __snd_hda_jack_add_kctl(codec, nid, name, idx, false); +} EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl); +/* get the unique index number for the given kctl name */ +static int get_unique_index(struct hda_codec *codec, const char *name, int idx) +{ + struct hda_jack_tbl *jack; + int i, len = strlen(name); + again: + jack = codec->jacktbl.list; + for (i = 0; i < codec->jacktbl.used; i++, jack++) { + /* jack->kctl.id contains "XXX Jack" name string with index */ + if (jack->kctl && + !strncmp(name, jack->kctl->id.name, len) && + !strcmp(" Jack", jack->kctl->id.name + len) && + jack->kctl->id.index == idx) { + idx++; + goto again; + } + } + return idx; +} + static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, - const struct auto_pin_cfg *cfg, - char *lastname, int *lastidx) + const struct auto_pin_cfg *cfg) { unsigned int def_conf, conn; char name[44]; int idx, err; + bool phantom_jack; if (!nid) return 0; - if (!is_jack_detectable(codec, nid)) - return 0; def_conf = snd_hda_codec_get_pincfg(codec, nid); conn = get_defcfg_connect(def_conf); - if (conn != AC_JACK_PORT_COMPLEX) + if (conn == AC_JACK_PORT_NONE) return 0; + phantom_jack = (conn != AC_JACK_PORT_COMPLEX) || + !is_jack_detectable(codec, nid); snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx); - if (!strcmp(name, lastname) && idx == *lastidx) - idx++; - strncpy(lastname, name, 44); - *lastidx = idx; - err = snd_hda_jack_add_kctl(codec, nid, name, idx); + if (phantom_jack) + /* Example final name: "Internal Mic Phantom Jack" */ + strncat(name, " Phantom", sizeof(name) - strlen(name) - 1); + idx = get_unique_index(codec, name, idx); + err = __snd_hda_jack_add_kctl(codec, nid, name, idx, phantom_jack); if (err < 0) return err; - return snd_hda_jack_detect_enable(codec, nid, 0); + + if (!phantom_jack) + return snd_hda_jack_detect_enable(codec, nid, 0); + return 0; } /** @@ -332,42 +372,41 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { const hda_nid_t *p; - int i, err, lastidx = 0; - char lastname[44] = ""; + int i, err; for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) { - err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); + err = add_jack_kctl(codec, *p, cfg); if (err < 0) return err; } for (i = 0, p = cfg->hp_pins; i < cfg->hp_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ break; - err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); + err = add_jack_kctl(codec, *p, cfg); if (err < 0) return err; } for (i = 0, p = cfg->speaker_pins; i < cfg->speaker_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ break; - err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); + err = add_jack_kctl(codec, *p, cfg); if (err < 0) return err; } for (i = 0; i < cfg->num_inputs; i++) { - err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, lastname, &lastidx); + err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg); if (err < 0) return err; } for (i = 0, p = cfg->dig_out_pins; i < cfg->dig_outs; i++, p++) { - err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); + err = add_jack_kctl(codec, *p, cfg); if (err < 0) return err; } - err = add_jack_kctl(codec, cfg->dig_in_pin, cfg, lastname, &lastidx); + err = add_jack_kctl(codec, cfg->dig_in_pin, cfg); if (err < 0) return err; - err = add_jack_kctl(codec, cfg->mono_out_pin, cfg, lastname, &lastidx); + err = add_jack_kctl(codec, cfg->mono_out_pin, cfg); if (err < 0) return err; return 0; diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index c66655cf413a..a9803da633c0 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -12,6 +12,8 @@ #ifndef __SOUND_HDA_JACK_H #define __SOUND_HDA_JACK_H +struct auto_pin_cfg; + struct hda_jack_tbl { hda_nid_t nid; unsigned char action; /* event action (0 = none) */ @@ -21,6 +23,7 @@ struct hda_jack_tbl { unsigned int pin_sense; /* cached pin-sense value */ unsigned int jack_detect:1; /* capable of jack-detection? */ unsigned int jack_dirty:1; /* needs to update? */ + unsigned int phantom_jack:1; /* a fixed, always present port? */ struct snd_kcontrol *kctl; /* assigned kctl for jack-detection */ #ifdef CONFIG_SND_HDA_INPUT_JACK int type; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 0ec9248165bc..1b4c12941baa 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -89,7 +89,7 @@ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ .subdevice = HDA_SUBDEV_AMP_FLAG, \ .info = snd_hda_mixer_amp_switch_info, \ - .get = snd_hda_mixer_amp_switch_get, \ + .get = snd_hda_mixer_amp_switch_get_beep, \ .put = snd_hda_mixer_amp_switch_put_beep, \ .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) } #else @@ -121,6 +121,8 @@ int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); #ifdef CONFIG_SND_HDA_INPUT_BEEP +int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); #endif @@ -262,6 +264,8 @@ int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *imux, struct snd_ctl_elem_value *ucontrol, hda_nid_t nid, unsigned int *cur_val); +int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label, + int index, int *type_index_ret); /* * Channel mode helper @@ -393,72 +397,7 @@ struct hda_bus_unsolicited { struct hda_bus *bus; }; -/* - * Helper for automatic pin configuration - */ - -enum { - AUTO_PIN_MIC, - AUTO_PIN_LINE_IN, - AUTO_PIN_CD, - AUTO_PIN_AUX, - AUTO_PIN_LAST -}; - -enum { - AUTO_PIN_LINE_OUT, - AUTO_PIN_SPEAKER_OUT, - AUTO_PIN_HP_OUT -}; - -#define AUTO_CFG_MAX_OUTS HDA_MAX_OUTS -#define AUTO_CFG_MAX_INS 8 - -struct auto_pin_cfg_item { - hda_nid_t pin; - int type; -}; - -struct auto_pin_cfg; -const char *hda_get_autocfg_input_label(struct hda_codec *codec, - const struct auto_pin_cfg *cfg, - int input); -int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, - const struct auto_pin_cfg *cfg, - char *label, int maxlen, int *indexp); -int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label, - int index, int *type_index_ret); - -enum { - INPUT_PIN_ATTR_UNUSED, /* pin not connected */ - INPUT_PIN_ATTR_INT, /* internal mic/line-in */ - INPUT_PIN_ATTR_DOCK, /* docking mic/line-in */ - INPUT_PIN_ATTR_NORMAL, /* mic/line-in jack */ - INPUT_PIN_ATTR_FRONT, /* mic/line-in jack in front */ - INPUT_PIN_ATTR_REAR, /* mic/line-in jack in rear */ -}; - -int snd_hda_get_input_pin_attr(unsigned int def_conf); - -struct auto_pin_cfg { - int line_outs; - /* sorted in the order of Front/Surr/CLFE/Side */ - hda_nid_t line_out_pins[AUTO_CFG_MAX_OUTS]; - int speaker_outs; - hda_nid_t speaker_pins[AUTO_CFG_MAX_OUTS]; - int hp_outs; - int line_out_type; /* AUTO_PIN_XXX_OUT */ - hda_nid_t hp_pins[AUTO_CFG_MAX_OUTS]; - int num_inputs; - struct auto_pin_cfg_item inputs[AUTO_CFG_MAX_INS]; - int dig_outs; - hda_nid_t dig_out_pins[2]; - hda_nid_t dig_in_pin; - hda_nid_t mono_out_pin; - int dig_out_type[2]; /* HDA_PCM_TYPE_XXX */ - int dig_in_type; /* HDA_PCM_TYPE_XXX */ -}; - +/* helper macros to retrieve pin default-config values */ #define get_defcfg_connect(cfg) \ ((cfg & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT) #define get_defcfg_association(cfg) \ @@ -472,19 +411,6 @@ struct auto_pin_cfg { #define get_defcfg_misc(cfg) \ ((cfg & AC_DEFCFG_MISC) >> AC_DEFCFG_MISC_SHIFT) -/* bit-flags for snd_hda_parse_pin_def_config() behavior */ -#define HDA_PINCFG_NO_HP_FIXUP (1 << 0) /* no HP-split */ -#define HDA_PINCFG_NO_LO_FIXUP (1 << 1) /* don't take other outs as LO */ - -int snd_hda_parse_pin_defcfg(struct hda_codec *codec, - struct auto_pin_cfg *cfg, - const hda_nid_t *ignore_nids, - unsigned int cond_flags); - -/* older function */ -#define snd_hda_parse_pin_def_config(codec, cfg, ignore) \ - snd_hda_parse_pin_defcfg(codec, cfg, ignore, 0) - /* amp values */ #define AMP_IN_MUTE(idx) (0x7080 | ((idx)<<8)) #define AMP_IN_UNMUTE(idx) (0x7000 | ((idx)<<8)) @@ -502,6 +428,46 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, #define PIN_HP (AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN) #define PIN_HP_AMP (AC_PINCTL_HP_EN) +unsigned int snd_hda_get_default_vref(struct hda_codec *codec, hda_nid_t pin); +int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin, + unsigned int val, bool cached); + +/** + * _snd_hda_set_pin_ctl - Set a pin-control value safely + * @codec: the codec instance + * @pin: the pin NID to set the control + * @val: the pin-control value (AC_PINCTL_* bits) + * + * This function sets the pin-control value to the given pin, but + * filters out the invalid pin-control bits when the pin has no such + * capabilities. For example, when PIN_HP is passed but the pin has no + * HP-drive capability, the HP bit is omitted. + * + * The function doesn't check the input VREF capability bits, though. + * Use snd_hda_get_default_vref() to guess the right value. + * Also, this function is only for analog pins, not for HDMI pins. + */ +static inline int +snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin, unsigned int val) +{ + return _snd_hda_set_pin_ctl(codec, pin, val, false); +} + +/** + * snd_hda_set_pin_ctl_cache - Set a pin-control value safely + * @codec: the codec instance + * @pin: the pin NID to set the control + * @val: the pin-control value (AC_PINCTL_* bits) + * + * Just like snd_hda_set_pin_ctl() but write to cache as well. + */ +static inline int +snd_hda_set_pin_ctl_cache(struct hda_codec *codec, hda_nid_t pin, + unsigned int val) +{ + return _snd_hda_set_pin_ctl(codec, pin, val, true); +} + /* * get widget capabilities */ diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index e59e2f059b6e..7e46258fc700 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -426,10 +426,10 @@ static void print_digital_conv(struct snd_info_buffer *buffer, static const char *get_pwr_state(u32 state) { - static const char * const buf[4] = { - "D0", "D1", "D2", "D3" + static const char * const buf[] = { + "D0", "D1", "D2", "D3", "D3cold" }; - if (state < 4) + if (state < ARRAY_SIZE(buf)) return buf[state]; return "UNKNOWN"; } @@ -451,14 +451,21 @@ static void print_power_state(struct snd_info_buffer *buffer, int sup = snd_hda_param_read(codec, nid, AC_PAR_POWER_STATE); int pwr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); - if (sup) + if (sup != -1) snd_iprintf(buffer, " Power states: %s\n", bits_names(sup, names, ARRAY_SIZE(names))); - snd_iprintf(buffer, " Power: setting=%s, actual=%s\n", + snd_iprintf(buffer, " Power: setting=%s, actual=%s", get_pwr_state(pwr & AC_PWRST_SETTING), get_pwr_state((pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT)); + if (pwr & AC_PWRST_ERROR) + snd_iprintf(buffer, ", Error"); + if (pwr & AC_PWRST_CLK_STOP_OK) + snd_iprintf(buffer, ", Clock-stop-OK"); + if (pwr & AC_PWRST_SETTING_RESET) + snd_iprintf(buffer, ", Setting-reset"); + snd_iprintf(buffer, "\n"); } static void print_unsol_cap(struct snd_info_buffer *buffer, diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 7143393927da..0208fa121e5a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -28,6 +28,7 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_beep.h" #include "hda_jack.h" @@ -641,7 +642,7 @@ static void ad198x_free(struct hda_codec *codec) } #ifdef CONFIG_PM -static int ad198x_suspend(struct hda_codec *codec, pm_message_t state) +static int ad198x_suspend(struct hda_codec *codec) { ad198x_shutup(codec); return 0; @@ -1742,9 +1743,7 @@ static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol, if (! ad198x_eapd_put(kcontrol, ucontrol)) return 0; /* change speaker pin appropriately */ - snd_hda_codec_write(codec, 0x05, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - spec->cur_eapd ? PIN_OUT : 0); + snd_hda_set_pin_ctl(codec, 0x05, spec->cur_eapd ? PIN_OUT : 0); /* toggle HP mute appropriately */ snd_hda_codec_amp_stereo(codec, 0x06, HDA_OUTPUT, 0, HDA_AMP_MUTE, @@ -3103,7 +3102,7 @@ static void ad1988_auto_set_output_and_unmute(struct hda_codec *codec, int dac_idx) { /* set as output */ - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); + snd_hda_set_pin_ctl(codec, nid, pin_type); snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); switch (nid) { case 0x11: /* port-A - DAC 03 */ @@ -3157,6 +3156,7 @@ static void ad1988_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; int type = cfg->inputs[i].type; + int val; switch (nid) { case 0x15: /* port-C */ snd_hda_codec_write(codec, 0x33, 0, AC_VERB_SET_CONNECT_SEL, 0x0); @@ -3165,8 +3165,10 @@ static void ad1988_auto_init_analog_input(struct hda_codec *codec) snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_CONNECT_SEL, 0x0); break; } - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - type == AUTO_PIN_MIC ? PIN_VREF80 : PIN_IN); + val = PIN_IN; + if (type == AUTO_PIN_MIC) + val |= snd_hda_get_default_vref(codec, nid); + snd_hda_set_pin_ctl(codec, nid, val); if (nid != AD1988_PIN_CD_NID) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 09ccfabb4a17..19ae14f739cb 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -26,6 +26,7 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" /* */ @@ -341,8 +342,7 @@ static int ca0110_build_pcms(struct hda_codec *codec) static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) { if (pin) { - snd_hda_codec_write(codec, pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + snd_hda_set_pin_ctl(codec, pin, PIN_HP); if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -356,8 +356,8 @@ static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc) { if (pin) { - snd_hda_codec_write(codec, pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80); + snd_hda_set_pin_ctl(codec, pin, PIN_IN | + snd_hda_get_default_vref(codec, pin)); if (get_wcaps(codec, pin) & AC_WCAP_IN_AMP) snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 21d91d580da8..d0d3540e39e7 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -30,6 +30,7 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #define WIDGET_CHIP_CTRL 0x15 #define WIDGET_DSP_CTRL 0x16 @@ -239,8 +240,7 @@ enum get_set { static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) { if (pin) { - snd_hda_codec_write(codec, pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + snd_hda_set_pin_ctl(codec, pin, PIN_HP); if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -254,9 +254,8 @@ static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc) { if (pin) { - snd_hda_codec_write(codec, pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - PIN_VREF80); + snd_hda_set_pin_ctl(codec, pin, PIN_IN | + snd_hda_get_default_vref(codec, pin)); if (get_wcaps(codec, pin) & AC_WCAP_IN_AMP) snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index c83ccdba1e5a..0c4c1a61b378 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -26,6 +26,7 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_jack.h" #include <sound/tlv.h> @@ -933,8 +934,7 @@ static void cs_automute(struct hda_codec *codec) pin_ctl = 0; nid = cfg->speaker_pins[i]; - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, pin_ctl); + snd_hda_set_pin_ctl(codec, nid, pin_ctl); } if (spec->gpio_eapd_hp) { unsigned int gpio = hp_present ? @@ -948,16 +948,14 @@ static void cs_automute(struct hda_codec *codec) /* mute HPs if spdif jack (SENSE_B) is present */ for (i = 0; i < cfg->hp_outs; i++) { nid = cfg->hp_pins[i]; - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, + snd_hda_set_pin_ctl(codec, nid, (spdif_present && spec->sense_b) ? 0 : PIN_HP); } /* SPDIF TX on/off */ if (cfg->dig_outs) { nid = cfg->dig_out_pins[0]; - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, + snd_hda_set_pin_ctl(codec, nid, spdif_present ? PIN_OUT : 0); } @@ -1024,13 +1022,11 @@ static void init_output(struct hda_codec *codec) /* set appropriate pin controls */ for (i = 0; i < cfg->line_outs; i++) - snd_hda_codec_write(codec, cfg->line_out_pins[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + snd_hda_set_pin_ctl(codec, cfg->line_out_pins[i], PIN_OUT); /* HP */ for (i = 0; i < cfg->hp_outs; i++) { hda_nid_t nid = cfg->hp_pins[i]; - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + snd_hda_set_pin_ctl(codec, nid, PIN_HP); if (!cfg->speaker_outs) continue; if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { @@ -1041,8 +1037,7 @@ static void init_output(struct hda_codec *codec) /* Speaker */ for (i = 0; i < cfg->speaker_outs; i++) - snd_hda_codec_write(codec, cfg->speaker_pins[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + snd_hda_set_pin_ctl(codec, cfg->speaker_pins[i], PIN_OUT); /* SPDIF is enabled on presence detect for CS421x */ if (spec->hp_detect || spec->spdif_detect) @@ -1063,14 +1058,9 @@ static void init_input(struct hda_codec *codec) continue; /* set appropriate pin control and mute first */ ctl = PIN_IN; - if (cfg->inputs[i].type == AUTO_PIN_MIC) { - unsigned int caps = snd_hda_query_pin_caps(codec, pin); - caps >>= AC_PINCAP_VREF_SHIFT; - if (caps & AC_PINCAP_VREF_80) - ctl = PIN_VREF80; - } - snd_hda_codec_write(codec, pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, ctl); + if (cfg->inputs[i].type == AUTO_PIN_MIC) + ctl |= snd_hda_get_default_vref(codec, pin); + snd_hda_set_pin_ctl(codec, pin, ctl); snd_hda_codec_write(codec, spec->adc_nid[i], 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(spec->adc_idx[i])); @@ -1902,7 +1892,7 @@ static int cs421x_parse_auto_config(struct hda_codec *codec) Manage PDREF, when transitioning to D3hot (DAC,ADC) -> D3, PDREF=1, AFG->D3 */ -static int cs421x_suspend(struct hda_codec *codec, pm_message_t state) +static int cs421x_suspend(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; unsigned int coef; diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index b6767b4ced44..c8fdaaefe702 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -29,6 +29,7 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #define NUM_PINS 11 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d906c5b74cf0..14361184ae1e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -30,6 +30,7 @@ #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_beep.h" #include "hda_jack.h" @@ -66,6 +67,7 @@ struct imux_info { }; struct conexant_spec { + struct hda_gen_spec gen; const struct snd_kcontrol_new *mixers[5]; int num_mixers; @@ -141,6 +143,7 @@ struct conexant_spec { unsigned int hp_laptop:1; unsigned int asus:1; unsigned int pin_eapd_ctrls:1; + unsigned int fixup_stereo_dmic:1; unsigned int adc_switching:1; @@ -442,8 +445,10 @@ static int conexant_init(struct hda_codec *codec) static void conexant_free(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; + snd_hda_gen_free(&spec->gen); snd_hda_detach_beep_device(codec); - kfree(codec->spec); + kfree(spec); } static const struct snd_kcontrol_new cxt_capture_mixers[] = { @@ -549,7 +554,7 @@ static int conexant_build_controls(struct hda_codec *codec) } #ifdef CONFIG_SND_HDA_POWER_SAVE -static int conexant_suspend(struct hda_codec *codec, pm_message_t state) +static int conexant_suspend(struct hda_codec *codec) { snd_hda_shutup_pins(codec); return 0; @@ -1601,17 +1606,13 @@ static void cxt5051_update_speaker(struct hda_codec *codec) unsigned int pinctl; /* headphone pin */ pinctl = (spec->hp_present && spec->cur_eapd) ? PIN_HP : 0; - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pinctl); + snd_hda_set_pin_ctl(codec, 0x16, pinctl); /* speaker pin */ pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pinctl); + snd_hda_set_pin_ctl(codec, 0x1a, pinctl); /* on ideapad there is an additional speaker (subwoofer) to mute */ if (spec->ideapad) - snd_hda_codec_write(codec, 0x1b, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pinctl); + snd_hda_set_pin_ctl(codec, 0x1b, pinctl); } /* turn on/off EAPD (+ mute HP) as a master switch */ @@ -1996,8 +1997,7 @@ static void cxt5066_update_speaker(struct hda_codec *codec) /* Port A (HP) */ pinctl = (hp_port_a_present(spec) && spec->cur_eapd) ? PIN_HP : 0; - snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pinctl); + snd_hda_set_pin_ctl(codec, 0x19, pinctl); /* Port D (HP/LO) */ pinctl = spec->cur_eapd ? spec->port_d_mode : 0; @@ -2010,13 +2010,11 @@ static void cxt5066_update_speaker(struct hda_codec *codec) if (!hp_port_d_present(spec)) pinctl = 0; } - snd_hda_codec_write(codec, 0x1c, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pinctl); + snd_hda_set_pin_ctl(codec, 0x1c, pinctl); /* CLASS_D AMP */ pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pinctl); + snd_hda_set_pin_ctl(codec, 0x1f, pinctl); } /* turn on/off EAPD (+ mute HP) as a master switch */ @@ -2047,8 +2045,7 @@ static int cxt5066_set_olpc_dc_bias(struct hda_codec *codec) /* Even though port F is the DC input, the bias is controlled on port B. * we also leave that port as an active input (but unselected) in DC mode * just in case that is necessary to make the bias setting take effect. */ - return snd_hda_codec_write_cache(codec, 0x1a, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, + return snd_hda_set_pin_ctl_cache(codec, 0x1a, cxt5066_olpc_dc_bias.items[spec->dc_input_bias].index); } @@ -2081,14 +2078,14 @@ static void cxt5066_olpc_select_mic(struct hda_codec *codec) } /* disable DC (port F) */ - snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, 0x1e, 0); /* external mic, port B */ - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + snd_hda_set_pin_ctl(codec, 0x1a, spec->ext_mic_present ? CXT5066_OLPC_EXT_MIC_BIAS : 0); /* internal mic, port C */ - snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + snd_hda_set_pin_ctl(codec, 0x1b, spec->ext_mic_present ? 0 : PIN_VREF80); } @@ -3357,9 +3354,7 @@ static void do_automute(struct hda_codec *codec, int num_pins, struct conexant_spec *spec = codec->spec; int i; for (i = 0; i < num_pins; i++) - snd_hda_codec_write(codec, pins[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - on ? PIN_OUT : 0); + snd_hda_set_pin_ctl(codec, pins[i], on ? PIN_OUT : 0); if (spec->pin_eapd_ctrls) cx_auto_turn_eapd(codec, num_pins, pins, on); } @@ -3976,8 +3971,7 @@ static void cx_auto_init_output(struct hda_codec *codec) if (snd_hda_query_pin_caps(codec, cfg->hp_pins[i]) & AC_PINCAP_HP_DRV) val |= AC_PINCTL_HP_EN; - snd_hda_codec_write(codec, cfg->hp_pins[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, val); + snd_hda_set_pin_ctl(codec, cfg->hp_pins[i], val); } mute_outputs(codec, cfg->hp_outs, cfg->hp_pins); mute_outputs(codec, cfg->line_outs, cfg->line_out_pins); @@ -4030,13 +4024,11 @@ static void cx_auto_init_input(struct hda_codec *codec) } for (i = 0; i < cfg->num_inputs; i++) { - unsigned int type; + hda_nid_t pin = cfg->inputs[i].pin; + unsigned int type = PIN_IN; if (cfg->inputs[i].type == AUTO_PIN_MIC) - type = PIN_VREF80; - else - type = PIN_IN; - snd_hda_codec_write(codec, cfg->inputs[i].pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, type); + type |= snd_hda_get_default_vref(codec, pin); + snd_hda_set_pin_ctl(codec, pin, type); } if (spec->auto_mic) { @@ -4063,17 +4055,15 @@ static void cx_auto_init_digital(struct hda_codec *codec) struct auto_pin_cfg *cfg = &spec->autocfg; if (spec->multiout.dig_out_nid) - snd_hda_codec_write(codec, cfg->dig_out_pins[0], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + snd_hda_set_pin_ctl(codec, cfg->dig_out_pins[0], PIN_OUT); if (spec->dig_in_nid) - snd_hda_codec_write(codec, cfg->dig_in_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN); + snd_hda_set_pin_ctl(codec, cfg->dig_in_pin, PIN_IN); } static int cx_auto_init(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - /*snd_hda_sequence_write(codec, cx_auto_init_verbs);*/ + snd_hda_gen_apply_verbs(codec); cx_auto_init_output(codec); cx_auto_init_input(codec); cx_auto_init_digital(codec); @@ -4084,9 +4074,9 @@ static int cx_auto_init(struct hda_codec *codec) static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, const char *dir, int cidx, - hda_nid_t nid, int hda_dir, int amp_idx) + hda_nid_t nid, int hda_dir, int amp_idx, int chs) { - static char name[32]; + static char name[44]; static struct snd_kcontrol_new knew[] = { HDA_CODEC_VOLUME(name, 0, 0, 0), HDA_CODEC_MUTE(name, 0, 0, 0), @@ -4096,7 +4086,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, for (i = 0; i < 2; i++) { struct snd_kcontrol *kctl; - knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx, + knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, chs, amp_idx, hda_dir); knew[i].subdevice = HDA_SUBDEV_AMP_FLAG; knew[i].index = cidx; @@ -4115,7 +4105,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, } #define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir) \ - cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0) + cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0, 3) #define cx_auto_add_pb_volume(codec, nid, str, idx) \ cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT) @@ -4185,6 +4175,36 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) return 0; } +/* Returns zero if this is a normal stereo channel, and non-zero if it should + be split in two independent channels. + dest_label must be at least 44 characters. */ +static int cx_auto_get_rightch_label(struct hda_codec *codec, const char *label, + char *dest_label, int nid) +{ + struct conexant_spec *spec = codec->spec; + int i; + + if (!spec->fixup_stereo_dmic) + return 0; + + for (i = 0; i < AUTO_CFG_MAX_INS; i++) { + int def_conf; + if (spec->autocfg.inputs[i].pin != nid) + continue; + + if (spec->autocfg.inputs[i].type != AUTO_PIN_MIC) + return 0; + def_conf = snd_hda_codec_get_pincfg(codec, nid); + if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT) + return 0; + + /* Finally found the inverted internal mic! */ + snprintf(dest_label, 44, "Inverted %s", label); + return 1; + } + return 0; +} + static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, const char *label, const char *pfx, int cidx) @@ -4193,14 +4213,25 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, int i; for (i = 0; i < spec->num_adc_nids; i++) { + char rightch_label[44]; hda_nid_t adc_nid = spec->adc_nids[i]; int idx = get_input_connection(codec, adc_nid, nid); if (idx < 0) continue; if (codec->single_adc_amp) idx = 0; + + if (cx_auto_get_rightch_label(codec, label, rightch_label, nid)) { + /* Make two independent kcontrols for left and right */ + int err = cx_auto_add_volume_idx(codec, label, pfx, + cidx, adc_nid, HDA_INPUT, idx, 1); + if (err < 0) + return err; + return cx_auto_add_volume_idx(codec, rightch_label, pfx, + cidx, adc_nid, HDA_INPUT, idx, 2); + } return cx_auto_add_volume_idx(codec, label, pfx, - cidx, adc_nid, HDA_INPUT, idx); + cidx, adc_nid, HDA_INPUT, idx, 3); } return 0; } @@ -4213,9 +4244,19 @@ static int cx_auto_add_boost_volume(struct hda_codec *codec, int idx, int i, con; nid = spec->imux_info[idx].pin; - if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) + if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) { + char rightch_label[44]; + if (cx_auto_get_rightch_label(codec, label, rightch_label, nid)) { + int err = cx_auto_add_volume_idx(codec, label, " Boost", + cidx, nid, HDA_INPUT, 0, 1); + if (err < 0) + return err; + return cx_auto_add_volume_idx(codec, rightch_label, " Boost", + cidx, nid, HDA_INPUT, 0, 2); + } return cx_auto_add_volume(codec, label, " Boost", cidx, nid, HDA_INPUT); + } con = __select_input_connection(codec, spec->imux_info[idx].adc, nid, &mux, false, 0); if (con < 0) @@ -4370,37 +4411,21 @@ static const struct hda_codec_ops cx_auto_patch_ops = { /* * pin fix-up */ -struct cxt_pincfg { - hda_nid_t nid; - u32 val; -}; - -static void apply_pincfg(struct hda_codec *codec, const struct cxt_pincfg *cfg) -{ - for (; cfg->nid; cfg++) - snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); - -} - -static void apply_pin_fixup(struct hda_codec *codec, - const struct snd_pci_quirk *quirk, - const struct cxt_pincfg **table) -{ - quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); - if (quirk) { - snd_printdd(KERN_INFO "hda_codec: applying pincfg for %s\n", - quirk->name); - apply_pincfg(codec, table[quirk->value]); - } -} - enum { CXT_PINCFG_LENOVO_X200, CXT_PINCFG_LENOVO_TP410, + CXT_FIXUP_STEREO_DMIC, }; +static void cxt_fixup_stereo_dmic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + spec->fixup_stereo_dmic = 1; +} + /* ThinkPad X200 & co with cxt5051 */ -static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { +static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ { 0x17, 0x21a11000 }, /* dock-mic */ { 0x19, 0x2121103f }, /* dock-HP */ @@ -4409,16 +4434,26 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { }; /* ThinkPad 410/420/510/520, X201 & co with cxt5066 */ -static const struct cxt_pincfg cxt_pincfg_lenovo_tp410[] = { +static const struct hda_pintbl cxt_pincfg_lenovo_tp410[] = { { 0x19, 0x042110ff }, /* HP (seq# overridden) */ { 0x1a, 0x21a190f0 }, /* dock-mic */ { 0x1c, 0x212140ff }, /* dock-HP */ {} }; -static const struct cxt_pincfg *cxt_pincfg_tbl[] = { - [CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200, - [CXT_PINCFG_LENOVO_TP410] = cxt_pincfg_lenovo_tp410, +static const struct hda_fixup cxt_fixups[] = { + [CXT_PINCFG_LENOVO_X200] = { + .type = HDA_FIXUP_PINS, + .v.pins = cxt_pincfg_lenovo_x200, + }, + [CXT_PINCFG_LENOVO_TP410] = { + .type = HDA_FIXUP_PINS, + .v.pins = cxt_pincfg_lenovo_tp410, + }, + [CXT_FIXUP_STEREO_DMIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_stereo_dmic, + }, }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -4432,6 +4467,8 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), {} }; @@ -4463,6 +4500,7 @@ static int patch_conexant_auto(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; + snd_hda_gen_init(&spec->gen); switch (codec->vendor_id) { case 0x14f15045: @@ -4471,13 +4509,16 @@ static int patch_conexant_auto(struct hda_codec *codec) case 0x14f15051: add_cx5051_fake_mutes(codec); codec->pin_amp_workaround = 1; - apply_pin_fixup(codec, cxt5051_fixups, cxt_pincfg_tbl); + snd_hda_pick_fixup(codec, NULL, cxt5051_fixups, cxt_fixups); break; default: codec->pin_amp_workaround = 1; - apply_pin_fixup(codec, cxt5066_fixups, cxt_pincfg_tbl); + snd_hda_pick_fixup(codec, NULL, cxt5066_fixups, cxt_fixups); + break; } + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + /* Show mute-led control only on HP laptops * This is a sort of white-list: on HP laptops, EAPD corresponds * only to the mute-LED without actualy amp function. Meanwhile, @@ -4556,6 +4597,12 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_conexant_auto }, { .id = 0x14f150b9, .name = "CX20665", .patch = patch_conexant_auto }, + { .id = 0x14f1510f, .name = "CX20751/2", + .patch = patch_conexant_auto }, + { .id = 0x14f15110, .name = "CX20751/2", + .patch = patch_conexant_auto }, + { .id = 0x14f15111, .name = "CX20753/4", + .patch = patch_conexant_auto }, {} /* terminator */ }; @@ -4576,6 +4623,9 @@ MODULE_ALIAS("snd-hda-codec-id:14f150ab"); MODULE_ALIAS("snd-hda-codec-id:14f150ac"); MODULE_ALIAS("snd-hda-codec-id:14f150b8"); MODULE_ALIAS("snd-hda-codec-id:14f150b9"); +MODULE_ALIAS("snd-hda-codec-id:14f1510f"); +MODULE_ALIAS("snd-hda-codec-id:14f15110"); +MODULE_ALIAS("snd-hda-codec-id:14f15111"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 83f345f3c961..641408dc28c0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -85,7 +85,7 @@ struct hdmi_spec { * Non-generic ATI/NVIDIA specific */ struct hda_multi_out multiout; - const struct hda_pcm_stream *pcm_playback; + struct hda_pcm_stream pcm_playback; }; @@ -787,7 +787,7 @@ static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) int cp_ready = !!(res & AC_UNSOL_RES_CP_READY); printk(KERN_INFO - "HDMI CP event: CODEC=%d PIN=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n", + "HDMI CP event: CODEC=%d TAG=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n", codec->addr, tag, subtag, @@ -876,7 +876,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, struct hdmi_spec_per_pin *per_pin; struct hdmi_eld *eld; struct hdmi_spec_per_cvt *per_cvt = NULL; - int pinctl; /* Validate hinfo */ pin_idx = hinfo_to_pin_index(spec, hinfo); @@ -912,11 +911,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, snd_hda_codec_write(codec, per_pin->pin_nid, 0, AC_VERB_SET_CONNECT_SEL, mux_idx); - pinctl = snd_hda_codec_read(codec, per_pin->pin_nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - snd_hda_codec_write(codec, per_pin->pin_nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pinctl | PIN_OUT); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); /* Initially set the converter's capabilities */ @@ -1153,11 +1147,17 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hdmi_spec *spec = codec->spec; int pin_idx = hinfo_to_pin_index(spec, hinfo); hda_nid_t pin_nid = spec->pins[pin_idx].pin_nid; + int pinctl; hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); hdmi_setup_audio_infoframe(codec, pin_idx, substream); + pinctl = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl | PIN_OUT); + return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } @@ -1277,23 +1277,34 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) return 0; } -static int generic_hdmi_init(struct hda_codec *codec) +static int generic_hdmi_init_per_pins(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; int pin_idx; for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; - hda_nid_t pin_nid = per_pin->pin_nid; struct hdmi_eld *eld = &per_pin->sink_eld; - hdmi_init_pin(codec, pin_nid); - snd_hda_jack_detect_enable(codec, pin_nid, pin_nid); - per_pin->codec = codec; INIT_DELAYED_WORK(&per_pin->work, hdmi_repoll_eld); snd_hda_eld_proc_new(codec, eld, pin_idx); } + return 0; +} + +static int generic_hdmi_init(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; + hda_nid_t pin_nid = per_pin->pin_nid; + + hdmi_init_pin(codec, pin_nid); + snd_hda_jack_detect_enable(codec, pin_nid, pin_nid); + } snd_hda_jack_report_sync(codec); return 0; } @@ -1338,6 +1349,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + generic_hdmi_init_per_pins(codec); init_channel_allocations(); @@ -1352,45 +1364,65 @@ static int simple_playback_build_pcms(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; struct hda_pcm *info = spec->pcm_rec; - int i; + unsigned int chans; + struct hda_pcm_stream *pstr; - codec->num_pcms = spec->num_cvts; + codec->num_pcms = 1; codec->pcm_info = info; - for (i = 0; i < codec->num_pcms; i++, info++) { - unsigned int chans; - struct hda_pcm_stream *pstr; - - chans = get_wcaps(codec, spec->cvts[i].cvt_nid); - chans = get_wcaps_channels(chans); + chans = get_wcaps(codec, spec->cvts[0].cvt_nid); + chans = get_wcaps_channels(chans); - info->name = get_hdmi_pcm_name(i); - info->pcm_type = HDA_PCM_TYPE_HDMI; - pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK]; - snd_BUG_ON(!spec->pcm_playback); - *pstr = *spec->pcm_playback; - pstr->nid = spec->cvts[i].cvt_nid; - if (pstr->channels_max <= 2 && chans && chans <= 16) - pstr->channels_max = chans; - } + info->name = get_hdmi_pcm_name(0); + info->pcm_type = HDA_PCM_TYPE_HDMI; + pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK]; + *pstr = spec->pcm_playback; + pstr->nid = spec->cvts[0].cvt_nid; + if (pstr->channels_max <= 2 && chans && chans <= 16) + pstr->channels_max = chans; return 0; } +/* unsolicited event for jack sensing */ +static void simple_hdmi_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + snd_hda_jack_set_dirty_all(codec); + snd_hda_jack_report_sync(codec); +} + +/* generic_hdmi_build_jack can be used for simple_hdmi, too, + * as long as spec->pins[] is set correctly + */ +#define simple_hdmi_build_jack generic_hdmi_build_jack + static int simple_playback_build_controls(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; int err; - int i; - for (i = 0; i < codec->num_pcms; i++) { - err = snd_hda_create_spdif_out_ctls(codec, - spec->cvts[i].cvt_nid, - spec->cvts[i].cvt_nid); - if (err < 0) - return err; - } + err = snd_hda_create_spdif_out_ctls(codec, + spec->cvts[0].cvt_nid, + spec->cvts[0].cvt_nid); + if (err < 0) + return err; + return simple_hdmi_build_jack(codec, 0); +} +static int simple_playback_init(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + hda_nid_t pin = spec->pins[0].pin_nid; + + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + /* some codecs require to unmute the pin */ + if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + snd_hda_jack_detect_enable(codec, pin, pin); + snd_hda_jack_report_sync(codec); return 0; } @@ -1418,7 +1450,15 @@ static const hda_nid_t nvhdmi_con_nids_7x[4] = { 0x6, 0x8, 0xa, 0xc, }; -static const struct hda_verb nvhdmi_basic_init_7x[] = { +static const struct hda_verb nvhdmi_basic_init_7x_2ch[] = { + /* set audio protect on */ + { 0x1, Nv_VERB_SET_Audio_Protection_On, 0x1}, + /* enable digital output on pin widget */ + { 0x5, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, + {} /* terminator */ +}; + +static const struct hda_verb nvhdmi_basic_init_7x_8ch[] = { /* set audio protect on */ { 0x1, Nv_VERB_SET_Audio_Protection_On, 0x1}, /* enable digital output on pin widget */ @@ -1446,9 +1486,15 @@ static const struct hda_verb nvhdmi_basic_init_7x[] = { (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) #endif -static int nvhdmi_7x_init(struct hda_codec *codec) +static int nvhdmi_7x_init_2ch(struct hda_codec *codec) { - snd_hda_sequence_write(codec, nvhdmi_basic_init_7x); + snd_hda_sequence_write(codec, nvhdmi_basic_init_7x_2ch); + return 0; +} + +static int nvhdmi_7x_init_8ch(struct hda_codec *codec) +{ + snd_hda_sequence_write(codec, nvhdmi_basic_init_7x_8ch); return 0; } @@ -1524,6 +1570,50 @@ static int simple_playback_pcm_prepare(struct hda_pcm_stream *hinfo, stream_tag, format, substream); } +static const struct hda_pcm_stream simple_pcm_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .ops = { + .open = simple_playback_pcm_open, + .close = simple_playback_pcm_close, + .prepare = simple_playback_pcm_prepare + }, +}; + +static const struct hda_codec_ops simple_hdmi_patch_ops = { + .build_controls = simple_playback_build_controls, + .build_pcms = simple_playback_build_pcms, + .init = simple_playback_init, + .free = simple_playback_free, + .unsol_event = simple_hdmi_unsol_event, +}; + +static int patch_simple_hdmi(struct hda_codec *codec, + hda_nid_t cvt_nid, hda_nid_t pin_nid) +{ + struct hdmi_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + + codec->spec = spec; + + spec->multiout.num_dacs = 0; /* no analog */ + spec->multiout.max_channels = 2; + spec->multiout.dig_out_nid = cvt_nid; + spec->num_cvts = 1; + spec->num_pins = 1; + spec->cvts[0].cvt_nid = cvt_nid; + spec->pins[0].pin_nid = pin_nid; + spec->pcm_playback = simple_pcm_playback; + + codec->patch_ops = simple_hdmi_patch_ops; + + return 0; +} + static void nvhdmi_8ch_7x_set_info_frame_parameters(struct hda_codec *codec, int channels) { @@ -1592,10 +1682,10 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo, unsigned int dataDCC2, channel_id; int i; struct hdmi_spec *spec = codec->spec; - struct hda_spdif_out *spdif = - snd_hda_spdif_out_of_nid(codec, spec->cvts[0].cvt_nid); + struct hda_spdif_out *spdif; mutex_lock(&codec->spdif_mutex); + spdif = snd_hda_spdif_out_of_nid(codec, spec->cvts[0].cvt_nid); chs = substream->runtime->channels; @@ -1696,54 +1786,20 @@ static const struct hda_pcm_stream nvhdmi_pcm_playback_8ch_7x = { }, }; -static const struct hda_pcm_stream nvhdmi_pcm_playback_2ch = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = nvhdmi_master_con_nid_7x, - .rates = SUPPORTED_RATES, - .maxbps = SUPPORTED_MAXBPS, - .formats = SUPPORTED_FORMATS, - .ops = { - .open = simple_playback_pcm_open, - .close = simple_playback_pcm_close, - .prepare = simple_playback_pcm_prepare - }, -}; - -static const struct hda_codec_ops nvhdmi_patch_ops_8ch_7x = { - .build_controls = simple_playback_build_controls, - .build_pcms = simple_playback_build_pcms, - .init = nvhdmi_7x_init, - .free = simple_playback_free, -}; - -static const struct hda_codec_ops nvhdmi_patch_ops_2ch = { - .build_controls = simple_playback_build_controls, - .build_pcms = simple_playback_build_pcms, - .init = nvhdmi_7x_init, - .free = simple_playback_free, -}; - static int patch_nvhdmi_2ch(struct hda_codec *codec) { struct hdmi_spec *spec; + int err = patch_simple_hdmi(codec, nvhdmi_master_con_nid_7x, + nvhdmi_master_pin_nid_7x); + if (err < 0) + return err; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; - - spec->multiout.num_dacs = 0; /* no analog */ - spec->multiout.max_channels = 2; - spec->multiout.dig_out_nid = nvhdmi_master_con_nid_7x; - spec->num_cvts = 1; - spec->cvts[0].cvt_nid = nvhdmi_master_con_nid_7x; - spec->pcm_playback = &nvhdmi_pcm_playback_2ch; - - codec->patch_ops = nvhdmi_patch_ops_2ch; - + codec->patch_ops.init = nvhdmi_7x_init_2ch; + /* override the PCM rates, etc, as the codec doesn't give full list */ + spec = codec->spec; + spec->pcm_playback.rates = SUPPORTED_RATES; + spec->pcm_playback.maxbps = SUPPORTED_MAXBPS; + spec->pcm_playback.formats = SUPPORTED_FORMATS; return 0; } @@ -1751,13 +1807,12 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) { struct hdmi_spec *spec; int err = patch_nvhdmi_2ch(codec); - if (err < 0) return err; spec = codec->spec; spec->multiout.max_channels = 8; - spec->pcm_playback = &nvhdmi_pcm_playback_8ch_7x; - codec->patch_ops = nvhdmi_patch_ops_8ch_7x; + spec->pcm_playback = nvhdmi_pcm_playback_8ch_7x; + codec->patch_ops.init = nvhdmi_7x_init_8ch; /* Initialize the audio infoframe channel mask and checksum to something * valid */ @@ -1801,69 +1856,26 @@ static int atihdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, return 0; } -static const struct hda_pcm_stream atihdmi_pcm_digital_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = ATIHDMI_CVT_NID, - .ops = { - .open = simple_playback_pcm_open, - .close = simple_playback_pcm_close, - .prepare = atihdmi_playback_pcm_prepare - }, -}; - -static const struct hda_verb atihdmi_basic_init[] = { - /* enable digital output on pin widget */ - { 0x03, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - {} /* terminator */ -}; - -static int atihdmi_init(struct hda_codec *codec) +static int patch_atihdmi(struct hda_codec *codec) { - struct hdmi_spec *spec = codec->spec; - - snd_hda_sequence_write(codec, atihdmi_basic_init); - /* SI codec requires to unmute the pin */ - if (get_wcaps(codec, spec->pins[0].pin_nid) & AC_WCAP_OUT_AMP) - snd_hda_codec_write(codec, spec->pins[0].pin_nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_UNMUTE); + struct hdmi_spec *spec; + int err = patch_simple_hdmi(codec, ATIHDMI_CVT_NID, ATIHDMI_PIN_NID); + if (err < 0) + return err; + spec = codec->spec; + spec->pcm_playback.ops.prepare = atihdmi_playback_pcm_prepare; return 0; } -static const struct hda_codec_ops atihdmi_patch_ops = { - .build_controls = simple_playback_build_controls, - .build_pcms = simple_playback_build_pcms, - .init = atihdmi_init, - .free = simple_playback_free, -}; +/* VIA HDMI Implementation */ +#define VIAHDMI_CVT_NID 0x02 /* audio converter1 */ +#define VIAHDMI_PIN_NID 0x03 /* HDMI output pin1 */ - -static int patch_atihdmi(struct hda_codec *codec) +static int patch_via_hdmi(struct hda_codec *codec) { - struct hdmi_spec *spec; - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; - - spec->multiout.num_dacs = 0; /* no analog */ - spec->multiout.max_channels = 2; - spec->multiout.dig_out_nid = ATIHDMI_CVT_NID; - spec->num_cvts = 1; - spec->cvts[0].cvt_nid = ATIHDMI_CVT_NID; - spec->pins[0].pin_nid = ATIHDMI_PIN_NID; - spec->pcm_playback = &atihdmi_pcm_digital_playback; - - codec->patch_ops = atihdmi_patch_ops; - - return 0; + return patch_simple_hdmi(codec, VIAHDMI_CVT_NID, VIAHDMI_PIN_NID); } - /* * patch entries */ @@ -1902,8 +1914,13 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_generic_hdmi }, +{ .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, +{ .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, +{ .id = 0x11069f81, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, +{ .id = 0x11069f84, .name = "VX11 HDMI/DP", .patch = patch_generic_hdmi }, +{ .id = 0x11069f85, .name = "VX11 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x80860054, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862801, .name = "Bearlake HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862802, .name = "Cantiga HDMI", .patch = patch_generic_hdmi }, @@ -1911,6 +1928,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x80862804, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862805, .name = "CougarPoint HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x80862807, .name = "Haswell HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi }, { .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi }, {} /* terminator */ @@ -1948,8 +1966,13 @@ MODULE_ALIAS("snd-hda-codec-id:10de0041"); MODULE_ALIAS("snd-hda-codec-id:10de0042"); MODULE_ALIAS("snd-hda-codec-id:10de0043"); MODULE_ALIAS("snd-hda-codec-id:10de0044"); +MODULE_ALIAS("snd-hda-codec-id:10de0051"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); +MODULE_ALIAS("snd-hda-codec-id:11069f80"); +MODULE_ALIAS("snd-hda-codec-id:11069f81"); +MODULE_ALIAS("snd-hda-codec-id:11069f84"); +MODULE_ALIAS("snd-hda-codec-id:11069f85"); MODULE_ALIAS("snd-hda-codec-id:17e80047"); MODULE_ALIAS("snd-hda-codec-id:80860054"); MODULE_ALIAS("snd-hda-codec-id:80862801"); @@ -1958,6 +1981,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862803"); MODULE_ALIAS("snd-hda-codec-id:80862804"); MODULE_ALIAS("snd-hda-codec-id:80862805"); MODULE_ALIAS("snd-hda-codec-id:80862806"); +MODULE_ALIAS("snd-hda-codec-id:80862807"); MODULE_ALIAS("snd-hda-codec-id:80862880"); MODULE_ALIAS("snd-hda-codec-id:808629fb"); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7810913d07a0..f141395dfee6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6,7 +6,7 @@ * Copyright (c) 2004 Kailang Yang <kailang@realtek.com.tw> * PeiSen Hou <pshou@realtek.com.tw> * Takashi Iwai <tiwai@suse.de> - * Jonathan Woithe <jwoithe@physics.adelaide.edu.au> + * Jonathan Woithe <jwoithe@just42.net> * * This driver is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -32,6 +32,7 @@ #include <sound/jack.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_beep.h" #include "hda_jack.h" @@ -66,8 +67,6 @@ struct alc_customize_define { unsigned int fixup:1; /* Means that this sku is set by driver, not read from hw */ }; -struct alc_fixup; - struct alc_multi_io { hda_nid_t pin; /* multi-io widget pin NID */ hda_nid_t dac; /* DAC to be connected */ @@ -82,19 +81,33 @@ enum { #define MAX_VOL_NIDS 0x40 +/* make compatible with old code */ +#define alc_apply_pincfgs snd_hda_apply_pincfgs +#define alc_apply_fixup snd_hda_apply_fixup +#define alc_pick_fixup snd_hda_pick_fixup +#define alc_fixup hda_fixup +#define alc_pincfg hda_pintbl +#define alc_model_fixup hda_model_fixup + +#define ALC_FIXUP_PINS HDA_FIXUP_PINS +#define ALC_FIXUP_VERBS HDA_FIXUP_VERBS +#define ALC_FIXUP_FUNC HDA_FIXUP_FUNC + +#define ALC_FIXUP_ACT_PRE_PROBE HDA_FIXUP_ACT_PRE_PROBE +#define ALC_FIXUP_ACT_PROBE HDA_FIXUP_ACT_PROBE +#define ALC_FIXUP_ACT_INIT HDA_FIXUP_ACT_INIT +#define ALC_FIXUP_ACT_BUILD HDA_FIXUP_ACT_BUILD + + struct alc_spec { + struct hda_gen_spec gen; + /* codec parameterization */ const struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ unsigned int num_mixers; const struct snd_kcontrol_new *cap_mixer; /* capture mixer */ unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ - const struct hda_verb *init_verbs[10]; /* initialization verbs - * don't forget NULL - * termination! - */ - unsigned int num_init_verbs; - char stream_name_analog[32]; /* analog PCM stream */ const struct hda_pcm_stream *stream_analog_playback; const struct hda_pcm_stream *stream_analog_capture; @@ -157,10 +170,10 @@ struct alc_spec { hda_nid_t imux_pins[HDA_MAX_NUM_INPUTS]; unsigned int dyn_adc_idx[HDA_MAX_NUM_INPUTS]; int int_mic_idx, ext_mic_idx, dock_mic_idx; /* for auto-mic */ + hda_nid_t inv_dmic_pin; /* hooks */ void (*init_hook)(struct hda_codec *codec); - void (*unsol_event)(struct hda_codec *codec, unsigned int res); #ifdef CONFIG_SND_HDA_POWER_SAVE void (*power_hook)(struct hda_codec *codec); #endif @@ -188,6 +201,8 @@ struct alc_spec { unsigned int vol_in_capsrc:1; /* use capsrc volume (ADC has no vol) */ unsigned int parse_flags; /* passed to snd_hda_parse_pin_defcfg() */ unsigned int shared_mic_hp:1; /* HP/Mic-in sharing */ + unsigned int inv_dmic_fixup:1; /* has inverted digital-mic workaround */ + unsigned int inv_dmic_muted:1; /* R-ch of inv d-mic is muted? */ /* auto-mute control */ int automute_mode; @@ -210,11 +225,6 @@ struct alc_spec { unsigned int pll_coef_idx, pll_coef_bit; unsigned int coef0; - /* fix-up list */ - int fixup_id; - const struct alc_fixup *fixup_list; - const char *fixup_name; - /* multi-io */ int multi_ios; struct alc_multi_io multi_io[4]; @@ -290,6 +300,39 @@ static inline hda_nid_t get_capsrc(struct alc_spec *spec, int idx) } static void call_update_outputs(struct hda_codec *codec); +static void alc_inv_dmic_sync(struct hda_codec *codec, bool force); + +/* for shared I/O, change the pin-control accordingly */ +static void update_shared_mic_hp(struct hda_codec *codec, bool set_as_mic) +{ + struct alc_spec *spec = codec->spec; + unsigned int val; + hda_nid_t pin = spec->autocfg.inputs[1].pin; + /* NOTE: this assumes that there are only two inputs, the + * first is the real internal mic and the second is HP/mic jack. + */ + + val = snd_hda_get_default_vref(codec, pin); + + /* This pin does not have vref caps - let's enable vref on pin 0x18 + instead, as suggested by Realtek */ + if (val == AC_PINCTL_VREF_HIZ) { + const hda_nid_t vref_pin = 0x18; + /* Sanity check pin 0x18 */ + if (get_wcaps_type(get_wcaps(codec, vref_pin)) == AC_WID_PIN && + get_defcfg_connect(snd_hda_codec_get_pincfg(codec, vref_pin)) == AC_JACK_PORT_NONE) { + unsigned int vref_val = snd_hda_get_default_vref(codec, vref_pin); + if (vref_val != AC_PINCTL_VREF_HIZ) + snd_hda_set_pin_ctl(codec, vref_pin, PIN_IN | (set_as_mic ? vref_val : 0)); + } + } + + val = set_as_mic ? val | PIN_IN : PIN_HP; + snd_hda_set_pin_ctl(codec, pin, val); + + spec->automute_speaker = !set_as_mic; + call_update_outputs(codec); +} /* select the given imux item; either unmute exclusively or select the route */ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, @@ -317,18 +360,8 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, return 0; spec->cur_mux[adc_idx] = idx; - /* for shared I/O, change the pin-control accordingly */ - if (spec->shared_mic_hp) { - /* NOTE: this assumes that there are only two inputs, the - * first is the real internal mic and the second is HP jack. - */ - snd_hda_codec_write(codec, spec->autocfg.inputs[1].pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - spec->cur_mux[adc_idx] ? - PIN_VREF80 : PIN_HP); - spec->automute_speaker = !spec->cur_mux[adc_idx]; - call_update_outputs(codec); - } + if (spec->shared_mic_hp) + update_shared_mic_hp(codec, spec->cur_mux[adc_idx]); if (spec->dyn_adc_switch) { alc_dyn_adc_pcm_resetup(codec, idx); @@ -338,7 +371,7 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, nid = get_capsrc(spec, adc_idx); /* no selection? */ - num_conns = snd_hda_get_conn_list(codec, nid, NULL); + num_conns = snd_hda_get_num_conns(codec, nid); if (num_conns <= 1) return 1; @@ -357,6 +390,7 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, AC_VERB_SET_CONNECT_SEL, imux->items[idx].index); } + alc_inv_dmic_sync(codec, true); return 1; } @@ -376,25 +410,9 @@ static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid, int auto_pin_type) { unsigned int val = PIN_IN; - - if (auto_pin_type == AUTO_PIN_MIC) { - unsigned int pincap; - unsigned int oldval; - oldval = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - pincap = snd_hda_query_pin_caps(codec, nid); - pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; - /* if the default pin setup is vref50, we give it priority */ - if ((pincap & AC_PINCAP_VREF_80) && oldval != PIN_VREF50) - val = PIN_VREF80; - else if (pincap & AC_PINCAP_VREF_50) - val = PIN_VREF50; - else if (pincap & AC_PINCAP_VREF_100) - val = PIN_VREF100; - else if (pincap & AC_PINCAP_VREF_GRD) - val = PIN_VREFGRD; - } - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val); + if (auto_pin_type == AUTO_PIN_MIC) + val |= snd_hda_get_default_vref(codec, nid); + snd_hda_set_pin_ctl(codec, nid, val); } /* @@ -409,13 +427,6 @@ static void add_mixer(struct alc_spec *spec, const struct snd_kcontrol_new *mix) spec->mixers[spec->num_mixers++] = mix; } -static void add_verb(struct alc_spec *spec, const struct hda_verb *verb) -{ - if (snd_BUG_ON(spec->num_init_verbs >= ARRAY_SIZE(spec->init_verbs))) - return; - spec->init_verbs[spec->num_init_verbs++] = verb; -} - /* * GPIO setup tables, used in initialization */ @@ -517,9 +528,7 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, } else val = 0; val |= pin_bits; - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - val); + snd_hda_set_pin_ctl(codec, nid, val); break; case ALC_AUTOMUTE_AMP: snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, @@ -678,7 +687,7 @@ static void alc_update_knob_master(struct hda_codec *codec, hda_nid_t nid) } /* unsolicited event for HP jack sensing */ -static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) +static void alc_unsol_event(struct hda_codec *codec, unsigned int res) { int action; @@ -1014,11 +1023,9 @@ static void alc_init_automute(struct hda_codec *codec) spec->automute_lo = spec->automute_lo_possible; spec->automute_speaker = spec->automute_speaker_possible; - if (spec->automute_speaker_possible || spec->automute_lo_possible) { + if (spec->automute_speaker_possible || spec->automute_lo_possible) /* create a control for automute mode */ alc_add_automute_mode_enum(codec); - spec->unsol_event = alc_sku_unsol_event; - } } /* return the position of NID in the list, or -1 if not found */ @@ -1181,7 +1188,6 @@ static void alc_init_auto_mic(struct hda_codec *codec) snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x/0x%x\n", ext, fixed, dock); - spec->unsol_event = alc_sku_unsol_event; } /* check the availabilities of auto-mute and auto-mic switches */ @@ -1200,6 +1206,16 @@ static void alc_auto_check_switches(struct hda_codec *codec) */ #define ALC_FIXUP_SKU_IGNORE (2) +static void alc_fixup_sku_ignore(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->cdefine.fixup = 1; + spec->cdefine.sku_cfg = ALC_FIXUP_SKU_IGNORE; + } +} + static int alc_auto_parse_customize_define(struct hda_codec *codec) { unsigned int ass, tmp, i; @@ -1403,178 +1419,6 @@ static void alc_ssid_check(struct hda_codec *codec, const hda_nid_t *ports) } /* - * Fix-up pin default configurations and add default verbs - */ - -struct alc_pincfg { - hda_nid_t nid; - u32 val; -}; - -struct alc_model_fixup { - const int id; - const char *name; -}; - -struct alc_fixup { - int type; - bool chained; - int chain_id; - union { - unsigned int sku; - const struct alc_pincfg *pins; - const struct hda_verb *verbs; - void (*func)(struct hda_codec *codec, - const struct alc_fixup *fix, - int action); - } v; -}; - -enum { - ALC_FIXUP_INVALID, - ALC_FIXUP_SKU, - ALC_FIXUP_PINS, - ALC_FIXUP_VERBS, - ALC_FIXUP_FUNC, -}; - -enum { - ALC_FIXUP_ACT_PRE_PROBE, - ALC_FIXUP_ACT_PROBE, - ALC_FIXUP_ACT_INIT, - ALC_FIXUP_ACT_BUILD, -}; - -static void alc_apply_pincfgs(struct hda_codec *codec, - const struct alc_pincfg *cfg) -{ - for (; cfg->nid; cfg++) - snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); -} - -static void alc_apply_fixup(struct hda_codec *codec, int action) -{ - struct alc_spec *spec = codec->spec; - int id = spec->fixup_id; -#ifdef CONFIG_SND_DEBUG_VERBOSE - const char *modelname = spec->fixup_name; -#endif - int depth = 0; - - if (!spec->fixup_list) - return; - - while (id >= 0) { - const struct alc_fixup *fix = spec->fixup_list + id; - const struct alc_pincfg *cfg; - - switch (fix->type) { - case ALC_FIXUP_SKU: - if (action != ALC_FIXUP_ACT_PRE_PROBE || !fix->v.sku) - break; - snd_printdd(KERN_INFO "hda_codec: %s: " - "Apply sku override for %s\n", - codec->chip_name, modelname); - spec->cdefine.sku_cfg = fix->v.sku; - spec->cdefine.fixup = 1; - break; - case ALC_FIXUP_PINS: - cfg = fix->v.pins; - if (action != ALC_FIXUP_ACT_PRE_PROBE || !cfg) - break; - snd_printdd(KERN_INFO "hda_codec: %s: " - "Apply pincfg for %s\n", - codec->chip_name, modelname); - alc_apply_pincfgs(codec, cfg); - break; - case ALC_FIXUP_VERBS: - if (action != ALC_FIXUP_ACT_PROBE || !fix->v.verbs) - break; - snd_printdd(KERN_INFO "hda_codec: %s: " - "Apply fix-verbs for %s\n", - codec->chip_name, modelname); - add_verb(codec->spec, fix->v.verbs); - break; - case ALC_FIXUP_FUNC: - if (!fix->v.func) - break; - snd_printdd(KERN_INFO "hda_codec: %s: " - "Apply fix-func for %s\n", - codec->chip_name, modelname); - fix->v.func(codec, fix, action); - break; - default: - snd_printk(KERN_ERR "hda_codec: %s: " - "Invalid fixup type %d\n", - codec->chip_name, fix->type); - break; - } - if (!fix->chained) - break; - if (++depth > 10) - break; - id = fix->chain_id; - } -} - -static void alc_pick_fixup(struct hda_codec *codec, - const struct alc_model_fixup *models, - const struct snd_pci_quirk *quirk, - const struct alc_fixup *fixlist) -{ - struct alc_spec *spec = codec->spec; - const struct snd_pci_quirk *q; - int id = -1; - const char *name = NULL; - - /* when model=nofixup is given, don't pick up any fixups */ - if (codec->modelname && !strcmp(codec->modelname, "nofixup")) { - spec->fixup_list = NULL; - spec->fixup_id = -1; - return; - } - - if (codec->modelname && models) { - while (models->name) { - if (!strcmp(codec->modelname, models->name)) { - id = models->id; - name = models->name; - break; - } - models++; - } - } - if (id < 0) { - q = snd_pci_quirk_lookup(codec->bus->pci, quirk); - if (q) { - id = q->value; -#ifdef CONFIG_SND_DEBUG_VERBOSE - name = q->name; -#endif - } - } - if (id < 0) { - for (q = quirk; q->subvendor; q++) { - unsigned int vendorid = - q->subdevice | (q->subvendor << 16); - if (vendorid == codec->subsystem_id) { - id = q->value; -#ifdef CONFIG_SND_DEBUG_VERBOSE - name = q->name; -#endif - break; - } - } - } - - spec->fixup_id = id; - if (id >= 0) { - spec->fixup_list = fixlist; - spec->fixup_name = name; - } -} - -/* * COEF access helper functions */ static int alc_read_coef_idx(struct hda_codec *codec, @@ -1621,8 +1465,7 @@ static void alc_auto_init_digital(struct hda_codec *codec) pin = spec->autocfg.dig_out_pins[i]; if (!pin) continue; - snd_hda_codec_write(codec, pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + snd_hda_set_pin_ctl(codec, pin, PIN_OUT); if (!i) dac = spec->multiout.dig_out_nid; else @@ -1635,9 +1478,7 @@ static void alc_auto_init_digital(struct hda_codec *codec) } pin = spec->autocfg.dig_in_pin; if (pin) - snd_hda_codec_write(codec, pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - PIN_IN); + snd_hda_set_pin_ctl(codec, pin, PIN_IN); } /* parse digital I/Os and set up NIDs in BIOS auto-parse mode */ @@ -1735,14 +1576,14 @@ typedef int (*getput_call_t)(struct snd_kcontrol *kcontrol, static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol, - getput_call_t func, bool check_adc_switch) + getput_call_t func, bool is_put) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; int i, err = 0; mutex_lock(&codec->control_mutex); - if (check_adc_switch && spec->dyn_adc_switch) { + if (is_put && spec->dyn_adc_switch) { for (i = 0; i < spec->num_adc_nids; i++) { kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[i], @@ -1763,6 +1604,8 @@ static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol, 3, 0, HDA_INPUT); err = func(kcontrol, ucontrol); } + if (err >= 0 && is_put) + alc_inv_dmic_sync(codec, false); error: mutex_unlock(&codec->control_mutex); return err; @@ -1855,6 +1698,116 @@ DEFINE_CAPMIX_NOSRC(2); DEFINE_CAPMIX_NOSRC(3); /* + * Inverted digital-mic handling + * + * First off, it's a bit tricky. The "Inverted Internal Mic Capture Switch" + * gives the additional mute only to the right channel of the digital mic + * capture stream. This is a workaround for avoiding the almost silence + * by summing the stereo stream from some (known to be ForteMedia) + * digital mic unit. + * + * The logic is to call alc_inv_dmic_sync() after each action (possibly) + * modifying ADC amp. When the mute flag is set, it mutes the R-channel + * without caching so that the cache can still keep the original value. + * The cached value is then restored when the flag is set off or any other + * than d-mic is used as the current input source. + */ +static void alc_inv_dmic_sync(struct hda_codec *codec, bool force) +{ + struct alc_spec *spec = codec->spec; + int i; + + if (!spec->inv_dmic_fixup) + return; + if (!spec->inv_dmic_muted && !force) + return; + for (i = 0; i < spec->num_adc_nids; i++) { + int src = spec->dyn_adc_switch ? 0 : i; + bool dmic_fixup = false; + hda_nid_t nid; + int parm, dir, v; + + if (spec->inv_dmic_muted && + spec->imux_pins[spec->cur_mux[src]] == spec->inv_dmic_pin) + dmic_fixup = true; + if (!dmic_fixup && !force) + continue; + if (spec->vol_in_capsrc) { + nid = spec->capsrc_nids[i]; + parm = AC_AMP_SET_RIGHT | AC_AMP_SET_OUTPUT; + dir = HDA_OUTPUT; + } else { + nid = spec->adc_nids[i]; + parm = AC_AMP_SET_RIGHT | AC_AMP_SET_INPUT; + dir = HDA_INPUT; + } + /* we care only right channel */ + v = snd_hda_codec_amp_read(codec, nid, 1, dir, 0); + if (v & 0x80) /* if already muted, we don't need to touch */ + continue; + if (dmic_fixup) /* add mute for d-mic */ + v |= 0x80; + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + parm | v); + } +} + +static int alc_inv_dmic_sw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + + ucontrol->value.integer.value[0] = !spec->inv_dmic_muted; + return 0; +} + +static int alc_inv_dmic_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + unsigned int val = !ucontrol->value.integer.value[0]; + + if (val == spec->inv_dmic_muted) + return 0; + spec->inv_dmic_muted = val; + alc_inv_dmic_sync(codec, true); + return 0; +} + +static const struct snd_kcontrol_new alc_inv_dmic_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = snd_ctl_boolean_mono_info, + .get = alc_inv_dmic_sw_get, + .put = alc_inv_dmic_sw_put, +}; + +static int alc_add_inv_dmic_mixer(struct hda_codec *codec, hda_nid_t nid) +{ + struct alc_spec *spec = codec->spec; + struct snd_kcontrol_new *knew = alc_kcontrol_new(spec); + if (!knew) + return -ENOMEM; + *knew = alc_inv_dmic_sw; + knew->name = kstrdup("Inverted Internal Mic Capture Switch", GFP_KERNEL); + if (!knew->name) + return -ENOMEM; + spec->inv_dmic_fixup = 1; + spec->inv_dmic_muted = 0; + spec->inv_dmic_pin = nid; + return 0; +} + +/* typically the digital mic is put at node 0x12 */ +static void alc_fixup_inv_dmic_0x12(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + if (action == ALC_FIXUP_ACT_PROBE) + alc_add_inv_dmic_mixer(codec, 0x12); +} + +/* * virtual master controls */ @@ -2044,13 +1997,31 @@ static int __alc_build_controls(struct hda_codec *codec) return 0; } -static int alc_build_controls(struct hda_codec *codec) +static int alc_build_jacks(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + + if (spec->shared_mic_hp) { + int err; + int nid = spec->autocfg.inputs[1].pin; + err = snd_hda_jack_add_kctl(codec, nid, "Headphone Mic", 0); + if (err < 0) + return err; + err = snd_hda_jack_detect_enable(codec, nid, 0); + if (err < 0) + return err; + } + + return snd_hda_jack_add_kctls(codec, &spec->autocfg); +} + +static int alc_build_controls(struct hda_codec *codec) +{ int err = __alc_build_controls(codec); if (err < 0) return err; - err = snd_hda_jack_add_kctls(codec, &spec->autocfg); + + err = alc_build_jacks(codec); if (err < 0) return err; alc_apply_fixup(codec, ALC_FIXUP_ACT_BUILD); @@ -2068,7 +2039,6 @@ static void alc_auto_init_std(struct hda_codec *codec); static int alc_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int i; if (spec->init_hook) spec->init_hook(codec); @@ -2076,8 +2046,7 @@ static int alc_init(struct hda_codec *codec) alc_fix_pll(codec); alc_auto_init_amp(codec, spec->init_amp); - for (i = 0; i < spec->num_init_verbs; i++) - snd_hda_sequence_write(codec, spec->init_verbs[i]); + snd_hda_gen_apply_verbs(codec); alc_init_special_input_src(codec); alc_auto_init_std(codec); @@ -2089,14 +2058,6 @@ static int alc_init(struct hda_codec *codec) return 0; } -static void alc_unsol_event(struct hda_codec *codec, unsigned int res) -{ - struct alc_spec *spec = codec->spec; - - if (spec->unsol_event) - spec->unsol_event(codec, res); -} - #ifdef CONFIG_SND_HDA_POWER_SAVE static int alc_check_power_status(struct hda_codec *codec, hda_nid_t nid) { @@ -2470,6 +2431,7 @@ static void alc_free(struct hda_codec *codec) alc_shutup(codec); alc_free_kctls(codec); alc_free_bind_ctls(codec); + snd_hda_gen_free(&spec->gen); kfree(spec); snd_hda_detach_beep_device(codec); } @@ -2480,7 +2442,7 @@ static void alc_power_eapd(struct hda_codec *codec) alc_auto_setup_eapd(codec, false); } -static int alc_suspend(struct hda_codec *codec, pm_message_t state) +static int alc_suspend(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; alc_shutup(codec); @@ -2497,6 +2459,7 @@ static int alc_resume(struct hda_codec *codec) codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); + alc_inv_dmic_sync(codec, true); hda_call_check_power_status(codec, 0x01); return 0; } @@ -2550,6 +2513,7 @@ static struct alc_codec_rename_table rename_tbl[] = { { 0x10ec0269, 0xffff, 0xa023, "ALC259" }, { 0x10ec0269, 0xffff, 0x6023, "ALC281X" }, { 0x10ec0269, 0x00f0, 0x0020, "ALC269VC" }, + { 0x10ec0269, 0x00f0, 0x0030, "ALC269VD" }, { 0x10ec0887, 0x00f0, 0x0030, "ALC887-VD" }, { 0x10ec0888, 0x00f0, 0x0030, "ALC888-VD" }, { 0x10ec0888, 0xf0f0, 0x3020, "ALC886" }, @@ -2725,7 +2689,6 @@ static int alc_auto_fill_adc_caps(struct hda_codec *codec) nid = codec->start_nid; for (i = 0; i < codec->num_nodes; i++, nid++) { hda_nid_t src; - const hda_nid_t *list; unsigned int caps = get_wcaps(codec, nid); int type = get_wcaps_type(caps); @@ -2743,13 +2706,14 @@ static int alc_auto_fill_adc_caps(struct hda_codec *codec) cap_nids[nums] = src; break; } - n = snd_hda_get_conn_list(codec, src, &list); + n = snd_hda_get_num_conns(codec, src); if (n > 1) { cap_nids[nums] = src; break; } else if (n != 1) break; - src = *list; + if (snd_hda_get_connections(codec, src, &src, 1) != 1) + break; } if (++nums >= max_nums) break; @@ -2856,8 +2820,7 @@ static int alc_auto_create_shared_input(struct hda_codec *codec) static void alc_set_pin_output(struct hda_codec *codec, hda_nid_t nid, unsigned int pin_type) { - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_type); + snd_hda_set_pin_ctl(codec, nid, pin_type); /* unmute pin */ if (nid_has_mute(codec, nid, HDA_OUTPUT)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -2891,7 +2854,7 @@ static void alc_auto_init_analog_input(struct hda_codec *codec) /* mute all loopback inputs */ if (spec->mixer_nid) { - int nums = snd_hda_get_conn_list(codec, spec->mixer_nid, NULL); + int nums = snd_hda_get_num_conns(codec, spec->mixer_nid); for (i = 0; i < nums; i++) snd_hda_codec_write(codec, spec->mixer_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -3521,7 +3484,7 @@ static int alc_auto_add_sw_ctl(struct hda_codec *codec, if (wid_type == AC_WID_PIN || wid_type == AC_WID_AUD_OUT) { type = ALC_CTL_WIDGET_MUTE; val = HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT); - } else if (snd_hda_get_conn_list(codec, nid, NULL) == 1) { + } else if (snd_hda_get_num_conns(codec, nid) == 1) { type = ALC_CTL_WIDGET_MUTE; val = HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_INPUT); } else { @@ -3998,9 +3961,7 @@ static int alc_set_multi_io(struct hda_codec *codec, int idx, bool output) snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); if (output) { - snd_hda_codec_update_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - PIN_OUT); + snd_hda_set_pin_ctl_cache(codec, nid, PIN_OUT); if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, HDA_AMP_MUTE, 0); @@ -4009,9 +3970,8 @@ static int alc_set_multi_io(struct hda_codec *codec, int idx, bool output) if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, HDA_AMP_MUTE, HDA_AMP_MUTE); - snd_hda_codec_update_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - spec->multi_io[idx].ctl_in); + snd_hda_set_pin_ctl_cache(codec, nid, + spec->multi_io[idx].ctl_in); } return 0; } @@ -4084,7 +4044,7 @@ static void alc_remove_invalid_adc_nids(struct hda_codec *codec) nums = 0; for (n = 0; n < spec->num_adc_nids; n++) { hda_nid_t cap = spec->private_capsrc_nids[n]; - int num_conns = snd_hda_get_conn_list(codec, cap, NULL); + int num_conns = snd_hda_get_num_conns(codec, cap); for (i = 0; i < imux->num_items; i++) { hda_nid_t pin = spec->imux_pins[i]; if (pin) { @@ -4213,7 +4173,7 @@ static void select_or_unmute_capsrc(struct hda_codec *codec, hda_nid_t cap, if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) { snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx, HDA_AMP_MUTE, 0); - } else if (snd_hda_get_conn_list(codec, cap, NULL) > 1) { + } else if (snd_hda_get_num_conns(codec, cap) > 1) { snd_hda_codec_write_cache(codec, cap, 0, AC_VERB_SET_CONNECT_SEL, idx); } @@ -4299,14 +4259,12 @@ static void set_capture_mixer(struct hda_codec *codec) */ static void alc_auto_init_std(struct hda_codec *codec) { - struct alc_spec *spec = codec->spec; alc_auto_init_multi_out(codec); alc_auto_init_extra_out(codec); alc_auto_init_analog_input(codec); alc_auto_init_input_src(codec); alc_auto_init_digital(codec); - if (spec->unsol_event) - alc_inithook(codec); + alc_inithook(codec); } /* @@ -4427,6 +4385,26 @@ static int alc_parse_auto_config(struct hda_codec *codec, return 1; } +/* common preparation job for alc_spec */ +static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid) +{ + struct alc_spec *spec = kzalloc(sizeof(*spec), GFP_KERNEL); + int err; + + if (!spec) + return -ENOMEM; + codec->spec = spec; + spec->mixer_nid = mixer_nid; + snd_hda_gen_init(&spec->gen); + + err = alc_codec_rename_from_preset(codec); + if (err < 0) { + kfree(spec); + return err; + } + return 0; +} + static int alc880_parse_auto_config(struct hda_codec *codec) { static const hda_nid_t alc880_ignore[] = { 0x1d, 0 }; @@ -4808,13 +4786,11 @@ static int patch_alc880(struct hda_codec *codec) struct alc_spec *spec; int err; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; + err = alc_alloc_spec(codec, 0x0b); + if (err < 0) + return err; - spec->mixer_nid = 0x0b; + spec = codec->spec; spec->need_dac_fix = 1; alc_pick_fixup(codec, alc880_fixup_models, alc880_fixup_tbl, @@ -4889,8 +4865,7 @@ static void alc260_fixup_gpio1_toggle(struct hda_codec *codec, spec->automute_speaker = 1; spec->autocfg.hp_pins[0] = 0x0f; /* copy it for automute */ snd_hda_jack_detect_enable(codec, 0x0f, ALC_HP_EVENT); - spec->unsol_event = alc_sku_unsol_event; - add_verb(codec->spec, alc_gpio1_init_verbs); + snd_hda_gen_add_verbs(&spec->gen, alc_gpio1_init_verbs); } } @@ -5001,13 +4976,11 @@ static int patch_alc260(struct hda_codec *codec) struct alc_spec *spec; int err; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; + err = alc_alloc_spec(codec, 0x07); + if (err < 0) + return err; - spec->mixer_nid = 0x07; + spec = codec->spec; alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); @@ -5076,6 +5049,7 @@ enum { ALC889_FIXUP_DAC_ROUTE, ALC889_FIXUP_MBP_VREF, ALC889_FIXUP_IMAC91_VREF, + ALC882_FIXUP_INV_DMIC, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -5171,8 +5145,7 @@ static void alc889_fixup_mbp_vref(struct hda_codec *codec, val = snd_hda_codec_read(codec, nids[i], 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); val |= AC_PINCTL_VREF_80; - snd_hda_codec_write(codec, nids[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, val); + snd_hda_set_pin_ctl(codec, nids[i], val); spec->keep_vref_in_automute = 1; break; } @@ -5193,8 +5166,7 @@ static void alc889_fixup_imac91_vref(struct hda_codec *codec, val = snd_hda_codec_read(codec, nids[i], 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); val |= AC_PINCTL_VREF_50; - snd_hda_codec_write(codec, nids[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, val); + snd_hda_set_pin_ctl(codec, nids[i], val); } spec->keep_vref_in_automute = 1; } @@ -5225,8 +5197,8 @@ static const struct alc_fixup alc882_fixups[] = { } }, [ALC882_FIXUP_ACER_ASPIRE_7736] = { - .type = ALC_FIXUP_SKU, - .v.sku = ALC_FIXUP_SKU_IGNORE, + .type = ALC_FIXUP_FUNC, + .v.func = alc_fixup_sku_ignore, }, [ALC882_FIXUP_ASUS_W90V] = { .type = ALC_FIXUP_PINS, @@ -5381,6 +5353,10 @@ static const struct alc_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC882_FIXUP_GPIO1, }, + [ALC882_FIXUP_INV_DMIC] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc_fixup_inv_dmic_0x12, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -5455,6 +5431,7 @@ static const struct alc_model_fixup alc882_fixup_models[] = { {.id = ALC882_FIXUP_ACER_ASPIRE_4930G, .name = "acer-aspire-4930g"}, {.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"}, {.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"}, + {.id = ALC882_FIXUP_INV_DMIC, .name = "inv-dmic"}, {} }; @@ -5476,13 +5453,11 @@ static int patch_alc882(struct hda_codec *codec) struct alc_spec *spec; int err; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; + err = alc_alloc_spec(codec, 0x0b); + if (err < 0) + return err; - spec->mixer_nid = 0x0b; + spec = codec->spec; switch (codec->vendor_id) { case 0x10ec0882: @@ -5494,10 +5469,6 @@ static int patch_alc882(struct hda_codec *codec) break; } - err = alc_codec_rename_from_preset(codec); - if (err < 0) - goto error; - alc_pick_fixup(codec, alc882_fixup_models, alc882_fixup_tbl, alc882_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); @@ -5548,6 +5519,7 @@ enum { ALC262_FIXUP_LENOVO_3000, ALC262_FIXUP_BENQ, ALC262_FIXUP_BENQ_T31, + ALC262_FIXUP_INV_DMIC, }; static const struct alc_fixup alc262_fixups[] = { @@ -5599,6 +5571,10 @@ static const struct alc_fixup alc262_fixups[] = { {} } }, + [ALC262_FIXUP_INV_DMIC] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc_fixup_inv_dmic_0x12, + }, }; static const struct snd_pci_quirk alc262_fixup_tbl[] = { @@ -5613,6 +5589,10 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = { {} }; +static const struct alc_model_fixup alc262_fixup_models[] = { + {.id = ALC262_FIXUP_INV_DMIC, .name = "inv-dmic"}, + {} +}; /* */ @@ -5621,13 +5601,11 @@ static int patch_alc262(struct hda_codec *codec) struct alc_spec *spec; int err; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; + err = alc_alloc_spec(codec, 0x0b); + if (err < 0) + return err; - spec->mixer_nid = 0x0b; + spec = codec->spec; #if 0 /* pshou 07/11/05 set a zero PCM sample to DAC when FIFO is @@ -5643,7 +5621,8 @@ static int patch_alc262(struct hda_codec *codec) #endif alc_fix_pll_init(codec, 0x20, 0x0a, 10); - alc_pick_fixup(codec, NULL, alc262_fixup_tbl, alc262_fixups); + alc_pick_fixup(codec, alc262_fixup_models, alc262_fixup_tbl, + alc262_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); alc_auto_parse_customize_define(codec); @@ -5699,6 +5678,22 @@ static const struct hda_verb alc268_beep_init_verbs[] = { { } }; +enum { + ALC268_FIXUP_INV_DMIC, +}; + +static const struct alc_fixup alc268_fixups[] = { + [ALC268_FIXUP_INV_DMIC] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc_fixup_inv_dmic_0x12, + }, +}; + +static const struct alc_model_fixup alc268_fixup_models[] = { + {.id = ALC268_FIXUP_INV_DMIC, .name = "inv-dmic"}, + {} +}; + /* * BIOS auto configuration */ @@ -5710,7 +5705,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) if (err > 0) { if (!spec->no_analog && spec->autocfg.speaker_pins[0] != 0x1d) { add_mixer(spec, alc268_beep_mixer); - add_verb(spec, alc268_beep_init_verbs); + snd_hda_gen_add_verbs(&spec->gen, alc268_beep_init_verbs); } } return err; @@ -5723,13 +5718,15 @@ static int patch_alc268(struct hda_codec *codec) struct alc_spec *spec; int i, has_beep, err; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; + /* ALC268 has no aa-loopback mixer */ + err = alc_alloc_spec(codec, 0); + if (err < 0) + return err; - codec->spec = spec; + spec = codec->spec; - /* ALC268 has no aa-loopback mixer */ + alc_pick_fixup(codec, alc268_fixup_models, NULL, alc268_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); /* automatic parse from the BIOS config */ err = alc268_parse_auto_config(codec); @@ -5760,6 +5757,8 @@ static int patch_alc268(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; spec->shutup = alc_eapd_shutup; + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + return 0; error: @@ -5796,6 +5795,7 @@ enum { ALC269_TYPE_ALC269VA, ALC269_TYPE_ALC269VB, ALC269_TYPE_ALC269VC, + ALC269_TYPE_ALC269VD, }; /* @@ -5807,8 +5807,21 @@ static int alc269_parse_auto_config(struct hda_codec *codec) static const hda_nid_t alc269_ssids[] = { 0, 0x1b, 0x14, 0x21 }; static const hda_nid_t alc269va_ssids[] = { 0x15, 0x1b, 0x14, 0 }; struct alc_spec *spec = codec->spec; - const hda_nid_t *ssids = spec->codec_variant == ALC269_TYPE_ALC269VA ? - alc269va_ssids : alc269_ssids; + const hda_nid_t *ssids; + + switch (spec->codec_variant) { + case ALC269_TYPE_ALC269VA: + case ALC269_TYPE_ALC269VC: + ssids = alc269va_ssids; + break; + case ALC269_TYPE_ALC269VB: + case ALC269_TYPE_ALC269VD: + ssids = alc269_ssids; + break; + default: + ssids = alc269_ssids; + break; + } return alc_parse_auto_config(codec, alc269_ignore, ssids); } @@ -5825,6 +5838,11 @@ static void alc269_toggle_power_output(struct hda_codec *codec, int power_up) static void alc269_shutup(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; + + if (spec->codec_variant != ALC269_TYPE_ALC269VB) + return; + if ((alc_get_coef0(codec) & 0x00ff) == 0x017) alc269_toggle_power_output(codec, 0); if ((alc_get_coef0(codec) & 0x00ff) == 0x018) { @@ -5836,19 +5854,24 @@ static void alc269_shutup(struct hda_codec *codec) #ifdef CONFIG_PM static int alc269_resume(struct hda_codec *codec) { - if ((alc_get_coef0(codec) & 0x00ff) == 0x018) { + struct alc_spec *spec = codec->spec; + + if (spec->codec_variant == ALC269_TYPE_ALC269VB || + (alc_get_coef0(codec) & 0x00ff) == 0x018) { alc269_toggle_power_output(codec, 0); msleep(150); } codec->patch_ops.init(codec); - if ((alc_get_coef0(codec) & 0x00ff) == 0x017) { + if (spec->codec_variant == ALC269_TYPE_ALC269VB || + (alc_get_coef0(codec) & 0x00ff) == 0x017) { alc269_toggle_power_output(codec, 1); msleep(200); } - if ((alc_get_coef0(codec) & 0x00ff) == 0x018) + if (spec->codec_variant == ALC269_TYPE_ALC269VB || + (alc_get_coef0(codec) & 0x00ff) == 0x018) alc269_toggle_power_output(codec, 1); snd_hda_codec_resume_amp(codec); @@ -5858,6 +5881,15 @@ static int alc269_resume(struct hda_codec *codec) } #endif /* CONFIG_PM */ +static void alc269_fixup_pincfg_no_hp_to_lineout(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == ALC_FIXUP_ACT_PRE_PROBE) + spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; +} + static void alc269_fixup_hweq(struct hda_codec *codec, const struct alc_fixup *fix, int action) { @@ -5946,9 +5978,7 @@ static void alc269_fixup_mic2_mute_hook(void *private_data, int enabled) { struct hda_codec *codec = private_data; unsigned int pinval = enabled ? 0x20 : 0x24; - snd_hda_codec_update_cache(codec, 0x19, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pinval); + snd_hda_set_pin_ctl_cache(codec, 0x19, pinval); } static void alc269_fixup_mic2_mute(struct hda_codec *codec, @@ -5966,6 +5996,7 @@ static void alc269_fixup_mic2_mute(struct hda_codec *codec, } } + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -5984,6 +6015,9 @@ enum { ALC269VB_FIXUP_AMIC, ALC269VB_FIXUP_DMIC, ALC269_FIXUP_MIC2_MUTE_LED, + ALC269_FIXUP_INV_DMIC, + ALC269_FIXUP_LENOVO_DOCK, + ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, }; static const struct alc_fixup alc269_fixups[] = { @@ -6015,8 +6049,8 @@ static const struct alc_fixup alc269_fixups[] = { } }, [ALC269_FIXUP_SKU_IGNORE] = { - .type = ALC_FIXUP_SKU, - .v.sku = ALC_FIXUP_SKU_IGNORE, + .type = ALC_FIXUP_FUNC, + .v.func = alc_fixup_sku_ignore, }, [ALC269_FIXUP_ASUS_G73JW] = { .type = ALC_FIXUP_PINS, @@ -6108,12 +6142,33 @@ static const struct alc_fixup alc269_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_mic2_mute, }, + [ALC269_FIXUP_INV_DMIC] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc_fixup_inv_dmic_0x12, + }, + [ALC269_FIXUP_LENOVO_DOCK] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x19, 0x23a11040 }, /* dock mic */ + { 0x1b, 0x2121103f }, /* dock headphone */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT + }, + [ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_pincfg_no_hp_to_lineout, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), + SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), @@ -6131,6 +6186,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), @@ -6189,6 +6245,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { static const struct alc_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_AMIC, .name = "laptop-amic"}, {.id = ALC269_FIXUP_DMIC, .name = "laptop-dmic"}, + {.id = ALC269_FIXUP_STEREO_DMIC, .name = "alc269-dmic"}, + {.id = ALC271_FIXUP_DMIC, .name = "alc271-dmic"}, + {.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"}, + {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"}, {} }; @@ -6242,19 +6302,13 @@ static void alc269_fill_coef(struct hda_codec *codec) static int patch_alc269(struct hda_codec *codec) { struct alc_spec *spec; - int err = 0; - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; - - spec->mixer_nid = 0x0b; + int err; - err = alc_codec_rename_from_preset(codec); + err = alc_alloc_spec(codec, 0x0b); if (err < 0) - goto error; + return err; + + spec = codec->spec; if (codec->vendor_id == 0x10ec0269) { spec->codec_variant = ALC269_TYPE_ALC269VA; @@ -6271,6 +6325,9 @@ static int patch_alc269(struct hda_codec *codec) err = alc_codec_rename(codec, "ALC3202"); spec->codec_variant = ALC269_TYPE_ALC269VC; break; + case 0x0030: + spec->codec_variant = ALC269_TYPE_ALC269VD; + break; default: alc_fix_pll_init(codec, 0x20, 0x04, 15); } @@ -6346,8 +6403,7 @@ static void alc861_fixup_asus_amp_vref_0f(struct hda_codec *codec, if (!(val & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN))) val |= AC_PINCTL_IN_EN; val |= AC_PINCTL_VREF_50; - snd_hda_codec_write(codec, 0x0f, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, val); + snd_hda_set_pin_ctl(codec, 0x0f, val); spec->keep_vref_in_automute = 1; } @@ -6401,13 +6457,11 @@ static int patch_alc861(struct hda_codec *codec) struct alc_spec *spec; int err; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; + err = alc_alloc_spec(codec, 0x15); + if (err < 0) + return err; - spec->mixer_nid = 0x15; + spec = codec->spec; alc_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); @@ -6491,12 +6545,6 @@ static const struct snd_pci_quirk alc861vd_fixup_tbl[] = { {} }; -static const struct hda_verb alc660vd_eapd_verbs[] = { - {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, - {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, - { } -}; - /* */ static int patch_alc861vd(struct hda_codec *codec) @@ -6504,13 +6552,11 @@ static int patch_alc861vd(struct hda_codec *codec) struct alc_spec *spec; int err; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; + err = alc_alloc_spec(codec, 0x0b); + if (err < 0) + return err; - spec->mixer_nid = 0x0b; + spec = codec->spec; alc_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); @@ -6520,11 +6566,6 @@ static int patch_alc861vd(struct hda_codec *codec) if (err < 0) goto error; - if (codec->vendor_id == 0x10ec0660) { - /* always turn on EAPD */ - add_verb(spec, alc660vd_eapd_verbs); - } - if (!spec->no_analog) { err = snd_hda_attach_beep_device(codec, 0x23); if (err < 0) @@ -6606,6 +6647,8 @@ enum { ALC662_FIXUP_ASUS_MODE7, ALC662_FIXUP_ASUS_MODE8, ALC662_FIXUP_NO_JACK_DETECT, + ALC662_FIXUP_ZOTAC_Z68, + ALC662_FIXUP_INV_DMIC, }; static const struct alc_fixup alc662_fixups[] = { @@ -6635,8 +6678,8 @@ static const struct alc_fixup alc662_fixups[] = { } }, [ALC662_FIXUP_SKU_IGNORE] = { - .type = ALC_FIXUP_SKU, - .v.sku = ALC_FIXUP_SKU_IGNORE, + .type = ALC_FIXUP_FUNC, + .v.func = alc_fixup_sku_ignore, }, [ALC662_FIXUP_HP_RP5800] = { .type = ALC_FIXUP_PINS, @@ -6755,12 +6798,24 @@ static const struct alc_fixup alc662_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc_fixup_no_jack_detect, }, + [ALC662_FIXUP_ZOTAC_Z68] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1b, 0x02214020 }, /* Front HP */ + { } + } + }, + [ALC662_FIXUP_INV_DMIC] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc_fixup_inv_dmic_0x12, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), @@ -6768,6 +6823,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), + SND_PCI_QUIRK(0x19da, 0xa130, "Zotac Z68", ALC662_FIXUP_ZOTAC_Z68), SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T), #if 0 @@ -6840,33 +6896,62 @@ static const struct alc_model_fixup alc662_fixup_models[] = { {.id = ALC662_FIXUP_ASUS_MODE6, .name = "asus-mode6"}, {.id = ALC662_FIXUP_ASUS_MODE7, .name = "asus-mode7"}, {.id = ALC662_FIXUP_ASUS_MODE8, .name = "asus-mode8"}, + {.id = ALC662_FIXUP_INV_DMIC, .name = "inv-dmic"}, {} }; +static void alc662_fill_coef(struct hda_codec *codec) +{ + int val, coef; + + coef = alc_get_coef0(codec); + + switch (codec->vendor_id) { + case 0x10ec0662: + if ((coef & 0x00f0) == 0x0030) { + val = alc_read_coef_idx(codec, 0x4); /* EAPD Ctrl */ + alc_write_coef_idx(codec, 0x4, val & ~(1<<10)); + } + break; + case 0x10ec0272: + case 0x10ec0273: + case 0x10ec0663: + case 0x10ec0665: + case 0x10ec0670: + case 0x10ec0671: + case 0x10ec0672: + val = alc_read_coef_idx(codec, 0xd); /* EAPD Ctrl */ + alc_write_coef_idx(codec, 0xd, val | (1<<14)); + break; + } +} /* */ static int patch_alc662(struct hda_codec *codec) { struct alc_spec *spec; - int err = 0; - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (!spec) - return -ENOMEM; + int err; - codec->spec = spec; + err = alc_alloc_spec(codec, 0x0b); + if (err < 0) + return err; - spec->mixer_nid = 0x0b; + spec = codec->spec; /* handle multiple HPs as is */ spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; alc_fix_pll_init(codec, 0x20, 0x04, 15); - err = alc_codec_rename_from_preset(codec); - if (err < 0) - goto error; + spec->init_hook = alc662_fill_coef; + alc662_fill_coef(codec); + + alc_pick_fixup(codec, alc662_fixup_models, + alc662_fixup_tbl, alc662_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + + alc_auto_parse_customize_define(codec); if ((alc_get_coef0(codec) & (1 << 14)) && codec->bus->pci->subsystem_vendor == 0x1025 && @@ -6875,12 +6960,6 @@ static int patch_alc662(struct hda_codec *codec) goto error; } - alc_pick_fixup(codec, alc662_fixup_models, - alc662_fixup_tbl, alc662_fixups); - alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - - alc_auto_parse_customize_define(codec); - /* automatic parse from the BIOS config */ err = alc662_parse_auto_config(codec); if (err < 0) @@ -6930,16 +7009,12 @@ static int alc680_parse_auto_config(struct hda_codec *codec) */ static int patch_alc680(struct hda_codec *codec) { - struct alc_spec *spec; int err; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; - /* ALC680 has no aa-loopback mixer */ + err = alc_alloc_spec(codec, 0); + if (err < 0) + return err; /* automatic parse from the BIOS config */ err = alc680_parse_auto_config(codec); @@ -6967,6 +7042,8 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 }, { .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 }, { .id = 0x10ec0276, .name = "ALC276", .patch = patch_alc269 }, + { .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 }, + { .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", .patch = patch_alc861 }, { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd }, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 4742cac26aa9..a1596a3b171c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -36,6 +36,7 @@ #include <sound/tlv.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_beep.h" #include "hda_jack.h" @@ -221,6 +222,7 @@ struct sigmatel_spec { unsigned char aloopback_shift; /* power management */ + unsigned int power_map_bits; unsigned int num_pwrs; const hda_nid_t *pwr_nids; const hda_nid_t *dac_list; @@ -314,6 +316,9 @@ struct sigmatel_spec { struct hda_vmaster_mute_hook vmaster_mute; }; +#define AC_VERB_IDT_SET_POWER_MAP 0x7ec +#define AC_VERB_IDT_GET_POWER_MAP 0xfec + static const hda_nid_t stac9200_adc_nids[1] = { 0x03, }; @@ -681,8 +686,7 @@ static int stac_vrefout_set(struct hda_codec *codec, pinctl &= ~AC_PINCTL_VREFEN; pinctl |= (new_vref & AC_PINCTL_VREFEN); - error = snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); + error = snd_hda_set_pin_ctl_cache(codec, nid, pinctl); if (error < 0) return error; @@ -706,8 +710,7 @@ static unsigned int stac92xx_vref_set(struct hda_codec *codec, else pincfg |= AC_PINCTL_IN_EN; - error = snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, pincfg); + error = snd_hda_set_pin_ctl_cache(codec, nid, pincfg); if (error < 0) return error; else @@ -2505,27 +2508,10 @@ static int stac92xx_build_pcms(struct hda_codec *codec) return 0; } -static unsigned int stac92xx_get_default_vref(struct hda_codec *codec, - hda_nid_t nid) -{ - unsigned int pincap = snd_hda_query_pin_caps(codec, nid); - pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; - if (pincap & AC_PINCAP_VREF_100) - return AC_PINCTL_VREF_100; - if (pincap & AC_PINCAP_VREF_80) - return AC_PINCTL_VREF_80; - if (pincap & AC_PINCAP_VREF_50) - return AC_PINCTL_VREF_50; - if (pincap & AC_PINCAP_VREF_GRD) - return AC_PINCTL_VREF_GRD; - return 0; -} - static void stac92xx_auto_set_pinctl(struct hda_codec *codec, hda_nid_t nid, int pin_type) { - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); + snd_hda_set_pin_ctl_cache(codec, nid, pin_type); } #define stac92xx_hp_switch_info snd_ctl_boolean_mono_info @@ -2594,7 +2580,7 @@ static int stac92xx_dc_bias_get(struct snd_kcontrol *kcontrol, hda_nid_t nid = kcontrol->private_value; unsigned int vref = stac92xx_vref_get(codec, nid); - if (vref == stac92xx_get_default_vref(codec, nid)) + if (vref == snd_hda_get_default_vref(codec, nid)) ucontrol->value.enumerated.item[0] = 0; else if (vref == AC_PINCTL_VREF_GRD) ucontrol->value.enumerated.item[0] = 1; @@ -2613,7 +2599,7 @@ static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol, hda_nid_t nid = kcontrol->private_value; if (ucontrol->value.enumerated.item[0] == 0) - new_vref = stac92xx_get_default_vref(codec, nid); + new_vref = snd_hda_get_default_vref(codec, nid); else if (ucontrol->value.enumerated.item[0] == 1) new_vref = AC_PINCTL_VREF_GRD; else if (ucontrol->value.enumerated.item[0] == 2) @@ -2679,7 +2665,7 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ else { unsigned int pinctl = AC_PINCTL_IN_EN; if (io_idx) /* set VREF for mic */ - pinctl |= stac92xx_get_default_vref(codec, nid); + pinctl |= snd_hda_get_default_vref(codec, nid); stac92xx_auto_set_pinctl(codec, nid, pinctl); } @@ -2847,7 +2833,7 @@ static inline int stac92xx_add_jack_mode_control(struct hda_codec *codec, char name[22]; if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT) { - if (stac92xx_get_default_vref(codec, nid) == AC_PINCTL_VREF_GRD + if (snd_hda_get_default_vref(codec, nid) == AC_PINCTL_VREF_GRD && nid == spec->line_switch) control = STAC_CTL_WIDGET_IO_SWITCH; else if (snd_hda_query_pin_caps(codec, nid) @@ -4250,13 +4236,6 @@ static void stac_store_hints(struct hda_codec *codec) val = snd_hda_get_bool_hint(codec, "eapd_switch"); if (val >= 0) spec->eapd_switch = val; - get_int_hint(codec, "gpio_led_polarity", &spec->gpio_led_polarity); - if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) { - spec->gpio_mask |= spec->gpio_led; - spec->gpio_dir |= spec->gpio_led; - if (spec->gpio_led_polarity) - spec->gpio_data |= spec->gpio_led; - } } static void stac_issue_unsol_events(struct hda_codec *codec, int num_pins, @@ -4354,7 +4333,7 @@ static int stac92xx_init(struct hda_codec *codec) unsigned int pinctl, conf; if (type == AUTO_PIN_MIC) { /* for mic pins, force to initialize */ - pinctl = stac92xx_get_default_vref(codec, nid); + pinctl = snd_hda_get_default_vref(codec, nid); pinctl |= AC_PINCTL_IN_EN; stac92xx_auto_set_pinctl(codec, nid, pinctl); } else { @@ -4388,12 +4367,25 @@ static int stac92xx_init(struct hda_codec *codec) AC_PINCTL_IN_EN); for (i = 0; i < spec->num_pwrs; i++) { hda_nid_t nid = spec->pwr_nids[i]; - int pinctl, def_conf; + unsigned int pinctl, def_conf; + def_conf = snd_hda_codec_get_pincfg(codec, nid); + def_conf = get_defcfg_connect(def_conf); + if (def_conf == AC_JACK_PORT_NONE) { + /* power off unused ports */ + stac_toggle_power_map(codec, nid, 0); + continue; + } + if (def_conf == AC_JACK_PORT_FIXED) { + /* no need for jack detection for fixed pins */ + stac_toggle_power_map(codec, nid, 1); + continue; + } /* power on when no jack detection is available */ /* or when the VREF is used for controlling LED */ if (!spec->hp_detect || - spec->vref_mute_led_nid == nid) { + spec->vref_mute_led_nid == nid || + !is_jack_detectable(codec, nid)) { stac_toggle_power_map(codec, nid, 1); continue; } @@ -4411,15 +4403,6 @@ static int stac92xx_init(struct hda_codec *codec) stac_toggle_power_map(codec, nid, 1); continue; } - def_conf = snd_hda_codec_get_pincfg(codec, nid); - def_conf = get_defcfg_connect(def_conf); - /* skip any ports that don't have jacks since presence - * detection is useless */ - if (def_conf != AC_JACK_PORT_COMPLEX) { - if (def_conf != AC_JACK_PORT_NONE) - stac_toggle_power_map(codec, nid, 1); - continue; - } if (enable_pin_detect(codec, nid, STAC_PWR_EVENT)) { stac_issue_unsol_event(codec, nid); continue; @@ -4432,6 +4415,12 @@ static int stac92xx_init(struct hda_codec *codec) /* sync mute LED */ snd_hda_sync_vmaster_hook(&spec->vmaster_mute); + + /* sync the power-map */ + if (spec->num_pwrs) + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_IDT_SET_POWER_MAP, + spec->power_map_bits); if (spec->dac_list) stac92xx_power_down(codec); return 0; @@ -4460,8 +4449,7 @@ static void stac92xx_shutup_pins(struct hda_codec *codec) struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); def_conf = snd_hda_codec_get_pincfg(codec, pin->nid); if (get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE) - snd_hda_codec_write(codec, pin->nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, pin->nid, 0); } } @@ -4517,9 +4505,7 @@ static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, pin_ctl |= flag; if (old_ctl != pin_ctl) - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_ctl); + snd_hda_set_pin_ctl_cache(codec, nid, pin_ctl); } static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, @@ -4528,9 +4514,7 @@ static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, unsigned int pin_ctl = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00); if (pin_ctl & flag) - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_ctl & ~flag); + snd_hda_set_pin_ctl_cache(codec, nid, pin_ctl & ~flag); } static inline int get_pin_presence(struct hda_codec *codec, hda_nid_t nid) @@ -4682,14 +4666,18 @@ static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid, idx = 1 << idx; - val = snd_hda_codec_read(codec, codec->afg, 0, 0x0fec, 0x0) & 0xff; + val = spec->power_map_bits; if (enable) val &= ~idx; else val |= idx; /* power down unused output ports */ - snd_hda_codec_write(codec, codec->afg, 0, 0x7ec, val); + if (val != spec->power_map_bits) { + spec->power_map_bits = val; + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_IDT_SET_POWER_MAP, val); + } } static void stac92xx_pin_sense(struct hda_codec *codec, hda_nid_t nid) @@ -4866,6 +4854,11 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) struct sigmatel_spec *spec = codec->spec; const struct dmi_device *dev = NULL; + if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) { + get_int_hint(codec, "gpio_led_polarity", + &spec->gpio_led_polarity); + return 1; + } if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) { while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, NULL, dev))) { @@ -4952,7 +4945,8 @@ static void stac92hd_proc_hook(struct snd_info_buffer *buffer, { if (nid == codec->afg) snd_iprintf(buffer, "Power-Map: 0x%02x\n", - snd_hda_codec_read(codec, nid, 0, 0x0fec, 0x0)); + snd_hda_codec_read(codec, nid, 0, + AC_VERB_IDT_GET_POWER_MAP, 0)); } static void analog_loop_proc_hook(struct snd_info_buffer *buffer, @@ -5003,26 +4997,12 @@ static int stac92xx_resume(struct hda_codec *codec) return 0; } -static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) +static int stac92xx_suspend(struct hda_codec *codec) { stac92xx_shutup(codec); return 0; } -static int stac92xx_pre_resume(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - - /* sync mute LED */ - if (spec->vref_mute_led_nid) - stac_vrefout_set(codec, spec->vref_mute_led_nid, - spec->vref_led); - else if (spec->gpio_led) - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data); - return 0; -} - static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state) { @@ -5046,7 +5026,6 @@ static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg, #else #define stac92xx_suspend NULL #define stac92xx_resume NULL -#define stac92xx_pre_resume NULL #define stac92xx_set_power_state NULL #endif /* CONFIG_PM */ @@ -5592,9 +5571,6 @@ again: codec->patch_ops.set_power_state = stac92xx_set_power_state; } -#ifdef CONFIG_PM - codec->patch_ops.pre_resume = stac92xx_pre_resume; -#endif } err = stac92xx_parse_auto_config(codec); @@ -5901,9 +5877,6 @@ again: codec->patch_ops.set_power_state = stac92xx_set_power_state; } -#ifdef CONFIG_PM - codec->patch_ops.pre_resume = stac92xx_pre_resume; -#endif } spec->multiout.dac_nids = spec->dac_nids; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 06214fdc9486..90645560ed39 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -54,6 +54,7 @@ #include <sound/asoundef.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_jack.h" /* Pin Widget NID */ @@ -484,7 +485,7 @@ static void activate_output_mix(struct hda_codec *codec, struct nid_path *path, if (!path) return; - num = snd_hda_get_conn_list(codec, mix_nid, NULL); + num = snd_hda_get_num_conns(codec, mix_nid); for (i = 0; i < num; i++) { if (i == idx) val = AMP_IN_UNMUTE(i); @@ -532,8 +533,7 @@ static void init_output_pin(struct hda_codec *codec, hda_nid_t pin, { if (!pin) return; - snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_type); + snd_hda_set_pin_ctl(codec, pin, pin_type); if (snd_hda_query_pin_caps(codec, pin) & AC_PINCAP_EAPD) snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x02); @@ -662,12 +662,12 @@ static void via_auto_init_analog_input(struct hda_codec *codec) hda_nid_t nid = cfg->inputs[i].pin; if (spec->smart51_enabled && is_smart51_pins(codec, nid)) ctl = PIN_OUT; - else if (cfg->inputs[i].type == AUTO_PIN_MIC) - ctl = PIN_VREF50; - else + else { ctl = PIN_IN; - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, ctl); + if (cfg->inputs[i].type == AUTO_PIN_MIC) + ctl |= snd_hda_get_default_vref(codec, nid); + } + snd_hda_set_pin_ctl(codec, nid, ctl); } /* init input-src */ @@ -1006,9 +1006,7 @@ static int via_smart51_put(struct snd_kcontrol *kcontrol, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); parm &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); parm |= out_in; - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - parm); + snd_hda_set_pin_ctl(codec, nid, parm); if (out_in == AC_PINCTL_OUT_EN) { mute_aa_path(codec, 1); notify_aa_path_ctls(codec); @@ -1647,8 +1645,7 @@ static void toggle_output_mutes(struct hda_codec *codec, int num_pins, parm &= ~AC_PINCTL_OUT_EN; else parm |= AC_PINCTL_OUT_EN; - snd_hda_codec_write(codec, pins[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, parm); + snd_hda_set_pin_ctl(codec, pins[i], parm); } } @@ -1709,8 +1706,7 @@ static void via_gpio_control(struct hda_codec *codec) if (gpio_data == 0x02) { /* unmute line out */ - snd_hda_codec_write(codec, spec->autocfg.line_out_pins[0], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, + snd_hda_set_pin_ctl(codec, spec->autocfg.line_out_pins[0], PIN_OUT); if (vol_counter & 0x20) { /* decrease volume */ @@ -1728,9 +1724,7 @@ static void via_gpio_control(struct hda_codec *codec) } } else if (!(gpio_data & 0x02)) { /* mute line out */ - snd_hda_codec_write(codec, spec->autocfg.line_out_pins[0], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - 0); + snd_hda_set_pin_ctl(codec, spec->autocfg.line_out_pins[0], 0); } } @@ -1754,7 +1748,7 @@ static void via_unsol_event(struct hda_codec *codec, } #ifdef CONFIG_PM -static int via_suspend(struct hda_codec *codec, pm_message_t state) +static int via_suspend(struct hda_codec *codec) { struct via_spec *spec = codec->spec; vt1708_stop_hp_work(spec); @@ -2757,8 +2751,7 @@ static void via_auto_init_dig_in(struct hda_codec *codec) struct via_spec *spec = codec->spec; if (!spec->dig_in_nid) return; - snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN); + snd_hda_set_pin_ctl(codec, spec->autocfg.dig_in_pin, PIN_IN); } /* initialize the unsolicited events */ diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 132a86e09d07..5be2e120a14e 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2803,22 +2803,11 @@ static void __devexit snd_ice1712_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver ice1712_driver = { .name = KBUILD_MODNAME, .id_table = snd_ice1712_ids, .probe = snd_ice1712_probe, .remove = __devexit_p(snd_ice1712_remove), }; -static int __init alsa_card_ice1712_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_ice1712_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_ice1712_init) -module_exit(alsa_card_ice1712_exit) +module_pci_driver(ice1712_driver); diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 812d10e43ae0..bed9f34f4efe 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2793,9 +2793,10 @@ static void __devexit snd_vt1724_remove(struct pci_dev *pci) } #ifdef CONFIG_PM -static int snd_vt1724_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_vt1724_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_ice1712 *ice = card->private_data; if (!ice->pm_suspend_enabled) @@ -2820,13 +2821,14 @@ static int snd_vt1724_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_vt1724_resume(struct pci_dev *pci) +static int snd_vt1724_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_ice1712 *ice = card->private_data; if (!ice->pm_suspend_enabled) @@ -2871,28 +2873,21 @@ static int snd_vt1724_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -#endif -static struct pci_driver driver = { +static SIMPLE_DEV_PM_OPS(snd_vt1724_pm, snd_vt1724_suspend, snd_vt1724_resume); +#define SND_VT1724_PM_OPS &snd_vt1724_pm +#else +#define SND_VT1724_PM_OPS NULL +#endif /* CONFIG_PM */ + +static struct pci_driver vt1724_driver = { .name = KBUILD_MODNAME, .id_table = snd_vt1724_ids, .probe = snd_vt1724_probe, .remove = __devexit_p(snd_vt1724_remove), -#ifdef CONFIG_PM - .suspend = snd_vt1724_suspend, - .resume = snd_vt1724_resume, -#endif + .driver = { + .pm = SND_VT1724_PM_OPS, + }, }; -static int __init alsa_card_ice1724_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_ice1724_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_ice1724_init) -module_exit(alsa_card_ice1724_exit) +module_pci_driver(vt1724_driver); diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index e0a4263baa20..cd553f592e2d 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2624,9 +2624,10 @@ static int snd_intel8x0_free(struct intel8x0 *chip) /* * power management */ -static int intel8x0_suspend(struct pci_dev *pci, pm_message_t state) +static int intel8x0_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct intel8x0 *chip = card->private_data; int i; @@ -2658,13 +2659,14 @@ static int intel8x0_suspend(struct pci_dev *pci, pm_message_t state) /* The call below may disable built-in speaker on some laptops * after S2RAM. So, don't touch it. */ - /* pci_set_power_state(pci, pci_choose_state(pci, state)); */ + /* pci_set_power_state(pci, PCI_D3hot); */ return 0; } -static int intel8x0_resume(struct pci_dev *pci) +static int intel8x0_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct intel8x0 *chip = card->private_data; int i; @@ -2734,6 +2736,11 @@ static int intel8x0_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(intel8x0_pm, intel8x0_suspend, intel8x0_resume); +#define INTEL8X0_PM_OPS &intel8x0_pm +#else +#define INTEL8X0_PM_OPS NULL #endif /* CONFIG_PM */ #define INTEL8X0_TESTBUF_SIZE 32768 /* enough large for one shot */ @@ -3338,27 +3345,14 @@ static void __devexit snd_intel8x0_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver intel8x0_driver = { .name = KBUILD_MODNAME, .id_table = snd_intel8x0_ids, .probe = snd_intel8x0_probe, .remove = __devexit_p(snd_intel8x0_remove), -#ifdef CONFIG_PM - .suspend = intel8x0_suspend, - .resume = intel8x0_resume, -#endif + .driver = { + .pm = INTEL8X0_PM_OPS, + }, }; - -static int __init alsa_card_intel8x0_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_intel8x0_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_intel8x0_init) -module_exit(alsa_card_intel8x0_exit) +module_pci_driver(intel8x0_driver); diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index d689913a61be..da44bb3f8e7a 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -1012,9 +1012,10 @@ static int snd_intel8x0m_free(struct intel8x0m *chip) /* * power management */ -static int intel8x0m_suspend(struct pci_dev *pci, pm_message_t state) +static int intel8x0m_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct intel8x0m *chip = card->private_data; int i; @@ -1028,13 +1029,14 @@ static int intel8x0m_suspend(struct pci_dev *pci, pm_message_t state) } pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int intel8x0m_resume(struct pci_dev *pci) +static int intel8x0m_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct intel8x0m *chip = card->private_data; pci_set_power_state(pci, PCI_D0); @@ -1060,6 +1062,11 @@ static int intel8x0m_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(intel8x0m_pm, intel8x0m_suspend, intel8x0m_resume); +#define INTEL8X0M_PM_OPS &intel8x0m_pm +#else +#define INTEL8X0M_PM_OPS NULL #endif /* CONFIG_PM */ #ifdef CONFIG_PROC_FS @@ -1324,27 +1331,14 @@ static void __devexit snd_intel8x0m_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver intel8x0m_driver = { .name = KBUILD_MODNAME, .id_table = snd_intel8x0m_ids, .probe = snd_intel8x0m_probe, .remove = __devexit_p(snd_intel8x0m_remove), -#ifdef CONFIG_PM - .suspend = intel8x0m_suspend, - .resume = intel8x0m_resume, -#endif + .driver = { + .pm = INTEL8X0M_PM_OPS, + }, }; - -static int __init alsa_card_intel8x0m_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_intel8x0m_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_intel8x0m_init) -module_exit(alsa_card_intel8x0m_exit) +module_pci_driver(intel8x0m_driver); diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 8fea45ab5882..e69ce5f9c31e 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -2476,22 +2476,11 @@ static void __devexit snd_korg1212_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver korg1212_driver = { .name = KBUILD_MODNAME, .id_table = snd_korg1212_ids, .probe = snd_korg1212_probe, .remove = __devexit_p(snd_korg1212_remove), }; -static int __init alsa_card_korg1212_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_korg1212_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_korg1212_init) -module_exit(alsa_card_korg1212_exit) +module_pci_driver(korg1212_driver); diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index 375982736858..ac15166bee68 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -770,22 +770,11 @@ static DEFINE_PCI_DEVICE_TABLE(lola_ids) = { MODULE_DEVICE_TABLE(pci, lola_ids); /* pci_driver definition */ -static struct pci_driver driver = { +static struct pci_driver lola_driver = { .name = KBUILD_MODNAME, .id_table = lola_ids, .probe = lola_probe, .remove = __devexit_p(lola_remove), }; -static int __init alsa_card_lola_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_lola_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_lola_init) -module_exit(alsa_card_lola_exit) +module_pci_driver(lola_driver); diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index d94c0c292bd0..d1ab43706735 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -1141,24 +1141,11 @@ static void __devexit snd_lx6464es_remove(struct pci_dev *pci) } -static struct pci_driver driver = { +static struct pci_driver lx6464es_driver = { .name = KBUILD_MODNAME, .id_table = snd_lx6464es_ids, .probe = snd_lx6464es_probe, .remove = __devexit_p(snd_lx6464es_remove), }; - -/* module initialization */ -static int __init mod_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit mod_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(mod_init); -module_exit(mod_exit); +module_pci_driver(lx6464es_driver); diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 78229b0dad2b..c85d1ffcc955 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -361,74 +361,6 @@ MODULE_PARM_DESC(amp_gpio, "GPIO pin number for external amp. (default = -1)"); #define DSP2HOST_REQ_I2SRATE 0x02 #define DSP2HOST_REQ_TIMER 0x04 -/* AC97 registers */ -/* XXX fix this crap up */ -/*#define AC97_RESET 0x00*/ - -#define AC97_VOL_MUTE_B 0x8000 -#define AC97_VOL_M 0x1F -#define AC97_LEFT_VOL_S 8 - -#define AC97_MASTER_VOL 0x02 -#define AC97_LINE_LEVEL_VOL 0x04 -#define AC97_MASTER_MONO_VOL 0x06 -#define AC97_PC_BEEP_VOL 0x0A -#define AC97_PC_BEEP_VOL_M 0x0F -#define AC97_SROUND_MASTER_VOL 0x38 -#define AC97_PC_BEEP_VOL_S 1 - -/*#define AC97_PHONE_VOL 0x0C -#define AC97_MIC_VOL 0x0E*/ -#define AC97_MIC_20DB_ENABLE 0x40 - -/*#define AC97_LINEIN_VOL 0x10 -#define AC97_CD_VOL 0x12 -#define AC97_VIDEO_VOL 0x14 -#define AC97_AUX_VOL 0x16*/ -#define AC97_PCM_OUT_VOL 0x18 -/*#define AC97_RECORD_SELECT 0x1A*/ -#define AC97_RECORD_MIC 0x00 -#define AC97_RECORD_CD 0x01 -#define AC97_RECORD_VIDEO 0x02 -#define AC97_RECORD_AUX 0x03 -#define AC97_RECORD_MONO_MUX 0x02 -#define AC97_RECORD_DIGITAL 0x03 -#define AC97_RECORD_LINE 0x04 -#define AC97_RECORD_STEREO 0x05 -#define AC97_RECORD_MONO 0x06 -#define AC97_RECORD_PHONE 0x07 - -/*#define AC97_RECORD_GAIN 0x1C*/ -#define AC97_RECORD_VOL_M 0x0F - -/*#define AC97_GENERAL_PURPOSE 0x20*/ -#define AC97_POWER_DOWN_CTRL 0x26 -#define AC97_ADC_READY 0x0001 -#define AC97_DAC_READY 0x0002 -#define AC97_ANALOG_READY 0x0004 -#define AC97_VREF_ON 0x0008 -#define AC97_PR0 0x0100 -#define AC97_PR1 0x0200 -#define AC97_PR2 0x0400 -#define AC97_PR3 0x0800 -#define AC97_PR4 0x1000 - -#define AC97_RESERVED1 0x28 - -#define AC97_VENDOR_TEST 0x5A - -#define AC97_CLOCK_DELAY 0x5C -#define AC97_LINEOUT_MUX_SEL 0x0001 -#define AC97_MONO_MUX_SEL 0x0002 -#define AC97_CLOCK_DELAY_SEL 0x1F -#define AC97_DAC_CDS_SHIFT 6 -#define AC97_ADC_CDS_SHIFT 11 - -#define AC97_MULTI_CHANNEL_SEL 0x74 - -/*#define AC97_VENDOR_ID1 0x7C -#define AC97_VENDOR_ID2 0x7E*/ - /* * ASSP control regs */ @@ -2459,9 +2391,10 @@ static int snd_m3_free(struct snd_m3 *chip) * APM support */ #ifdef CONFIG_PM -static int m3_suspend(struct pci_dev *pci, pm_message_t state) +static int m3_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_m3 *chip = card->private_data; int i, dsp_index; @@ -2489,13 +2422,14 @@ static int m3_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int m3_resume(struct pci_dev *pci) +static int m3_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_m3 *chip = card->private_data; int i, dsp_index; @@ -2546,6 +2480,11 @@ static int m3_resume(struct pci_dev *pci) chip->in_suspend = 0; return 0; } + +static SIMPLE_DEV_PM_OPS(m3_pm, m3_suspend, m3_resume); +#define M3_PM_OPS &m3_pm +#else +#define M3_PM_OPS NULL #endif /* CONFIG_PM */ #ifdef CONFIG_SND_MAESTRO3_INPUT @@ -2837,26 +2776,14 @@ static void __devexit snd_m3_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver m3_driver = { .name = KBUILD_MODNAME, .id_table = snd_m3_ids, .probe = snd_m3_probe, .remove = __devexit_p(snd_m3_remove), -#ifdef CONFIG_PM - .suspend = m3_suspend, - .resume = m3_resume, -#endif + .driver = { + .pm = M3_PM_OPS, + }, }; -static int __init alsa_card_m3_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_m3_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_m3_init) -module_exit(alsa_card_m3_exit) +module_pci_driver(m3_driver); diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 487837c01c9f..0762610c99c0 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1380,22 +1380,11 @@ static void __devexit snd_mixart_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver mixart_driver = { .name = KBUILD_MODNAME, .id_table = snd_mixart_ids, .probe = snd_mixart_probe, .remove = __devexit_p(snd_mixart_remove), }; -static int __init alsa_card_mixart_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_mixart_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_mixart_init) -module_exit(alsa_card_mixart_exit) +module_pci_driver(mixart_driver); diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index ade2c64bd606..465cff25b146 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -1382,9 +1382,10 @@ snd_nm256_peek_for_sig(struct nm256 *chip) * APM event handler, so the card is properly reinitialized after a power * event. */ -static int nm256_suspend(struct pci_dev *pci, pm_message_t state) +static int nm256_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct nm256 *chip = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -1393,13 +1394,14 @@ static int nm256_suspend(struct pci_dev *pci, pm_message_t state) chip->coeffs_current = 0; pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int nm256_resume(struct pci_dev *pci) +static int nm256_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct nm256 *chip = card->private_data; int i; @@ -1434,6 +1436,11 @@ static int nm256_resume(struct pci_dev *pci) chip->in_resume = 0; return 0; } + +static SIMPLE_DEV_PM_OPS(nm256_pm, nm256_suspend, nm256_resume); +#define NM256_PM_OPS &nm256_pm +#else +#define NM256_PM_OPS NULL #endif /* CONFIG_PM */ static int snd_nm256_free(struct nm256 *chip) @@ -1742,27 +1749,14 @@ static void __devexit snd_nm256_remove(struct pci_dev *pci) } -static struct pci_driver driver = { +static struct pci_driver nm256_driver = { .name = KBUILD_MODNAME, .id_table = snd_nm256_ids, .probe = snd_nm256_probe, .remove = __devexit_p(snd_nm256_remove), -#ifdef CONFIG_PM - .suspend = nm256_suspend, - .resume = nm256_resume, -#endif + .driver = { + .pm = NM256_PM_OPS, + }, }; - -static int __init alsa_card_nm256_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_nm256_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_nm256_init) -module_exit(alsa_card_nm256_exit) +module_pci_driver(nm256_driver); diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index eab663eef117..37520a2b4dcf 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -94,6 +94,7 @@ enum { MODEL_2CH_OUTPUT, MODEL_HG2PCI, MODEL_XONAR_DG, + MODEL_XONAR_DGX, }; static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { @@ -109,6 +110,8 @@ static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { { OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF }, /* Asus Xonar DG */ { OXYGEN_PCI_SUBID(0x1043, 0x8467), .driver_data = MODEL_XONAR_DG }, + /* Asus Xonar DGX */ + { OXYGEN_PCI_SUBID(0x1043, 0x8521), .driver_data = MODEL_XONAR_DGX }, /* PCI 2.0 HD Audio */ { OXYGEN_PCI_SUBID(0x13f6, 0x8782), .driver_data = MODEL_2CH_OUTPUT }, /* Kuroutoshikou CMI8787-HG2PCI */ @@ -827,6 +830,11 @@ static int __devinit get_oxygen_model(struct oxygen *chip, break; case MODEL_XONAR_DG: chip->model = model_xonar_dg; + chip->model.shortname = "Xonar DG"; + break; + case MODEL_XONAR_DGX: + chip->model = model_xonar_dg; + chip->model.shortname = "Xonar DGX"; break; } if (id->driver_data == MODEL_MERIDIAN || @@ -865,20 +873,10 @@ static struct pci_driver oxygen_driver = { .probe = generic_oxygen_probe, .remove = __devexit_p(oxygen_pci_remove), #ifdef CONFIG_PM - .suspend = oxygen_pci_suspend, - .resume = oxygen_pci_resume, + .driver = { + .pm = &oxygen_pci_pm, + }, #endif }; -static int __init alsa_card_oxygen_init(void) -{ - return pci_register_driver(&oxygen_driver); -} - -static void __exit alsa_card_oxygen_exit(void) -{ - pci_unregister_driver(&oxygen_driver); -} - -module_init(alsa_card_oxygen_init) -module_exit(alsa_card_oxygen_exit) +module_pci_driver(oxygen_driver); diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index f53897a708b4..7112a89fb8bd 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -162,8 +162,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, ); void oxygen_pci_remove(struct pci_dev *pci); #ifdef CONFIG_PM -int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state); -int oxygen_pci_resume(struct pci_dev *pci); +extern const struct dev_pm_ops oxygen_pci_pm; #endif void oxygen_pci_shutdown(struct pci_dev *pci); diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 92e2d67f16a1..ab8738e21ad1 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -727,9 +727,10 @@ void oxygen_pci_remove(struct pci_dev *pci) EXPORT_SYMBOL(oxygen_pci_remove); #ifdef CONFIG_PM -int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state) +static int oxygen_pci_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct oxygen *chip = card->private_data; unsigned int i, saved_interrupt_mask; @@ -756,10 +757,9 @@ int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -EXPORT_SYMBOL(oxygen_pci_suspend); static const u32 registers_to_restore[OXYGEN_IO_SIZE / 32] = { 0xffffffff, 0x00ff077f, 0x00011d08, 0x007f00ff, @@ -787,9 +787,10 @@ static void oxygen_restore_ac97(struct oxygen *chip, unsigned int codec) chip->saved_ac97_registers[codec][i]); } -int oxygen_pci_resume(struct pci_dev *pci) +static int oxygen_pci_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct oxygen *chip = card->private_data; unsigned int i; @@ -820,7 +821,9 @@ int oxygen_pci_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -EXPORT_SYMBOL(oxygen_pci_resume); + +SIMPLE_DEV_PM_OPS(oxygen_pci_pm, oxygen_pci_suspend, oxygen_pci_resume); +EXPORT_SYMBOL(oxygen_pci_pm); #endif /* CONFIG_PM */ void oxygen_pci_shutdown(struct pci_dev *pci) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 3fdee4950174..d3b606b69f3b 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -94,21 +94,11 @@ static struct pci_driver xonar_driver = { .probe = xonar_probe, .remove = __devexit_p(oxygen_pci_remove), #ifdef CONFIG_PM - .suspend = oxygen_pci_suspend, - .resume = oxygen_pci_resume, + .driver = { + .pm = &oxygen_pci_pm, + }, #endif .shutdown = oxygen_pci_shutdown, }; -static int __init alsa_card_xonar_init(void) -{ - return pci_register_driver(&xonar_driver); -} - -static void __exit alsa_card_xonar_exit(void) -{ - pci_unregister_driver(&xonar_driver); -} - -module_init(alsa_card_xonar_init) -module_exit(alsa_card_xonar_exit) +module_pci_driver(xonar_driver); diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index 793bdf03d7e0..77acd790ea47 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -1,5 +1,5 @@ /* - * card driver for the Xonar DG + * card driver for the Xonar DG/DGX * * Copyright (c) Clemens Ladisch <clemens@ladisch.de> * @@ -17,8 +17,8 @@ */ /* - * Xonar DG - * -------- + * Xonar DG/DGX + * ------------ * * CMI8788: * @@ -581,7 +581,6 @@ static void dump_cs4245_registers(struct oxygen *chip, } struct oxygen_model model_xonar_dg = { - .shortname = "Xonar DG", .longname = "C-Media Oxygen HD Audio", .chip = "CMI8786", .init = dg_init, diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index fd1809ab73b4..e3ac1f768ff6 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1368,6 +1368,67 @@ static void pcxhr_proc_gpo_write(struct snd_info_entry *entry, } } +/* Access to the results of the CMD_GET_TIME_CODE RMH */ +#define TIME_CODE_VALID_MASK 0x00800000 +#define TIME_CODE_NEW_MASK 0x00400000 +#define TIME_CODE_BACK_MASK 0x00200000 +#define TIME_CODE_WAIT_MASK 0x00100000 + +/* Values for the CMD_MANAGE_SIGNAL RMH */ +#define MANAGE_SIGNAL_TIME_CODE 0x01 +#define MANAGE_SIGNAL_MIDI 0x02 + +/* linear time code read proc*/ +static void pcxhr_proc_ltc(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_pcxhr *chip = entry->private_data; + struct pcxhr_mgr *mgr = chip->mgr; + struct pcxhr_rmh rmh; + unsigned int ltcHrs, ltcMin, ltcSec, ltcFrm; + int err; + /* commands available when embedded DSP is running */ + if (!(mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX))) { + snd_iprintf(buffer, "no firmware loaded\n"); + return; + } + if (!mgr->capture_ltc) { + pcxhr_init_rmh(&rmh, CMD_MANAGE_SIGNAL); + rmh.cmd[0] |= MANAGE_SIGNAL_TIME_CODE; + err = pcxhr_send_msg(mgr, &rmh); + if (err) { + snd_iprintf(buffer, "ltc not activated (%d)\n", err); + return; + } + if (mgr->is_hr_stereo) + hr222_manage_timecode(mgr, 1); + else + pcxhr_write_io_num_reg_cont(mgr, REG_CONT_VALSMPTE, + REG_CONT_VALSMPTE, NULL); + mgr->capture_ltc = 1; + } + pcxhr_init_rmh(&rmh, CMD_GET_TIME_CODE); + err = pcxhr_send_msg(mgr, &rmh); + if (err) { + snd_iprintf(buffer, "ltc read error (err=%d)\n", err); + return ; + } + ltcHrs = 10*((rmh.stat[0] >> 8) & 0x3) + (rmh.stat[0] & 0xf); + ltcMin = 10*((rmh.stat[1] >> 16) & 0x7) + ((rmh.stat[1] >> 8) & 0xf); + ltcSec = 10*(rmh.stat[1] & 0x7) + ((rmh.stat[2] >> 16) & 0xf); + ltcFrm = 10*((rmh.stat[2] >> 8) & 0x3) + (rmh.stat[2] & 0xf); + + snd_iprintf(buffer, "timecode: %02u:%02u:%02u-%02u\n", + ltcHrs, ltcMin, ltcSec, ltcFrm); + snd_iprintf(buffer, "raw: 0x%04x%06x%06x\n", rmh.stat[0] & 0x00ffff, + rmh.stat[1] & 0xffffff, rmh.stat[2] & 0xffffff); + /*snd_iprintf(buffer, "dsp ref time: 0x%06x%06x\n", + rmh.stat[3] & 0xffffff, rmh.stat[4] & 0xffffff);*/ + if (!(rmh.stat[0] & TIME_CODE_VALID_MASK)) { + snd_iprintf(buffer, "warning: linear timecode not valid\n"); + } +} + static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip) { struct snd_info_entry *entry; @@ -1383,6 +1444,8 @@ static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip) entry->c.text.write = pcxhr_proc_gpo_write; entry->mode |= S_IWUSR; } + if (!snd_card_proc_new(chip->card, "ltc", &entry)) + snd_info_set_text_ops(entry, chip, pcxhr_proc_ltc); } /* end of proc interface */ @@ -1607,22 +1670,11 @@ static void __devexit pcxhr_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver pcxhr_driver = { .name = KBUILD_MODNAME, .id_table = pcxhr_ids, .probe = pcxhr_probe, .remove = __devexit_p(pcxhr_remove), }; -static int __init pcxhr_module_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit pcxhr_module_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(pcxhr_module_init) -module_exit(pcxhr_module_exit) +module_pci_driver(pcxhr_driver); diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h index bda776c49884..a4c602c45173 100644 --- a/sound/pci/pcxhr/pcxhr.h +++ b/sound/pci/pcxhr/pcxhr.h @@ -103,6 +103,7 @@ struct pcxhr_mgr { unsigned int board_has_mic:1; /* if 1 the board has microphone input */ unsigned int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */ unsigned int mono_capture:1; /* if 1 the board does mono capture */ + unsigned int capture_ltc:1; /* if 1 the board captures LTC input */ struct snd_dma_buffer hostport; diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index 304411c1fe4b..b33db1e006e7 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -504,6 +504,8 @@ static struct pcxhr_cmd_info pcxhr_dsp_cmds[] = { [CMD_FORMAT_STREAM_IN] = { 0x870000, 0, RMH_SSIZE_FIXED }, [CMD_STREAM_SAMPLE_COUNT] = { 0x902000, 2, RMH_SSIZE_FIXED }, [CMD_AUDIO_LEVEL_ADJUST] = { 0xc22000, 0, RMH_SSIZE_FIXED }, +[CMD_GET_TIME_CODE] = { 0x060000, 5, RMH_SSIZE_FIXED }, +[CMD_MANAGE_SIGNAL] = { 0x0f0000, 0, RMH_SSIZE_FIXED }, }; #ifdef CONFIG_SND_DEBUG_VERBOSE @@ -533,6 +535,8 @@ static char* cmd_names[] = { [CMD_FORMAT_STREAM_IN] = "CMD_FORMAT_STREAM_IN", [CMD_STREAM_SAMPLE_COUNT] = "CMD_STREAM_SAMPLE_COUNT", [CMD_AUDIO_LEVEL_ADJUST] = "CMD_AUDIO_LEVEL_ADJUST", +[CMD_GET_TIME_CODE] = "CMD_GET_TIME_CODE", +[CMD_MANAGE_SIGNAL] = "CMD_MANAGE_SIGNAL", }; #endif @@ -1133,13 +1137,12 @@ static u_int64_t pcxhr_stream_read_position(struct pcxhr_mgr *mgr, hw_sample_count = ((u_int64_t)rmh.stat[0]) << 24; hw_sample_count += (u_int64_t)rmh.stat[1]; - snd_printdd("stream %c%d : abs samples real(%ld) timer(%ld)\n", + snd_printdd("stream %c%d : abs samples real(%llu) timer(%llu)\n", stream->pipe->is_capture ? 'C' : 'P', stream->substream->number, - (long unsigned int)hw_sample_count, - (long unsigned int)(stream->timer_abs_periods + - stream->timer_period_frag + - mgr->granularity)); + hw_sample_count, + stream->timer_abs_periods + stream->timer_period_frag + + mgr->granularity); return hw_sample_count; } @@ -1243,10 +1246,18 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) if ((dsp_time_diff < 0) && (mgr->dsp_time_last != PCXHR_DSP_TIME_INVALID)) { - snd_printdd("ERROR DSP TIME old(%d) new(%d) -> " - "resynchronize all streams\n", + /* handle dsp counter wraparound without resync */ + int tmp_diff = dsp_time_diff + PCXHR_DSP_TIME_MASK + 1; + snd_printdd("WARNING DSP timestamp old(%d) new(%d)", mgr->dsp_time_last, dsp_time_new); - mgr->dsp_time_err++; + if (tmp_diff > 0 && tmp_diff <= (2*mgr->granularity)) { + snd_printdd("-> timestamp wraparound OK: " + "diff=%d\n", tmp_diff); + dsp_time_diff = tmp_diff; + } else { + snd_printdd("-> resynchronize all streams\n"); + mgr->dsp_time_err++; + } } #ifdef CONFIG_SND_DEBUG_VERBOSE if (dsp_time_diff == 0) diff --git a/sound/pci/pcxhr/pcxhr_core.h b/sound/pci/pcxhr/pcxhr_core.h index be0173796cdb..a81ab6b811e7 100644 --- a/sound/pci/pcxhr/pcxhr_core.h +++ b/sound/pci/pcxhr/pcxhr_core.h @@ -79,6 +79,8 @@ enum { CMD_FORMAT_STREAM_IN, /* cmd_len >= 4 stat_len = 0 */ CMD_STREAM_SAMPLE_COUNT, /* cmd_len = 2 stat_len = (2 * nb_stream) */ CMD_AUDIO_LEVEL_ADJUST, /* cmd_len = 3 stat_len = 0 */ + CMD_GET_TIME_CODE, /* cmd_len = 1 stat_len = 5 */ + CMD_MANAGE_SIGNAL, /* cmd_len = 1 stat_len = 0 */ CMD_LAST_INDEX }; @@ -116,7 +118,7 @@ int pcxhr_send_msg(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh); #define IO_NUM_REG_OUT_ANA_LEVEL 20 #define IO_NUM_REG_IN_ANA_LEVEL 21 - +#define REG_CONT_VALSMPTE 0x000800 #define REG_CONT_UNMUTE_INPUTS 0x020000 /* parameters used with register IO_NUM_REG_STATUS */ diff --git a/sound/pci/pcxhr/pcxhr_mix22.c b/sound/pci/pcxhr/pcxhr_mix22.c index 1cb82c0a9cb3..84fe57626eba 100644 --- a/sound/pci/pcxhr/pcxhr_mix22.c +++ b/sound/pci/pcxhr/pcxhr_mix22.c @@ -53,6 +53,7 @@ #define PCXHR_DSP_RESET_DSP 0x01 #define PCXHR_DSP_RESET_MUTE 0x02 #define PCXHR_DSP_RESET_CODEC 0x08 +#define PCXHR_DSP_RESET_SMPTE 0x10 #define PCXHR_DSP_RESET_GPO_OFFSET 5 #define PCXHR_DSP_RESET_GPO_MASK 0x60 @@ -527,6 +528,16 @@ int hr222_write_gpo(struct pcxhr_mgr *mgr, int value) return 0; } +int hr222_manage_timecode(struct pcxhr_mgr *mgr, int enable) +{ + if (enable) + mgr->dsp_reset |= PCXHR_DSP_RESET_SMPTE; + else + mgr->dsp_reset &= ~PCXHR_DSP_RESET_SMPTE; + + PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, mgr->dsp_reset); + return 0; +} int hr222_update_analog_audio_level(struct snd_pcxhr *chip, int is_capture, int channel) diff --git a/sound/pci/pcxhr/pcxhr_mix22.h b/sound/pci/pcxhr/pcxhr_mix22.h index 5a37a0007e8f..5971b9933f41 100644 --- a/sound/pci/pcxhr/pcxhr_mix22.h +++ b/sound/pci/pcxhr/pcxhr_mix22.h @@ -34,6 +34,7 @@ int hr222_get_external_clock(struct pcxhr_mgr *mgr, int hr222_read_gpio(struct pcxhr_mgr *mgr, int is_gpi, int *value); int hr222_write_gpo(struct pcxhr_mgr *mgr, int value); +int hr222_manage_timecode(struct pcxhr_mgr *mgr, int enable); #define HR222_LINE_PLAYBACK_LEVEL_MIN 0 /* -25.5 dB */ #define HR222_LINE_PLAYBACK_ZERO_LEVEL 51 /* 0.0 dB */ diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 0481d94aac9b..760ee467cd9a 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1151,9 +1151,10 @@ static void riptide_handleirq(unsigned long dev_id) } #ifdef CONFIG_PM -static int riptide_suspend(struct pci_dev *pci, pm_message_t state) +static int riptide_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_riptide *chip = card->private_data; chip->in_suspend = 1; @@ -1162,13 +1163,14 @@ static int riptide_suspend(struct pci_dev *pci, pm_message_t state) snd_ac97_suspend(chip->ac97); pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int riptide_resume(struct pci_dev *pci) +static int riptide_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_riptide *chip = card->private_data; pci_set_power_state(pci, PCI_D0); @@ -1186,7 +1188,12 @@ static int riptide_resume(struct pci_dev *pci) chip->in_suspend = 0; return 0; } -#endif + +static SIMPLE_DEV_PM_OPS(riptide_pm, riptide_suspend, riptide_resume); +#define RIPTIDE_PM_OPS &riptide_pm +#else +#define RIPTIDE_PM_OPS NULL +#endif /* CONFIG_PM */ static int try_to_load_firmware(struct cmdif *cif, struct snd_riptide *chip) { @@ -1837,8 +1844,7 @@ static int snd_riptide_free(struct snd_riptide *chip) } if (chip->irq >= 0) free_irq(chip->irq, chip); - if (chip->fw_entry) - release_firmware(chip->fw_entry); + release_firmware(chip->fw_entry); release_and_free_resource(chip->res_port); kfree(chip); return 0; @@ -2181,10 +2187,9 @@ static struct pci_driver driver = { .id_table = snd_riptide_ids, .probe = snd_card_riptide_probe, .remove = __devexit_p(snd_card_riptide_remove), -#ifdef CONFIG_PM - .suspend = riptide_suspend, - .resume = riptide_resume, -#endif + .driver = { + .pm = RIPTIDE_PM_OPS, + }, }; #ifdef SUPPORT_JOYSTICK diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index b4819d5e41db..46b3629dda22 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -1984,22 +1984,11 @@ static void __devexit snd_rme32_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver rme32_driver = { .name = KBUILD_MODNAME, .id_table = snd_rme32_ids, .probe = snd_rme32_probe, .remove = __devexit_p(snd_rme32_remove), }; -static int __init alsa_card_rme32_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_rme32_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_rme32_init) -module_exit(alsa_card_rme32_exit) +module_pci_driver(rme32_driver); diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index ba894158e76c..9b98dc406988 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -2395,22 +2395,11 @@ static void __devexit snd_rme96_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver rme96_driver = { .name = KBUILD_MODNAME, .id_table = snd_rme96_ids, .probe = snd_rme96_probe, .remove = __devexit_p(snd_rme96_remove), }; -static int __init alsa_card_rme96_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_rme96_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_rme96_init) -module_exit(alsa_card_rme96_exit) +module_pci_driver(rme96_driver); diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 0b2aea2ce172..0d6930c4f4b7 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5636,22 +5636,11 @@ static void __devexit snd_hdsp_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver hdsp_driver = { .name = KBUILD_MODNAME, .id_table = snd_hdsp_ids, .probe = snd_hdsp_probe, .remove = __devexit_p(snd_hdsp_remove), }; -static int __init alsa_card_hdsp_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_hdsp_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_hdsp_init) -module_exit(alsa_card_hdsp_exit) +module_pci_driver(hdsp_driver); diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index bc030a2088da..b8ac8710f47f 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1988,6 +1988,13 @@ static int hdspm_get_system_sample_rate(struct hdspm *hdspm) period = hdspm_read(hdspm, HDSPM_RD_PLL_FREQ); rate = hdspm_calc_dds_value(hdspm, period); + if (rate > 207000) { + /* Unreasonable high sample rate as seen on PCI MADI cards. + * Use the cached value instead. + */ + rate = hdspm->system_sample_rate; + } + return rate; } @@ -6918,23 +6925,11 @@ static void __devexit snd_hdspm_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver hdspm_driver = { .name = KBUILD_MODNAME, .id_table = snd_hdspm_ids, .probe = snd_hdspm_probe, .remove = __devexit_p(snd_hdspm_remove), }; - -static int __init alsa_card_hdspm_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_hdspm_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_hdspm_init) -module_exit(alsa_card_hdspm_exit) +module_pci_driver(hdspm_driver); diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index b737d1619cc7..a15fc100ab0c 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -2631,22 +2631,11 @@ static void __devexit snd_rme9652_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver rme9652_driver = { .name = KBUILD_MODNAME, .id_table = snd_rme9652_ids, .probe = snd_rme9652_probe, .remove = __devexit_p(snd_rme9652_remove), }; -static int __init alsa_card_hammerfall_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_hammerfall_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_hammerfall_init) -module_exit(alsa_card_hammerfall_exit) +module_pci_driver(rme9652_driver); diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index ff500a87f769..512434efcc31 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1209,9 +1209,10 @@ static int sis_chip_init(struct sis7019 *sis) } #ifdef CONFIG_PM -static int sis_suspend(struct pci_dev *pci, pm_message_t state) +static int sis_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct sis7019 *sis = card->private_data; void __iomem *ioaddr = sis->ioaddr; int i; @@ -1241,13 +1242,14 @@ static int sis_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int sis_resume(struct pci_dev *pci) +static int sis_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct sis7019 *sis = card->private_data; void __iomem *ioaddr = sis->ioaddr; int i; @@ -1298,6 +1300,11 @@ error: snd_card_disconnect(card); return -EIO; } + +static SIMPLE_DEV_PM_OPS(sis_pm, sis_suspend, sis_resume); +#define SIS_PM_OPS &sis_pm +#else +#define SIS_PM_OPS NULL #endif /* CONFIG_PM */ static int sis_alloc_suspend(struct sis7019 *sis) @@ -1481,22 +1488,9 @@ static struct pci_driver sis7019_driver = { .id_table = snd_sis7019_ids, .probe = snd_sis7019_probe, .remove = __devexit_p(snd_sis7019_remove), - -#ifdef CONFIG_PM - .suspend = sis_suspend, - .resume = sis_resume, -#endif + .driver = { + .pm = SIS_PM_OPS, + }, }; -static int __init sis7019_init(void) -{ - return pci_register_driver(&sis7019_driver); -} - -static void __exit sis7019_exit(void) -{ - pci_unregister_driver(&sis7019_driver); -} - -module_init(sis7019_init); -module_exit(sis7019_exit); +module_pci_driver(sis7019_driver); diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 54cc802050f7..baa9946bedf0 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -1530,22 +1530,11 @@ static void __devexit snd_sonic_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver sonicvibes_driver = { .name = KBUILD_MODNAME, .id_table = snd_sonic_ids, .probe = snd_sonic_probe, .remove = __devexit_p(snd_sonic_remove), }; -static int __init alsa_card_sonicvibes_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_sonicvibes_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_sonicvibes_init) -module_exit(alsa_card_sonicvibes_exit) +module_pci_driver(sonicvibes_driver); diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index 5f1def7f45e5..d36e6ca147e1 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -26,7 +26,7 @@ #include <linux/time.h> #include <linux/module.h> #include <sound/core.h> -#include <sound/trident.h> +#include "trident.h" #include <sound/initval.h> MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, <audio@tridentmicro.com>"); @@ -172,26 +172,16 @@ static void __devexit snd_trident_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver trident_driver = { .name = KBUILD_MODNAME, .id_table = snd_trident_ids, .probe = snd_trident_probe, .remove = __devexit_p(snd_trident_remove), #ifdef CONFIG_PM - .suspend = snd_trident_suspend, - .resume = snd_trident_resume, + .driver = { + .pm = &snd_trident_pm, + }, #endif }; -static int __init alsa_card_trident_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_trident_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_trident_init) -module_exit(alsa_card_trident_exit) +module_pci_driver(trident_driver); diff --git a/sound/pci/trident/trident.h b/sound/pci/trident/trident.h new file mode 100644 index 000000000000..5f110eb56e47 --- /dev/null +++ b/sound/pci/trident/trident.h @@ -0,0 +1,444 @@ +#ifndef __SOUND_TRIDENT_H +#define __SOUND_TRIDENT_H + +/* + * audio@tridentmicro.com + * Fri Feb 19 15:55:28 MST 1999 + * Definitions for Trident 4DWave DX/NX chips + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <sound/pcm.h> +#include <sound/mpu401.h> +#include <sound/ac97_codec.h> +#include <sound/util_mem.h> + +#define TRIDENT_DEVICE_ID_DX ((PCI_VENDOR_ID_TRIDENT<<16)|PCI_DEVICE_ID_TRIDENT_4DWAVE_DX) +#define TRIDENT_DEVICE_ID_NX ((PCI_VENDOR_ID_TRIDENT<<16)|PCI_DEVICE_ID_TRIDENT_4DWAVE_NX) +#define TRIDENT_DEVICE_ID_SI7018 ((PCI_VENDOR_ID_SI<<16)|PCI_DEVICE_ID_SI_7018) + +#define SNDRV_TRIDENT_VOICE_TYPE_PCM 0 +#define SNDRV_TRIDENT_VOICE_TYPE_SYNTH 1 +#define SNDRV_TRIDENT_VOICE_TYPE_MIDI 2 + +#define SNDRV_TRIDENT_VFLG_RUNNING (1<<0) + +/* TLB code constants */ +#define SNDRV_TRIDENT_PAGE_SIZE 4096 +#define SNDRV_TRIDENT_PAGE_SHIFT 12 +#define SNDRV_TRIDENT_PAGE_MASK ((1<<SNDRV_TRIDENT_PAGE_SHIFT)-1) +#define SNDRV_TRIDENT_MAX_PAGES 4096 + +/* + * Direct registers + */ + +#define TRID_REG(trident, x) ((trident)->port + (x)) + +#define ID_4DWAVE_DX 0x2000 +#define ID_4DWAVE_NX 0x2001 + +/* Bank definitions */ + +#define T4D_BANK_A 0 +#define T4D_BANK_B 1 +#define T4D_NUM_BANKS 2 + +/* Register definitions */ + +/* Global registers */ + +enum global_control_bits { + CHANNEL_IDX = 0x0000003f, + OVERRUN_IE = 0x00000400, /* interrupt enable: capture overrun */ + UNDERRUN_IE = 0x00000800, /* interrupt enable: playback underrun */ + ENDLP_IE = 0x00001000, /* interrupt enable: end of buffer */ + MIDLP_IE = 0x00002000, /* interrupt enable: middle buffer */ + ETOG_IE = 0x00004000, /* interrupt enable: envelope toggling */ + EDROP_IE = 0x00008000, /* interrupt enable: envelope drop */ + BANK_B_EN = 0x00010000, /* SiS: enable bank B (64 channels) */ + PCMIN_B_MIX = 0x00020000, /* SiS: PCM IN B mixing enable */ + I2S_OUT_ASSIGN = 0x00040000, /* SiS: I2S Out contains surround PCM */ + SPDIF_OUT_ASSIGN= 0x00080000, /* SiS: 0=S/PDIF L/R | 1=PCM Out FIFO */ + MAIN_OUT_ASSIGN = 0x00100000, /* SiS: 0=PCM Out FIFO | 1=MMC Out buffer */ +}; + +enum miscint_bits { + PB_UNDERRUN_IRQ = 0x00000001, REC_OVERRUN_IRQ = 0x00000002, + SB_IRQ = 0x00000004, MPU401_IRQ = 0x00000008, + OPL3_IRQ = 0x00000010, ADDRESS_IRQ = 0x00000020, + ENVELOPE_IRQ = 0x00000040, PB_UNDERRUN = 0x00000100, + REC_OVERRUN = 0x00000200, MIXER_UNDERFLOW = 0x00000400, + MIXER_OVERFLOW = 0x00000800, NX_SB_IRQ_DISABLE = 0x00001000, + ST_TARGET_REACHED = 0x00008000, + PB_24K_MODE = 0x00010000, ST_IRQ_EN = 0x00800000, + ACGPIO_IRQ = 0x01000000 +}; + +/* T2 legacy dma control registers. */ +#define LEGACY_DMAR0 0x00 // ADR0 +#define LEGACY_DMAR4 0x04 // CNT0 +#define LEGACY_DMAR6 0x06 // CNT0 - High bits +#define LEGACY_DMAR11 0x0b // MOD +#define LEGACY_DMAR15 0x0f // MMR + +#define T4D_START_A 0x80 +#define T4D_STOP_A 0x84 +#define T4D_DLY_A 0x88 +#define T4D_SIGN_CSO_A 0x8c +#define T4D_CSPF_A 0x90 +#define T4D_CSPF_B 0xbc +#define T4D_CEBC_A 0x94 +#define T4D_AINT_A 0x98 +#define T4D_AINTEN_A 0x9c +#define T4D_LFO_GC_CIR 0xa0 +#define T4D_MUSICVOL_WAVEVOL 0xa8 +#define T4D_SBDELTA_DELTA_R 0xac +#define T4D_MISCINT 0xb0 +#define T4D_START_B 0xb4 +#define T4D_STOP_B 0xb8 +#define T4D_SBBL_SBCL 0xc0 +#define T4D_SBCTRL_SBE2R_SBDD 0xc4 +#define T4D_STIMER 0xc8 +#define T4D_AINT_B 0xd8 +#define T4D_AINTEN_B 0xdc +#define T4D_RCI 0x70 + +/* MPU-401 UART */ +#define T4D_MPU401_BASE 0x20 +#define T4D_MPUR0 0x20 +#define T4D_MPUR1 0x21 +#define T4D_MPUR2 0x22 +#define T4D_MPUR3 0x23 + +/* S/PDIF Registers */ +#define NX_SPCTRL_SPCSO 0x24 +#define NX_SPLBA 0x28 +#define NX_SPESO 0x2c +#define NX_SPCSTATUS 0x64 + +/* Joystick */ +#define GAMEPORT_GCR 0x30 +#define GAMEPORT_MODE_ADC 0x80 +#define GAMEPORT_LEGACY 0x31 +#define GAMEPORT_AXES 0x34 + +/* NX Specific Registers */ +#define NX_TLBC 0x6c + +/* Channel Registers */ + +#define CH_START 0xe0 + +#define CH_DX_CSO_ALPHA_FMS 0xe0 +#define CH_DX_ESO_DELTA 0xe8 +#define CH_DX_FMC_RVOL_CVOL 0xec + +#define CH_NX_DELTA_CSO 0xe0 +#define CH_NX_DELTA_ESO 0xe8 +#define CH_NX_ALPHA_FMS_FMC_RVOL_CVOL 0xec + +#define CH_LBA 0xe4 +#define CH_GVSEL_PAN_VOL_CTRL_EC 0xf0 +#define CH_EBUF1 0xf4 +#define CH_EBUF2 0xf8 + +/* AC-97 Registers */ + +#define DX_ACR0_AC97_W 0x40 +#define DX_ACR1_AC97_R 0x44 +#define DX_ACR2_AC97_COM_STAT 0x48 + +#define NX_ACR0_AC97_COM_STAT 0x40 +#define NX_ACR1_AC97_W 0x44 +#define NX_ACR2_AC97_R_PRIMARY 0x48 +#define NX_ACR3_AC97_R_SECONDARY 0x4c + +#define SI_AC97_WRITE 0x40 +#define SI_AC97_READ 0x44 +#define SI_SERIAL_INTF_CTRL 0x48 +#define SI_AC97_GPIO 0x4c +#define SI_ASR0 0x50 +#define SI_SPDIF_CS 0x70 +#define SI_GPIO 0x7c + +enum trident_nx_ac97_bits { + /* ACR1-3 */ + NX_AC97_BUSY_WRITE = 0x0800, + NX_AC97_BUSY_READ = 0x0800, + NX_AC97_BUSY_DATA = 0x0400, + NX_AC97_WRITE_SECONDARY = 0x0100, + /* ACR0 */ + NX_AC97_SECONDARY_READY = 0x0040, + NX_AC97_SECONDARY_RECORD = 0x0020, + NX_AC97_SURROUND_OUTPUT = 0x0010, + NX_AC97_PRIMARY_READY = 0x0008, + NX_AC97_PRIMARY_RECORD = 0x0004, + NX_AC97_PCM_OUTPUT = 0x0002, + NX_AC97_WARM_RESET = 0x0001 +}; + +enum trident_dx_ac97_bits { + DX_AC97_BUSY_WRITE = 0x8000, + DX_AC97_BUSY_READ = 0x8000, + DX_AC97_READY = 0x0010, + DX_AC97_RECORD = 0x0008, + DX_AC97_PLAYBACK = 0x0002 +}; + +enum sis7018_ac97_bits { + SI_AC97_BUSY_WRITE = 0x00008000, + SI_AC97_AUDIO_BUSY = 0x00004000, + SI_AC97_MODEM_BUSY = 0x00002000, + SI_AC97_BUSY_READ = 0x00008000, + SI_AC97_SECONDARY = 0x00000080, +}; + +enum serial_intf_ctrl_bits { + WARM_RESET = 0x00000001, + COLD_RESET = 0x00000002, + I2S_CLOCK = 0x00000004, + PCM_SEC_AC97 = 0x00000008, + AC97_DBL_RATE = 0x00000010, + SPDIF_EN = 0x00000020, + I2S_OUTPUT_EN = 0x00000040, + I2S_INPUT_EN = 0x00000080, + PCMIN = 0x00000100, + LINE1IN = 0x00000200, + MICIN = 0x00000400, + LINE2IN = 0x00000800, + HEAD_SET_IN = 0x00001000, + GPIOIN = 0x00002000, + /* 7018 spec says id = 01 but the demo board routed to 10 + SECONDARY_ID= 0x00004000, */ + SECONDARY_ID = 0x00004000, + PCMOUT = 0x00010000, + SURROUT = 0x00020000, + CENTEROUT = 0x00040000, + LFEOUT = 0x00080000, + LINE1OUT = 0x00100000, + LINE2OUT = 0x00200000, + GPIOOUT = 0x00400000, + SI_AC97_PRIMARY_READY = 0x01000000, + SI_AC97_SECONDARY_READY = 0x02000000, + SI_AC97_POWERDOWN = 0x04000000, +}; + +/* PCM defaults */ + +#define T4D_DEFAULT_PCM_VOL 10 /* 0 - 255 */ +#define T4D_DEFAULT_PCM_PAN 0 /* 0 - 127 */ +#define T4D_DEFAULT_PCM_RVOL 127 /* 0 - 127 */ +#define T4D_DEFAULT_PCM_CVOL 127 /* 0 - 127 */ + +struct snd_trident; +struct snd_trident_voice; +struct snd_trident_pcm_mixer; + +struct snd_trident_port { + struct snd_midi_channel_set * chset; + struct snd_trident * trident; + int mode; /* operation mode */ + int client; /* sequencer client number */ + int port; /* sequencer port number */ + unsigned int midi_has_voices: 1; +}; + +struct snd_trident_memblk_arg { + short first_page, last_page; +}; + +struct snd_trident_tlb { + unsigned int * entries; /* 16k-aligned TLB table */ + dma_addr_t entries_dmaaddr; /* 16k-aligned PCI address to TLB table */ + unsigned long * shadow_entries; /* shadow entries with virtual addresses */ + struct snd_dma_buffer buffer; + struct snd_util_memhdr * memhdr; /* page allocation list */ + struct snd_dma_buffer silent_page; +}; + +struct snd_trident_voice { + unsigned int number; + unsigned int use: 1, + pcm: 1, + synth:1, + midi: 1; + unsigned int flags; + unsigned char client; + unsigned char port; + unsigned char index; + + struct snd_trident_sample_ops *sample_ops; + + /* channel parameters */ + unsigned int CSO; /* 24 bits (16 on DX) */ + unsigned int ESO; /* 24 bits (16 on DX) */ + unsigned int LBA; /* 30 bits */ + unsigned short EC; /* 12 bits */ + unsigned short Alpha; /* 12 bits */ + unsigned short Delta; /* 16 bits */ + unsigned short Attribute; /* 16 bits - SiS 7018 */ + unsigned short Vol; /* 12 bits (6.6) */ + unsigned char Pan; /* 7 bits (1.4.2) */ + unsigned char GVSel; /* 1 bit */ + unsigned char RVol; /* 7 bits (5.2) */ + unsigned char CVol; /* 7 bits (5.2) */ + unsigned char FMC; /* 2 bits */ + unsigned char CTRL; /* 4 bits */ + unsigned char FMS; /* 4 bits */ + unsigned char LFO; /* 8 bits */ + + unsigned int negCSO; /* nonzero - use negative CSO */ + + struct snd_util_memblk *memblk; /* memory block if TLB enabled */ + + /* PCM data */ + + struct snd_trident *trident; + struct snd_pcm_substream *substream; + struct snd_trident_voice *extra; /* extra PCM voice (acts as interrupt generator) */ + unsigned int running: 1, + capture: 1, + spdif: 1, + foldback: 1, + isync: 1, + isync2: 1, + isync3: 1; + int foldback_chan; /* foldback subdevice number */ + unsigned int stimer; /* global sample timer (to detect spurious interrupts) */ + unsigned int spurious_threshold; /* spurious threshold */ + unsigned int isync_mark; + unsigned int isync_max; + unsigned int isync_ESO; + + /* --- */ + + void *private_data; + void (*private_free)(struct snd_trident_voice *voice); +}; + +struct snd_4dwave { + int seq_client; + + struct snd_trident_port seq_ports[4]; + struct snd_trident_voice voices[64]; + + int ChanSynthCount; /* number of allocated synth channels */ + int max_size; /* maximum synth memory size in bytes */ + int current_size; /* current allocated synth mem in bytes */ +}; + +struct snd_trident_pcm_mixer { + struct snd_trident_voice *voice; /* active voice */ + unsigned short vol; /* front volume */ + unsigned char pan; /* pan control */ + unsigned char rvol; /* rear volume */ + unsigned char cvol; /* center volume */ + unsigned char pad; +}; + +struct snd_trident { + int irq; + + unsigned int device; /* device ID */ + + unsigned char bDMAStart; + + unsigned long port; + unsigned long midi_port; + + unsigned int spurious_irq_count; + unsigned int spurious_irq_max_delta; + + struct snd_trident_tlb tlb; /* TLB entries for NX cards */ + + unsigned char spdif_ctrl; + unsigned char spdif_pcm_ctrl; + unsigned int spdif_bits; + unsigned int spdif_pcm_bits; + struct snd_kcontrol *spdif_pcm_ctl; /* S/PDIF settings */ + unsigned int ac97_ctrl; + + unsigned int ChanMap[2]; /* allocation map for hardware channels */ + + int ChanPCM; /* max number of PCM channels */ + int ChanPCMcnt; /* actual number of PCM channels */ + + unsigned int ac97_detect: 1; /* 1 = AC97 in detection phase */ + unsigned int in_suspend: 1; /* 1 during suspend/resume */ + + struct snd_4dwave synth; /* synth specific variables */ + + spinlock_t event_lock; + spinlock_t voice_alloc; + + struct snd_dma_device dma_dev; + + struct pci_dev *pci; + struct snd_card *card; + struct snd_pcm *pcm; /* ADC/DAC PCM */ + struct snd_pcm *foldback; /* Foldback PCM */ + struct snd_pcm *spdif; /* SPDIF PCM */ + struct snd_rawmidi *rmidi; + + struct snd_ac97_bus *ac97_bus; + struct snd_ac97 *ac97; + struct snd_ac97 *ac97_sec; + + unsigned int musicvol_wavevol; + struct snd_trident_pcm_mixer pcm_mixer[32]; + struct snd_kcontrol *ctl_vol; /* front volume */ + struct snd_kcontrol *ctl_pan; /* pan */ + struct snd_kcontrol *ctl_rvol; /* rear volume */ + struct snd_kcontrol *ctl_cvol; /* center volume */ + + spinlock_t reg_lock; + + struct gameport *gameport; +}; + +int snd_trident_create(struct snd_card *card, + struct pci_dev *pci, + int pcm_streams, + int pcm_spdif_device, + int max_wavetable_size, + struct snd_trident ** rtrident); +int snd_trident_create_gameport(struct snd_trident *trident); + +int snd_trident_pcm(struct snd_trident * trident, int device, struct snd_pcm **rpcm); +int snd_trident_foldback_pcm(struct snd_trident * trident, int device, struct snd_pcm **rpcm); +int snd_trident_spdif_pcm(struct snd_trident * trident, int device, struct snd_pcm **rpcm); +int snd_trident_attach_synthesizer(struct snd_trident * trident); +struct snd_trident_voice *snd_trident_alloc_voice(struct snd_trident * trident, int type, + int client, int port); +void snd_trident_free_voice(struct snd_trident * trident, struct snd_trident_voice *voice); +void snd_trident_start_voice(struct snd_trident * trident, unsigned int voice); +void snd_trident_stop_voice(struct snd_trident * trident, unsigned int voice); +void snd_trident_write_voice_regs(struct snd_trident * trident, struct snd_trident_voice *voice); +extern const struct dev_pm_ops snd_trident_pm; + +/* TLB memory allocation */ +struct snd_util_memblk *snd_trident_alloc_pages(struct snd_trident *trident, + struct snd_pcm_substream *substream); +int snd_trident_free_pages(struct snd_trident *trident, struct snd_util_memblk *blk); +struct snd_util_memblk *snd_trident_synth_alloc(struct snd_trident *trident, unsigned int size); +int snd_trident_synth_free(struct snd_trident *trident, struct snd_util_memblk *blk); +int snd_trident_synth_copy_from_user(struct snd_trident *trident, struct snd_util_memblk *blk, + int offset, const char __user *data, int size); + +#endif /* __SOUND_TRIDENT_H */ diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 61d3c0e8d4ce..94011dcae731 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -41,7 +41,7 @@ #include <sound/info.h> #include <sound/control.h> #include <sound/tlv.h> -#include <sound/trident.h> +#include "trident.h" #include <sound/asoundef.h> #include <asm/io.h> @@ -3920,9 +3920,10 @@ static void snd_trident_clear_voices(struct snd_trident * trident, unsigned shor } #ifdef CONFIG_PM -int snd_trident_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_trident_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_trident *trident = card->private_data; trident->in_suspend = 1; @@ -3936,13 +3937,14 @@ int snd_trident_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -int snd_trident_resume(struct pci_dev *pci) +static int snd_trident_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_trident *trident = card->private_data; pci_set_power_state(pci, PCI_D0); @@ -3979,4 +3981,6 @@ int snd_trident_resume(struct pci_dev *pci) trident->in_suspend = 0; return 0; } + +SIMPLE_DEV_PM_OPS(snd_trident_pm, snd_trident_suspend, snd_trident_resume); #endif /* CONFIG_PM */ diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c index f9779e23fe57..3102a579660b 100644 --- a/sound/pci/trident/trident_memory.c +++ b/sound/pci/trident/trident_memory.c @@ -29,7 +29,7 @@ #include <linux/mutex.h> #include <sound/core.h> -#include <sound/trident.h> +#include "trident.h" /* page arguments of these two macros are Trident page (4096 bytes), not like * aligned pages in others diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 75630408c6db..0eb7245dd362 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2242,9 +2242,10 @@ static int snd_via82xx_chip_init(struct via82xx *chip) /* * power management */ -static int snd_via82xx_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_via82xx_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct via82xx *chip = card->private_data; int i; @@ -2265,13 +2266,14 @@ static int snd_via82xx_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_via82xx_resume(struct pci_dev *pci) +static int snd_via82xx_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct via82xx *chip = card->private_data; int i; @@ -2306,6 +2308,11 @@ static int snd_via82xx_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(snd_via82xx_pm, snd_via82xx_suspend, snd_via82xx_resume); +#define SND_VIA82XX_PM_OPS &snd_via82xx_pm +#else +#define SND_VIA82XX_PM_OPS NULL #endif /* CONFIG_PM */ static int snd_via82xx_free(struct via82xx *chip) @@ -2619,26 +2626,14 @@ static void __devexit snd_via82xx_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver via82xx_driver = { .name = KBUILD_MODNAME, .id_table = snd_via82xx_ids, .probe = snd_via82xx_probe, .remove = __devexit_p(snd_via82xx_remove), -#ifdef CONFIG_PM - .suspend = snd_via82xx_suspend, - .resume = snd_via82xx_resume, -#endif + .driver = { + .pm = SND_VIA82XX_PM_OPS, + }, }; -static int __init alsa_card_via82xx_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_via82xx_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_via82xx_init) -module_exit(alsa_card_via82xx_exit) +module_pci_driver(via82xx_driver); diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 5efcbcac506a..e886bc16999d 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -1023,9 +1023,10 @@ static int snd_via82xx_chip_init(struct via82xx_modem *chip) /* * power management */ -static int snd_via82xx_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_via82xx_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct via82xx_modem *chip = card->private_data; int i; @@ -1039,13 +1040,14 @@ static int snd_via82xx_suspend(struct pci_dev *pci, pm_message_t state) pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -static int snd_via82xx_resume(struct pci_dev *pci) +static int snd_via82xx_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct via82xx_modem *chip = card->private_data; int i; @@ -1069,6 +1071,11 @@ static int snd_via82xx_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +static SIMPLE_DEV_PM_OPS(snd_via82xx_pm, snd_via82xx_suspend, snd_via82xx_resume); +#define SND_VIA82XX_PM_OPS &snd_via82xx_pm +#else +#define SND_VIA82XX_PM_OPS NULL #endif /* CONFIG_PM */ static int snd_via82xx_free(struct via82xx_modem *chip) @@ -1223,26 +1230,14 @@ static void __devexit snd_via82xx_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver via82xx_modem_driver = { .name = KBUILD_MODNAME, .id_table = snd_via82xx_modem_ids, .probe = snd_via82xx_probe, .remove = __devexit_p(snd_via82xx_remove), -#ifdef CONFIG_PM - .suspend = snd_via82xx_suspend, - .resume = snd_via82xx_resume, -#endif + .driver = { + .pm = SND_VIA82XX_PM_OPS, + }, }; -static int __init alsa_card_via82xx_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_via82xx_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_via82xx_init) -module_exit(alsa_card_via82xx_exit) +module_pci_driver(via82xx_modem_driver); diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index 6a534bfe1274..b89e7a86e9d8 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -258,22 +258,24 @@ static void __devexit snd_vx222_remove(struct pci_dev *pci) } #ifdef CONFIG_PM -static int snd_vx222_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_vx222_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_vx222 *vx = card->private_data; int err; - err = snd_vx_suspend(&vx->core, state); + err = snd_vx_suspend(&vx->core); pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return err; } -static int snd_vx222_resume(struct pci_dev *pci) +static int snd_vx222_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_vx222 *vx = card->private_data; pci_set_power_state(pci, PCI_D0); @@ -287,28 +289,21 @@ static int snd_vx222_resume(struct pci_dev *pci) pci_set_master(pci); return snd_vx_resume(&vx->core); } + +static SIMPLE_DEV_PM_OPS(snd_vx222_pm, snd_vx222_suspend, snd_vx222_resume); +#define SND_VX222_PM_OPS &snd_vx222_pm +#else +#define SND_VX222_PM_OPS NULL #endif -static struct pci_driver driver = { +static struct pci_driver vx222_driver = { .name = KBUILD_MODNAME, .id_table = snd_vx222_ids, .probe = snd_vx222_probe, .remove = __devexit_p(snd_vx222_remove), -#ifdef CONFIG_PM - .suspend = snd_vx222_suspend, - .resume = snd_vx222_resume, -#endif + .driver = { + .pm = SND_VX222_PM_OPS, + }, }; -static int __init alsa_card_vx222_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_vx222_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_vx222_init) -module_exit(alsa_card_vx222_exit) +module_pci_driver(vx222_driver); diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 94ab728f5ca8..4810356b97ba 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -24,7 +24,7 @@ #include <linux/time.h> #include <linux/module.h> #include <sound/core.h> -#include <sound/ymfpci.h> +#include "ymfpci.h" #include <sound/mpu401.h> #include <sound/opl3.h> #include <sound/initval.h> @@ -350,26 +350,16 @@ static void __devexit snd_card_ymfpci_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_driver driver = { +static struct pci_driver ymfpci_driver = { .name = KBUILD_MODNAME, .id_table = snd_ymfpci_ids, .probe = snd_card_ymfpci_probe, .remove = __devexit_p(snd_card_ymfpci_remove), #ifdef CONFIG_PM - .suspend = snd_ymfpci_suspend, - .resume = snd_ymfpci_resume, + .driver = { + .pm = &snd_ymfpci_pm, + }, #endif }; -static int __init alsa_card_ymfpci_init(void) -{ - return pci_register_driver(&driver); -} - -static void __exit alsa_card_ymfpci_exit(void) -{ - pci_unregister_driver(&driver); -} - -module_init(alsa_card_ymfpci_init) -module_exit(alsa_card_ymfpci_exit) +module_pci_driver(ymfpci_driver); diff --git a/sound/pci/ymfpci/ymfpci.h b/sound/pci/ymfpci/ymfpci.h new file mode 100644 index 000000000000..bddc4052286b --- /dev/null +++ b/sound/pci/ymfpci/ymfpci.h @@ -0,0 +1,389 @@ +#ifndef __SOUND_YMFPCI_H +#define __SOUND_YMFPCI_H + +/* + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> + * Definitions for Yahama YMF724/740/744/754 chips + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <sound/pcm.h> +#include <sound/rawmidi.h> +#include <sound/ac97_codec.h> +#include <sound/timer.h> +#include <linux/gameport.h> + +/* + * Direct registers + */ + +#define YMFREG(chip, reg) (chip->port + YDSXGR_##reg) + +#define YDSXGR_INTFLAG 0x0004 +#define YDSXGR_ACTIVITY 0x0006 +#define YDSXGR_GLOBALCTRL 0x0008 +#define YDSXGR_ZVCTRL 0x000A +#define YDSXGR_TIMERCTRL 0x0010 +#define YDSXGR_TIMERCOUNT 0x0012 +#define YDSXGR_SPDIFOUTCTRL 0x0018 +#define YDSXGR_SPDIFOUTSTATUS 0x001C +#define YDSXGR_EEPROMCTRL 0x0020 +#define YDSXGR_SPDIFINCTRL 0x0034 +#define YDSXGR_SPDIFINSTATUS 0x0038 +#define YDSXGR_DSPPROGRAMDL 0x0048 +#define YDSXGR_DLCNTRL 0x004C +#define YDSXGR_GPIOININTFLAG 0x0050 +#define YDSXGR_GPIOININTENABLE 0x0052 +#define YDSXGR_GPIOINSTATUS 0x0054 +#define YDSXGR_GPIOOUTCTRL 0x0056 +#define YDSXGR_GPIOFUNCENABLE 0x0058 +#define YDSXGR_GPIOTYPECONFIG 0x005A +#define YDSXGR_AC97CMDDATA 0x0060 +#define YDSXGR_AC97CMDADR 0x0062 +#define YDSXGR_PRISTATUSDATA 0x0064 +#define YDSXGR_PRISTATUSADR 0x0066 +#define YDSXGR_SECSTATUSDATA 0x0068 +#define YDSXGR_SECSTATUSADR 0x006A +#define YDSXGR_SECCONFIG 0x0070 +#define YDSXGR_LEGACYOUTVOL 0x0080 +#define YDSXGR_LEGACYOUTVOLL 0x0080 +#define YDSXGR_LEGACYOUTVOLR 0x0082 +#define YDSXGR_NATIVEDACOUTVOL 0x0084 +#define YDSXGR_NATIVEDACOUTVOLL 0x0084 +#define YDSXGR_NATIVEDACOUTVOLR 0x0086 +#define YDSXGR_ZVOUTVOL 0x0088 +#define YDSXGR_ZVOUTVOLL 0x0088 +#define YDSXGR_ZVOUTVOLR 0x008A +#define YDSXGR_SECADCOUTVOL 0x008C +#define YDSXGR_SECADCOUTVOLL 0x008C +#define YDSXGR_SECADCOUTVOLR 0x008E +#define YDSXGR_PRIADCOUTVOL 0x0090 +#define YDSXGR_PRIADCOUTVOLL 0x0090 +#define YDSXGR_PRIADCOUTVOLR 0x0092 +#define YDSXGR_LEGACYLOOPVOL 0x0094 +#define YDSXGR_LEGACYLOOPVOLL 0x0094 +#define YDSXGR_LEGACYLOOPVOLR 0x0096 +#define YDSXGR_NATIVEDACLOOPVOL 0x0098 +#define YDSXGR_NATIVEDACLOOPVOLL 0x0098 +#define YDSXGR_NATIVEDACLOOPVOLR 0x009A +#define YDSXGR_ZVLOOPVOL 0x009C +#define YDSXGR_ZVLOOPVOLL 0x009E +#define YDSXGR_ZVLOOPVOLR 0x009E +#define YDSXGR_SECADCLOOPVOL 0x00A0 +#define YDSXGR_SECADCLOOPVOLL 0x00A0 +#define YDSXGR_SECADCLOOPVOLR 0x00A2 +#define YDSXGR_PRIADCLOOPVOL 0x00A4 +#define YDSXGR_PRIADCLOOPVOLL 0x00A4 +#define YDSXGR_PRIADCLOOPVOLR 0x00A6 +#define YDSXGR_NATIVEADCINVOL 0x00A8 +#define YDSXGR_NATIVEADCINVOLL 0x00A8 +#define YDSXGR_NATIVEADCINVOLR 0x00AA +#define YDSXGR_NATIVEDACINVOL 0x00AC +#define YDSXGR_NATIVEDACINVOLL 0x00AC +#define YDSXGR_NATIVEDACINVOLR 0x00AE +#define YDSXGR_BUF441OUTVOL 0x00B0 +#define YDSXGR_BUF441OUTVOLL 0x00B0 +#define YDSXGR_BUF441OUTVOLR 0x00B2 +#define YDSXGR_BUF441LOOPVOL 0x00B4 +#define YDSXGR_BUF441LOOPVOLL 0x00B4 +#define YDSXGR_BUF441LOOPVOLR 0x00B6 +#define YDSXGR_SPDIFOUTVOL 0x00B8 +#define YDSXGR_SPDIFOUTVOLL 0x00B8 +#define YDSXGR_SPDIFOUTVOLR 0x00BA +#define YDSXGR_SPDIFLOOPVOL 0x00BC +#define YDSXGR_SPDIFLOOPVOLL 0x00BC +#define YDSXGR_SPDIFLOOPVOLR 0x00BE +#define YDSXGR_ADCSLOTSR 0x00C0 +#define YDSXGR_RECSLOTSR 0x00C4 +#define YDSXGR_ADCFORMAT 0x00C8 +#define YDSXGR_RECFORMAT 0x00CC +#define YDSXGR_P44SLOTSR 0x00D0 +#define YDSXGR_STATUS 0x0100 +#define YDSXGR_CTRLSELECT 0x0104 +#define YDSXGR_MODE 0x0108 +#define YDSXGR_SAMPLECOUNT 0x010C +#define YDSXGR_NUMOFSAMPLES 0x0110 +#define YDSXGR_CONFIG 0x0114 +#define YDSXGR_PLAYCTRLSIZE 0x0140 +#define YDSXGR_RECCTRLSIZE 0x0144 +#define YDSXGR_EFFCTRLSIZE 0x0148 +#define YDSXGR_WORKSIZE 0x014C +#define YDSXGR_MAPOFREC 0x0150 +#define YDSXGR_MAPOFEFFECT 0x0154 +#define YDSXGR_PLAYCTRLBASE 0x0158 +#define YDSXGR_RECCTRLBASE 0x015C +#define YDSXGR_EFFCTRLBASE 0x0160 +#define YDSXGR_WORKBASE 0x0164 +#define YDSXGR_DSPINSTRAM 0x1000 +#define YDSXGR_CTRLINSTRAM 0x4000 + +#define YDSXG_AC97READCMD 0x8000 +#define YDSXG_AC97WRITECMD 0x0000 + +#define PCIR_DSXG_LEGACY 0x40 +#define PCIR_DSXG_ELEGACY 0x42 +#define PCIR_DSXG_CTRL 0x48 +#define PCIR_DSXG_PWRCTRL1 0x4a +#define PCIR_DSXG_PWRCTRL2 0x4e +#define PCIR_DSXG_FMBASE 0x60 +#define PCIR_DSXG_SBBASE 0x62 +#define PCIR_DSXG_MPU401BASE 0x64 +#define PCIR_DSXG_JOYBASE 0x66 + +#define YDSXG_DSPLENGTH 0x0080 +#define YDSXG_CTRLLENGTH 0x3000 + +#define YDSXG_DEFAULT_WORK_SIZE 0x0400 + +#define YDSXG_PLAYBACK_VOICES 64 +#define YDSXG_CAPTURE_VOICES 2 +#define YDSXG_EFFECT_VOICES 5 + +#define YMFPCI_LEGACY_SBEN (1 << 0) /* soundblaster enable */ +#define YMFPCI_LEGACY_FMEN (1 << 1) /* OPL3 enable */ +#define YMFPCI_LEGACY_JPEN (1 << 2) /* joystick enable */ +#define YMFPCI_LEGACY_MEN (1 << 3) /* MPU401 enable */ +#define YMFPCI_LEGACY_MIEN (1 << 4) /* MPU RX irq enable */ +#define YMFPCI_LEGACY_IOBITS (1 << 5) /* i/o bits range, 0 = 16bit, 1 =10bit */ +#define YMFPCI_LEGACY_SDMA (3 << 6) /* SB DMA select */ +#define YMFPCI_LEGACY_SBIRQ (7 << 8) /* SB IRQ select */ +#define YMFPCI_LEGACY_MPUIRQ (7 << 11) /* MPU IRQ select */ +#define YMFPCI_LEGACY_SIEN (1 << 14) /* serialized IRQ */ +#define YMFPCI_LEGACY_LAD (1 << 15) /* legacy audio disable */ + +#define YMFPCI_LEGACY2_FMIO (3 << 0) /* OPL3 i/o address (724/740) */ +#define YMFPCI_LEGACY2_SBIO (3 << 2) /* SB i/o address (724/740) */ +#define YMFPCI_LEGACY2_MPUIO (3 << 4) /* MPU401 i/o address (724/740) */ +#define YMFPCI_LEGACY2_JSIO (3 << 6) /* joystick i/o address (724/740) */ +#define YMFPCI_LEGACY2_MAIM (1 << 8) /* MPU401 ack intr mask */ +#define YMFPCI_LEGACY2_SMOD (3 << 11) /* SB DMA mode */ +#define YMFPCI_LEGACY2_SBVER (3 << 13) /* SB version select */ +#define YMFPCI_LEGACY2_IMOD (1 << 15) /* legacy IRQ mode */ +/* SIEN:IMOD 0:0 = legacy irq, 0:1 = INTA, 1:0 = serialized IRQ */ + +#if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE)) +#define SUPPORT_JOYSTICK +#endif + +/* + * + */ + +struct snd_ymfpci_playback_bank { + u32 format; + u32 loop_default; + u32 base; /* 32-bit address */ + u32 loop_start; /* 32-bit offset */ + u32 loop_end; /* 32-bit offset */ + u32 loop_frac; /* 8-bit fraction - loop_start */ + u32 delta_end; /* pitch delta end */ + u32 lpfK_end; + u32 eg_gain_end; + u32 left_gain_end; + u32 right_gain_end; + u32 eff1_gain_end; + u32 eff2_gain_end; + u32 eff3_gain_end; + u32 lpfQ; + u32 status; + u32 num_of_frames; + u32 loop_count; + u32 start; + u32 start_frac; + u32 delta; + u32 lpfK; + u32 eg_gain; + u32 left_gain; + u32 right_gain; + u32 eff1_gain; + u32 eff2_gain; + u32 eff3_gain; + u32 lpfD1; + u32 lpfD2; + }; + +struct snd_ymfpci_capture_bank { + u32 base; /* 32-bit address */ + u32 loop_end; /* 32-bit offset */ + u32 start; /* 32-bit offset */ + u32 num_of_loops; /* counter */ +}; + +struct snd_ymfpci_effect_bank { + u32 base; /* 32-bit address */ + u32 loop_end; /* 32-bit offset */ + u32 start; /* 32-bit offset */ + u32 temp; +}; + +struct snd_ymfpci_pcm; +struct snd_ymfpci; + +enum snd_ymfpci_voice_type { + YMFPCI_PCM, + YMFPCI_SYNTH, + YMFPCI_MIDI +}; + +struct snd_ymfpci_voice { + struct snd_ymfpci *chip; + int number; + unsigned int use: 1, + pcm: 1, + synth: 1, + midi: 1; + struct snd_ymfpci_playback_bank *bank; + dma_addr_t bank_addr; + void (*interrupt)(struct snd_ymfpci *chip, struct snd_ymfpci_voice *voice); + struct snd_ymfpci_pcm *ypcm; +}; + +enum snd_ymfpci_pcm_type { + PLAYBACK_VOICE, + CAPTURE_REC, + CAPTURE_AC97, + EFFECT_DRY_LEFT, + EFFECT_DRY_RIGHT, + EFFECT_EFF1, + EFFECT_EFF2, + EFFECT_EFF3 +}; + +struct snd_ymfpci_pcm { + struct snd_ymfpci *chip; + enum snd_ymfpci_pcm_type type; + struct snd_pcm_substream *substream; + struct snd_ymfpci_voice *voices[2]; /* playback only */ + unsigned int running: 1, + use_441_slot: 1, + output_front: 1, + output_rear: 1, + swap_rear: 1; + unsigned int update_pcm_vol; + u32 period_size; /* cached from runtime->period_size */ + u32 buffer_size; /* cached from runtime->buffer_size */ + u32 period_pos; + u32 last_pos; + u32 capture_bank_number; + u32 shift; +}; + +struct snd_ymfpci { + int irq; + + unsigned int device_id; /* PCI device ID */ + unsigned char rev; /* PCI revision */ + unsigned long reg_area_phys; + void __iomem *reg_area_virt; + struct resource *res_reg_area; + struct resource *fm_res; + struct resource *mpu_res; + + unsigned short old_legacy_ctrl; +#ifdef SUPPORT_JOYSTICK + struct gameport *gameport; +#endif + + struct snd_dma_buffer work_ptr; + + unsigned int bank_size_playback; + unsigned int bank_size_capture; + unsigned int bank_size_effect; + unsigned int work_size; + + void *bank_base_playback; + void *bank_base_capture; + void *bank_base_effect; + void *work_base; + dma_addr_t bank_base_playback_addr; + dma_addr_t bank_base_capture_addr; + dma_addr_t bank_base_effect_addr; + dma_addr_t work_base_addr; + struct snd_dma_buffer ac3_tmp_base; + + u32 *ctrl_playback; + struct snd_ymfpci_playback_bank *bank_playback[YDSXG_PLAYBACK_VOICES][2]; + struct snd_ymfpci_capture_bank *bank_capture[YDSXG_CAPTURE_VOICES][2]; + struct snd_ymfpci_effect_bank *bank_effect[YDSXG_EFFECT_VOICES][2]; + + int start_count; + + u32 active_bank; + struct snd_ymfpci_voice voices[64]; + int src441_used; + + struct snd_ac97_bus *ac97_bus; + struct snd_ac97 *ac97; + struct snd_rawmidi *rawmidi; + struct snd_timer *timer; + unsigned int timer_ticks; + + struct pci_dev *pci; + struct snd_card *card; + struct snd_pcm *pcm; + struct snd_pcm *pcm2; + struct snd_pcm *pcm_spdif; + struct snd_pcm *pcm_4ch; + struct snd_pcm_substream *capture_substream[YDSXG_CAPTURE_VOICES]; + struct snd_pcm_substream *effect_substream[YDSXG_EFFECT_VOICES]; + struct snd_kcontrol *ctl_vol_recsrc; + struct snd_kcontrol *ctl_vol_adcrec; + struct snd_kcontrol *ctl_vol_spdifrec; + unsigned short spdif_bits, spdif_pcm_bits; + struct snd_kcontrol *spdif_pcm_ctl; + int mode_dup4ch; + int rear_opened; + int spdif_opened; + struct snd_ymfpci_pcm_mixer { + u16 left; + u16 right; + struct snd_kcontrol *ctl; + } pcm_mixer[32]; + + spinlock_t reg_lock; + spinlock_t voice_lock; + wait_queue_head_t interrupt_sleep; + atomic_t interrupt_sleep_count; + struct snd_info_entry *proc_entry; + const struct firmware *dsp_microcode; + const struct firmware *controller_microcode; + +#ifdef CONFIG_PM + u32 *saved_regs; + u32 saved_ydsxgr_mode; + u16 saved_dsxg_legacy; + u16 saved_dsxg_elegacy; +#endif +}; + +int snd_ymfpci_create(struct snd_card *card, + struct pci_dev *pci, + unsigned short old_legacy_ctrl, + struct snd_ymfpci ** rcodec); +void snd_ymfpci_free_gameport(struct snd_ymfpci *chip); + +extern const struct dev_pm_ops snd_ymfpci_pm; + +int snd_ymfpci_pcm(struct snd_ymfpci *chip, int device, struct snd_pcm **rpcm); +int snd_ymfpci_pcm2(struct snd_ymfpci *chip, int device, struct snd_pcm **rpcm); +int snd_ymfpci_pcm_spdif(struct snd_ymfpci *chip, int device, struct snd_pcm **rpcm); +int snd_ymfpci_pcm_4ch(struct snd_ymfpci *chip, int device, struct snd_pcm **rpcm); +int snd_ymfpci_mixer(struct snd_ymfpci *chip, int rear_switch); +int snd_ymfpci_timer(struct snd_ymfpci *chip, int device); + +#endif /* __SOUND_YMFPCI_H */ diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index a8159b81e9c4..62b23635b754 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -33,7 +33,7 @@ #include <sound/control.h> #include <sound/info.h> #include <sound/tlv.h> -#include <sound/ymfpci.h> +#include "ymfpci.h" #include <sound/asoundef.h> #include <sound/mpu401.h> @@ -2302,9 +2302,10 @@ static int saved_regs_index[] = { }; #define YDSXGR_NUM_SAVED_REGS ARRAY_SIZE(saved_regs_index) -int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state) +static int snd_ymfpci_suspend(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_ymfpci *chip = card->private_data; unsigned int i; @@ -2326,13 +2327,14 @@ int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state) snd_ymfpci_disable_dsp(chip); pci_disable_device(pci); pci_save_state(pci); - pci_set_power_state(pci, pci_choose_state(pci, state)); + pci_set_power_state(pci, PCI_D3hot); return 0; } -int snd_ymfpci_resume(struct pci_dev *pci) +static int snd_ymfpci_resume(struct device *dev) { - struct snd_card *card = pci_get_drvdata(pci); + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_ymfpci *chip = card->private_data; unsigned int i; @@ -2370,6 +2372,8 @@ int snd_ymfpci_resume(struct pci_dev *pci) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } + +SIMPLE_DEV_PM_OPS(snd_ymfpci_pm, snd_ymfpci_suspend, snd_ymfpci_resume); #endif /* CONFIG_PM */ int __devinit snd_ymfpci_create(struct snd_card *card, diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 830839a874b6..f9b5229b2723 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -251,7 +251,7 @@ static int pdacf_suspend(struct pcmcia_device *link) snd_printdd(KERN_DEBUG "SUSPEND\n"); if (chip) { snd_printdd(KERN_DEBUG "snd_pdacf_suspend calling\n"); - snd_pdacf_suspend(chip, PMSG_SUSPEND); + snd_pdacf_suspend(chip); } return 0; diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.h b/sound/pcmcia/pdaudiocf/pdaudiocf.h index 6ce9ad700290..ea41e57d7179 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.h +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.h @@ -131,7 +131,7 @@ struct snd_pdacf *snd_pdacf_create(struct snd_card *card); int snd_pdacf_ak4117_create(struct snd_pdacf *pdacf); void snd_pdacf_powerdown(struct snd_pdacf *chip); #ifdef CONFIG_PM -int snd_pdacf_suspend(struct snd_pdacf *chip, pm_message_t state); +int snd_pdacf_suspend(struct snd_pdacf *chip); int snd_pdacf_resume(struct snd_pdacf *chip); #endif int snd_pdacf_pcm_new(struct snd_pdacf *chip); diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c index 9dce0bde5c05..ea0adfb984ad 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c @@ -262,7 +262,7 @@ void snd_pdacf_powerdown(struct snd_pdacf *chip) #ifdef CONFIG_PM -int snd_pdacf_suspend(struct snd_pdacf *chip, pm_message_t state) +int snd_pdacf_suspend(struct snd_pdacf *chip) { u16 val; diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 512f0b472375..8f9350475c7b 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -260,7 +260,7 @@ static int vxp_suspend(struct pcmcia_device *link) snd_printdd(KERN_DEBUG "SUSPEND\n"); if (chip) { snd_printdd(KERN_DEBUG "snd_vx_suspend calling\n"); - snd_vx_suspend(chip, PMSG_SUSPEND); + snd_vx_suspend(chip); } return 0; diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index 5a4e263b5b0f..f5ceb6f282de 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -144,19 +144,24 @@ static int __devexit snd_pmac_remove(struct platform_device *devptr) } #ifdef CONFIG_PM -static int snd_pmac_driver_suspend(struct platform_device *devptr, pm_message_t state) +static int snd_pmac_driver_suspend(struct device *dev) { - struct snd_card *card = platform_get_drvdata(devptr); + struct snd_card *card = dev_get_drvdata(dev); snd_pmac_suspend(card->private_data); return 0; } -static int snd_pmac_driver_resume(struct platform_device *devptr) +static int snd_pmac_driver_resume(struct device *dev) { - struct snd_card *card = platform_get_drvdata(devptr); + struct snd_card *card = dev_get_drvdata(dev); snd_pmac_resume(card->private_data); return 0; } + +static SIMPLE_DEV_PM_OPS(snd_pmac_pm, snd_pmac_driver_suspend, snd_pmac_driver_resume); +#define SND_PMAC_PM_OPS &snd_pmac_pm +#else +#define SND_PMAC_PM_OPS NULL #endif #define SND_PMAC_DRIVER "snd_powermac" @@ -164,12 +169,10 @@ static int snd_pmac_driver_resume(struct platform_device *devptr) static struct platform_driver snd_pmac_driver = { .probe = snd_pmac_probe, .remove = __devexit_p(snd_pmac_remove), -#ifdef CONFIG_PM - .suspend = snd_pmac_driver_suspend, - .resume = snd_pmac_driver_resume, -#endif .driver = { - .name = SND_PMAC_DRIVER + .name = SND_PMAC_DRIVER, + .owner = THIS_MODULE, + .pm = SND_PMAC_PM_OPS, }, }; diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 391a38ca58bc..d48b523207eb 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -654,7 +654,9 @@ static struct platform_driver snd_aica_driver = { .probe = snd_aica_probe, .remove = __devexit_p(snd_aica_remove), .driver = { - .name = SND_AICA_DRIVER}, + .name = SND_AICA_DRIVER, + .owner = THIS_MODULE, + }, }; static int __init aica_init(void) diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index b11f82b5718f..0a3394751ed2 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -433,12 +433,13 @@ probe_error: /* * "driver" definition */ -static struct platform_driver driver = { +static struct platform_driver sh_dac_driver = { .probe = snd_sh_dac_probe, .remove = snd_sh_dac_remove, .driver = { .name = "dac_audio", + .owner = THIS_MODULE, }, }; -module_platform_driver(driver); +module_platform_driver(sh_dac_driver); diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 91c985599d32..c5de0a84566f 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -33,9 +33,9 @@ source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/blackfin/Kconfig" source "sound/soc/davinci/Kconfig" +source "sound/soc/dwc/Kconfig" source "sound/soc/ep93xx/Kconfig" source "sound/soc/fsl/Kconfig" -source "sound/soc/imx/Kconfig" source "sound/soc/jz4740/Kconfig" source "sound/soc/nuc900/Kconfig" source "sound/soc/omap/Kconfig" @@ -48,9 +48,13 @@ source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" +source "sound/soc/ux500/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" +# generic frame-work +source "sound/soc/generic/Kconfig" + endif # SND_SOC diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 2feaf376e94b..00a555a743b6 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -6,13 +6,14 @@ obj-$(CONFIG_SND_SOC_DMAENGINE_PCM) += snd-soc-dmaengine-pcm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ +obj-$(CONFIG_SND_SOC) += generic/ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ obj-$(CONFIG_SND_SOC) += blackfin/ obj-$(CONFIG_SND_SOC) += davinci/ +obj-$(CONFIG_SND_SOC) += dwc/ obj-$(CONFIG_SND_SOC) += ep93xx/ obj-$(CONFIG_SND_SOC) += fsl/ -obj-$(CONFIG_SND_SOC) += imx/ obj-$(CONFIG_SND_SOC) += jz4740/ obj-$(CONFIG_SND_SOC) += mid-x86/ obj-$(CONFIG_SND_SOC) += mxs/ @@ -25,3 +26,4 @@ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ +obj-$(CONFIG_SND_SOC) += ux500/ diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 9f6bc55fc399..16b88f5c26e2 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -1,7 +1,8 @@ config SND_BF5XX_I2S - tristate "SoC I2S Audio for the ADI BF5xx chip" + tristate "SoC I2S Audio for the ADI Blackfin chip" depends on BLACKFIN - select SND_BF5XX_SOC_SPORT + select SND_BF5XX_SOC_SPORT if !BF60x + select SND_BF6XX_SOC_SPORT if BF60x help Say Y or M if you want to add support for codecs attached to the Blackfin SPORT (synchronous serial ports) interface in I2S @@ -9,12 +10,14 @@ config SND_BF5XX_I2S You will also need to select the audio interfaces to support below. config SND_BF5XX_SOC_SSM2602 - tristate "SoC SSM2602 Audio support for BF52x ezkit" + tristate "SoC SSM2602 Audio Codec Add-On Card support" depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) - select SND_BF5XX_SOC_I2S + select SND_BF5XX_SOC_I2S if !BF60x + select SND_BF6XX_SOC_I2S if BF60x select SND_SOC_SSM2602 help - Say Y if you want to add support for SoC audio on BF527-EZKIT. + Say Y if you want to add support for the Analog Devices + SSM2602 Audio Codec Add-On Card. config SND_SOC_BFIN_EVAL_ADAU1701 tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards" @@ -162,9 +165,15 @@ config SND_BF5XX_SOC_AD1980 config SND_BF5XX_SOC_SPORT tristate +config SND_BF6XX_SOC_SPORT + tristate + config SND_BF5XX_SOC_I2S tristate +config SND_BF6XX_SOC_I2S + tristate + config SND_BF5XX_SOC_TDM tristate @@ -173,7 +182,7 @@ config SND_BF5XX_SOC_AC97 config SND_BF5XX_SPORT_NUM int "Set a SPORT for Sound chip" - depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM) + depends on (SND_BF5XX_SOC_SPORT || SND_BF6XX_SOC_SPORT) range 0 3 if BF54x range 0 1 if !BF54x default 0 diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 1bf86ccaa8de..6fea1f4cbee2 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -3,16 +3,20 @@ snd-bf5xx-ac97-objs := bf5xx-ac97-pcm.o snd-bf5xx-i2s-objs := bf5xx-i2s-pcm.o snd-bf5xx-tdm-objs := bf5xx-tdm-pcm.o snd-soc-bf5xx-sport-objs := bf5xx-sport.o +snd-soc-bf6xx-sport-objs := bf6xx-sport.o snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o +snd-soc-bf6xx-i2s-objs := bf6xx-i2s.o snd-soc-bf5xx-tdm-objs := bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_I2S) += snd-bf5xx-i2s.o obj-$(CONFIG_SND_BF5XX_TDM) += snd-bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o +obj-$(CONFIG_SND_BF6XX_SOC_SPORT) += snd-soc-bf6xx-sport.o obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o +obj-$(CONFIG_SND_BF6XX_SOC_I2S) += snd-soc-bf6xx-i2s.o obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o # Blackfin Machine Support diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index b39ad356b92b..7dbeef1099b4 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -44,16 +44,8 @@ static struct snd_soc_card bf5xx_ssm2602; -static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int bf5xx_ssm2602_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - unsigned int clk = 0; - int ret = 0; - - pr_debug("%s rate %d format %x\n", __func__, params_rate(params), - params_format(params)); /* * If you are using a crystal source which frequency is not 12MHz * then modify the below case statement with frequency of the crystal. @@ -61,31 +53,10 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream, * If you are using the SPORT to generate clocking then this is * where to do it. */ - - switch (params_rate(params)) { - case 8000: - case 16000: - case 48000: - case 96000: - case 11025: - case 22050: - case 44100: - clk = 12000000; - break; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, SSM2602_SYSCLK, clk, + return snd_soc_dai_set_sysclk(rtd->codec_dai, SSM2602_SYSCLK, 12000000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - return 0; } -static struct snd_soc_ops bf5xx_ssm2602_ops = { - .hw_params = bf5xx_ssm2602_hw_params, -}; - /* CODEC is master for BCLK and LRC in this configuration. */ #define BF5XX_SSM2602_DAIFMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \ SND_SOC_DAIFMT_CBM_CFM) @@ -98,7 +69,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { .codec_dai_name = "ssm2602-hifi", .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ssm2602.0-001b", - .ops = &bf5xx_ssm2602_ops, + .init = bf5xx_ssm2602_dai_init, .dai_fmt = BF5XX_SSM2602_DAIFMT, }, { @@ -108,7 +79,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { .codec_dai_name = "ssm2602-hifi", .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ssm2602.0-001b", - .ops = &bf5xx_ssm2602_ops, + .init = bf5xx_ssm2602_dai_init, .dai_fmt = BF5XX_SSM2602_DAIFMT, }, }; diff --git a/sound/soc/blackfin/bf6xx-i2s.c b/sound/soc/blackfin/bf6xx-i2s.c new file mode 100644 index 000000000000..c3c2466d3a42 --- /dev/null +++ b/sound/soc/blackfin/bf6xx-i2s.c @@ -0,0 +1,234 @@ +/* + * bf6xx-i2s.c - Analog Devices BF6XX i2s interface driver + * + * Copyright (c) 2012 Analog Devices Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#include <linux/device.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "bf6xx-sport.h" + +struct sport_params param; + +static int bfin_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct sport_device *sport = snd_soc_dai_get_drvdata(cpu_dai); + struct device *dev = &sport->pdev->dev; + int ret = 0; + + param.spctl &= ~(SPORT_CTL_OPMODE | SPORT_CTL_CKRE | SPORT_CTL_FSR + | SPORT_CTL_LFS | SPORT_CTL_LAFS); + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + param.spctl |= SPORT_CTL_OPMODE | SPORT_CTL_CKRE + | SPORT_CTL_LFS; + break; + case SND_SOC_DAIFMT_DSP_A: + param.spctl |= SPORT_CTL_FSR; + break; + case SND_SOC_DAIFMT_LEFT_J: + param.spctl |= SPORT_CTL_OPMODE | SPORT_CTL_LFS + | SPORT_CTL_LAFS; + break; + default: + dev_err(dev, "%s: Unknown DAI format type\n", __func__); + ret = -EINVAL; + break; + } + + param.spctl &= ~(SPORT_CTL_ICLK | SPORT_CTL_IFS); + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + ret = -EINVAL; + break; + default: + dev_err(dev, "%s: Unknown DAI master type\n", __func__); + ret = -EINVAL; + break; + } + + return ret; +} + +static int bfin_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct sport_device *sport = snd_soc_dai_get_drvdata(dai); + struct device *dev = &sport->pdev->dev; + int ret = 0; + + param.spctl &= ~SPORT_CTL_SLEN; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + param.spctl |= 0x70; + sport->wdsize = 1; + case SNDRV_PCM_FORMAT_S16_LE: + param.spctl |= 0xf0; + sport->wdsize = 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + param.spctl |= 0x170; + sport->wdsize = 3; + break; + case SNDRV_PCM_FORMAT_S32_LE: + param.spctl |= 0x1f0; + sport->wdsize = 4; + break; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = sport_set_tx_params(sport, ¶m); + if (ret) { + dev_err(dev, "SPORT tx is busy!\n"); + return ret; + } + } else { + ret = sport_set_rx_params(sport, ¶m); + if (ret) { + dev_err(dev, "SPORT rx is busy!\n"); + return ret; + } + } + return 0; +} + +#ifdef CONFIG_PM +static int bfin_i2s_suspend(struct snd_soc_dai *dai) +{ + struct sport_device *sport = snd_soc_dai_get_drvdata(dai); + + if (dai->capture_active) + sport_rx_stop(sport); + if (dai->playback_active) + sport_tx_stop(sport); + return 0; +} + +static int bfin_i2s_resume(struct snd_soc_dai *dai) +{ + struct sport_device *sport = snd_soc_dai_get_drvdata(dai); + struct device *dev = &sport->pdev->dev; + int ret; + + ret = sport_set_tx_params(sport, ¶m); + if (ret) { + dev_err(dev, "SPORT tx is busy!\n"); + return ret; + } + ret = sport_set_rx_params(sport, ¶m); + if (ret) { + dev_err(dev, "SPORT rx is busy!\n"); + return ret; + } + + return 0; +} + +#else +#define bfin_i2s_suspend NULL +#define bfin_i2s_resume NULL +#endif + +#define BFIN_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define BFIN_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops bfin_i2s_dai_ops = { + .hw_params = bfin_i2s_hw_params, + .set_fmt = bfin_i2s_set_dai_fmt, +}; + +static struct snd_soc_dai_driver bfin_i2s_dai = { + .suspend = bfin_i2s_suspend, + .resume = bfin_i2s_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = BFIN_I2S_RATES, + .formats = BFIN_I2S_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = BFIN_I2S_RATES, + .formats = BFIN_I2S_FORMATS, + }, + .ops = &bfin_i2s_dai_ops, +}; + +static int __devinit bfin_i2s_probe(struct platform_device *pdev) +{ + struct sport_device *sport; + struct device *dev = &pdev->dev; + int ret; + + sport = sport_create(pdev); + if (!sport) + return -ENODEV; + + /* register with the ASoC layers */ + ret = snd_soc_register_dai(dev, &bfin_i2s_dai); + if (ret) { + dev_err(dev, "Failed to register DAI: %d\n", ret); + sport_delete(sport); + return ret; + } + platform_set_drvdata(pdev, sport); + + return 0; +} + +static int __devexit bfin_i2s_remove(struct platform_device *pdev) +{ + struct sport_device *sport = platform_get_drvdata(pdev); + + snd_soc_unregister_dai(&pdev->dev); + sport_delete(sport); + + return 0; +} + +static struct platform_driver bfin_i2s_driver = { + .probe = bfin_i2s_probe, + .remove = __devexit_p(bfin_i2s_remove), + .driver = { + .name = "bfin-i2s", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(bfin_i2s_driver); + +MODULE_DESCRIPTION("Analog Devices BF6XX i2s interface driver"); +MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux@gmail.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c new file mode 100644 index 000000000000..318c5ba5360f --- /dev/null +++ b/sound/soc/blackfin/bf6xx-sport.c @@ -0,0 +1,422 @@ +/* + * bf6xx_sport.c Analog Devices BF6XX SPORT driver + * + * Copyright (c) 2012 Analog Devices Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#include <linux/device.h> +#include <linux/dma-mapping.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> + +#include <asm/blackfin.h> +#include <asm/dma.h> +#include <asm/portmux.h> + +#include "bf6xx-sport.h" + +int sport_set_tx_params(struct sport_device *sport, + struct sport_params *params) +{ + if (sport->tx_regs->spctl & SPORT_CTL_SPENPRI) + return -EBUSY; + sport->tx_regs->spctl = params->spctl | SPORT_CTL_SPTRAN; + sport->tx_regs->div = params->div; + SSYNC(); + return 0; +} +EXPORT_SYMBOL(sport_set_tx_params); + +int sport_set_rx_params(struct sport_device *sport, + struct sport_params *params) +{ + if (sport->rx_regs->spctl & SPORT_CTL_SPENPRI) + return -EBUSY; + sport->rx_regs->spctl = params->spctl & ~SPORT_CTL_SPTRAN; + sport->rx_regs->div = params->div; + SSYNC(); + return 0; +} +EXPORT_SYMBOL(sport_set_rx_params); + +static int compute_wdsize(size_t wdsize) +{ + switch (wdsize) { + case 1: + return WDSIZE_8 | PSIZE_8; + case 2: + return WDSIZE_16 | PSIZE_16; + default: + return WDSIZE_32 | PSIZE_32; + } +} + +void sport_tx_start(struct sport_device *sport) +{ + set_dma_next_desc_addr(sport->tx_dma_chan, sport->tx_desc); + set_dma_config(sport->tx_dma_chan, DMAFLOW_LIST | DI_EN + | compute_wdsize(sport->wdsize) | NDSIZE_6); + enable_dma(sport->tx_dma_chan); + sport->tx_regs->spctl |= SPORT_CTL_SPENPRI; + SSYNC(); +} +EXPORT_SYMBOL(sport_tx_start); + +void sport_rx_start(struct sport_device *sport) +{ + set_dma_next_desc_addr(sport->rx_dma_chan, sport->rx_desc); + set_dma_config(sport->rx_dma_chan, DMAFLOW_LIST | DI_EN | WNR + | compute_wdsize(sport->wdsize) | NDSIZE_6); + enable_dma(sport->rx_dma_chan); + sport->rx_regs->spctl |= SPORT_CTL_SPENPRI; + SSYNC(); +} +EXPORT_SYMBOL(sport_rx_start); + +void sport_tx_stop(struct sport_device *sport) +{ + sport->tx_regs->spctl &= ~SPORT_CTL_SPENPRI; + SSYNC(); + disable_dma(sport->tx_dma_chan); +} +EXPORT_SYMBOL(sport_tx_stop); + +void sport_rx_stop(struct sport_device *sport) +{ + sport->rx_regs->spctl &= ~SPORT_CTL_SPENPRI; + SSYNC(); + disable_dma(sport->rx_dma_chan); +} +EXPORT_SYMBOL(sport_rx_stop); + +void sport_set_tx_callback(struct sport_device *sport, + void (*tx_callback)(void *), void *tx_data) +{ + sport->tx_callback = tx_callback; + sport->tx_data = tx_data; +} +EXPORT_SYMBOL(sport_set_tx_callback); + +void sport_set_rx_callback(struct sport_device *sport, + void (*rx_callback)(void *), void *rx_data) +{ + sport->rx_callback = rx_callback; + sport->rx_data = rx_data; +} +EXPORT_SYMBOL(sport_set_rx_callback); + +static void setup_desc(struct dmasg *desc, void *buf, int fragcount, + size_t fragsize, unsigned int cfg, + unsigned int count, size_t wdsize) +{ + + int i; + + for (i = 0; i < fragcount; ++i) { + desc[i].next_desc_addr = &(desc[i + 1]); + desc[i].start_addr = (unsigned long)buf + i*fragsize; + desc[i].cfg = cfg; + desc[i].x_count = count; + desc[i].x_modify = wdsize; + desc[i].y_count = 0; + desc[i].y_modify = 0; + } + + /* make circular */ + desc[fragcount-1].next_desc_addr = desc; +} + +int sport_config_tx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize) +{ + unsigned int count; + unsigned int cfg; + dma_addr_t addr; + + count = fragsize/sport->wdsize; + + if (sport->tx_desc) + dma_free_coherent(NULL, sport->tx_desc_size, + sport->tx_desc, 0); + + sport->tx_desc = dma_alloc_coherent(NULL, + fragcount * sizeof(struct dmasg), &addr, 0); + sport->tx_desc_size = fragcount * sizeof(struct dmasg); + if (!sport->tx_desc) + return -ENOMEM; + + sport->tx_buf = buf; + sport->tx_fragsize = fragsize; + sport->tx_frags = fragcount; + cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) | NDSIZE_6; + + setup_desc(sport->tx_desc, buf, fragcount, fragsize, + cfg|DMAEN, count, sport->wdsize); + + return 0; +} +EXPORT_SYMBOL(sport_config_tx_dma); + +int sport_config_rx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize) +{ + unsigned int count; + unsigned int cfg; + dma_addr_t addr; + + count = fragsize/sport->wdsize; + + if (sport->rx_desc) + dma_free_coherent(NULL, sport->rx_desc_size, + sport->rx_desc, 0); + + sport->rx_desc = dma_alloc_coherent(NULL, + fragcount * sizeof(struct dmasg), &addr, 0); + sport->rx_desc_size = fragcount * sizeof(struct dmasg); + if (!sport->rx_desc) + return -ENOMEM; + + sport->rx_buf = buf; + sport->rx_fragsize = fragsize; + sport->rx_frags = fragcount; + cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) + | WNR | NDSIZE_6; + + setup_desc(sport->rx_desc, buf, fragcount, fragsize, + cfg|DMAEN, count, sport->wdsize); + + return 0; +} +EXPORT_SYMBOL(sport_config_rx_dma); + +unsigned long sport_curr_offset_tx(struct sport_device *sport) +{ + unsigned long curr = get_dma_curr_addr(sport->tx_dma_chan); + + return (unsigned char *)curr - sport->tx_buf; +} +EXPORT_SYMBOL(sport_curr_offset_tx); + +unsigned long sport_curr_offset_rx(struct sport_device *sport) +{ + unsigned long curr = get_dma_curr_addr(sport->rx_dma_chan); + + return (unsigned char *)curr - sport->rx_buf; +} +EXPORT_SYMBOL(sport_curr_offset_rx); + +static irqreturn_t sport_tx_irq(int irq, void *dev_id) +{ + struct sport_device *sport = dev_id; + static unsigned long status; + + status = get_dma_curr_irqstat(sport->tx_dma_chan); + if (status & (DMA_DONE|DMA_ERR)) { + clear_dma_irqstat(sport->tx_dma_chan); + SSYNC(); + } + if (sport->tx_callback) + sport->tx_callback(sport->tx_data); + return IRQ_HANDLED; +} + +static irqreturn_t sport_rx_irq(int irq, void *dev_id) +{ + struct sport_device *sport = dev_id; + unsigned long status; + + status = get_dma_curr_irqstat(sport->rx_dma_chan); + if (status & (DMA_DONE|DMA_ERR)) { + clear_dma_irqstat(sport->rx_dma_chan); + SSYNC(); + } + if (sport->rx_callback) + sport->rx_callback(sport->rx_data); + return IRQ_HANDLED; +} + +static irqreturn_t sport_err_irq(int irq, void *dev_id) +{ + struct sport_device *sport = dev_id; + struct device *dev = &sport->pdev->dev; + + if (sport->tx_regs->spctl & SPORT_CTL_DERRPRI) + dev_err(dev, "sport error: TUVF\n"); + if (sport->rx_regs->spctl & SPORT_CTL_DERRPRI) + dev_err(dev, "sport error: ROVF\n"); + + return IRQ_HANDLED; +} + +static int sport_get_resource(struct sport_device *sport) +{ + struct platform_device *pdev = sport->pdev; + struct device *dev = &pdev->dev; + struct bfin_snd_platform_data *pdata = dev->platform_data; + struct resource *res; + + if (!pdata) { + dev_err(dev, "No platform data\n"); + return -ENODEV; + } + sport->pin_req = pdata->pin_req; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(dev, "No tx MEM resource\n"); + return -ENODEV; + } + sport->tx_regs = (struct sport_register *)res->start; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 1); + if (!res) { + dev_err(dev, "No rx MEM resource\n"); + return -ENODEV; + } + sport->rx_regs = (struct sport_register *)res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(dev, "No tx DMA resource\n"); + return -ENODEV; + } + sport->tx_dma_chan = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(dev, "No rx DMA resource\n"); + return -ENODEV; + } + sport->rx_dma_chan = res->start; + + res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!res) { + dev_err(dev, "No tx error irq resource\n"); + return -ENODEV; + } + sport->tx_err_irq = res->start; + + res = platform_get_resource(pdev, IORESOURCE_IRQ, 1); + if (!res) { + dev_err(dev, "No rx error irq resource\n"); + return -ENODEV; + } + sport->rx_err_irq = res->start; + + return 0; +} + +static int sport_request_resource(struct sport_device *sport) +{ + struct device *dev = &sport->pdev->dev; + int ret; + + ret = peripheral_request_list(sport->pin_req, "soc-audio"); + if (ret) { + dev_err(dev, "Unable to request sport pin\n"); + return ret; + } + + ret = request_dma(sport->tx_dma_chan, "SPORT TX Data"); + if (ret) { + dev_err(dev, "Unable to allocate DMA channel for sport tx\n"); + goto err_tx_dma; + } + set_dma_callback(sport->tx_dma_chan, sport_tx_irq, sport); + + ret = request_dma(sport->rx_dma_chan, "SPORT RX Data"); + if (ret) { + dev_err(dev, "Unable to allocate DMA channel for sport rx\n"); + goto err_rx_dma; + } + set_dma_callback(sport->rx_dma_chan, sport_rx_irq, sport); + + ret = request_irq(sport->tx_err_irq, sport_err_irq, + 0, "SPORT TX ERROR", sport); + if (ret) { + dev_err(dev, "Unable to allocate tx error IRQ for sport\n"); + goto err_tx_irq; + } + + ret = request_irq(sport->rx_err_irq, sport_err_irq, + 0, "SPORT RX ERROR", sport); + if (ret) { + dev_err(dev, "Unable to allocate rx error IRQ for sport\n"); + goto err_rx_irq; + } + + return 0; +err_rx_irq: + free_irq(sport->tx_err_irq, sport); +err_tx_irq: + free_dma(sport->rx_dma_chan); +err_rx_dma: + free_dma(sport->tx_dma_chan); +err_tx_dma: + peripheral_free_list(sport->pin_req); + return ret; +} + +static void sport_free_resource(struct sport_device *sport) +{ + free_irq(sport->rx_err_irq, sport); + free_irq(sport->tx_err_irq, sport); + free_dma(sport->rx_dma_chan); + free_dma(sport->tx_dma_chan); + peripheral_free_list(sport->pin_req); +} + +struct sport_device *sport_create(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct sport_device *sport; + int ret; + + sport = kzalloc(sizeof(*sport), GFP_KERNEL); + if (!sport) { + dev_err(dev, "Unable to allocate memory for sport device\n"); + return NULL; + } + sport->pdev = pdev; + + ret = sport_get_resource(sport); + if (ret) { + kfree(sport); + return NULL; + } + + ret = sport_request_resource(sport); + if (ret) { + kfree(sport); + return NULL; + } + + dev_dbg(dev, "SPORT create success\n"); + return sport; +} +EXPORT_SYMBOL(sport_create); + +void sport_delete(struct sport_device *sport) +{ + sport_free_resource(sport); +} +EXPORT_SYMBOL(sport_delete); + +MODULE_DESCRIPTION("Analog Devices BF6XX SPORT driver"); +MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux@gmail.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/blackfin/bf6xx-sport.h b/sound/soc/blackfin/bf6xx-sport.h new file mode 100644 index 000000000000..307d193cfcef --- /dev/null +++ b/sound/soc/blackfin/bf6xx-sport.h @@ -0,0 +1,82 @@ +/* + * bf6xx_sport - Analog Devices BF6XX SPORT driver + * + * Copyright (c) 2012 Analog Devices Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#ifndef _BF6XX_SPORT_H_ +#define _BF6XX_SPORT_H_ + +#include <linux/platform_device.h> +#include <asm/bfin_sport3.h> + +struct sport_device { + struct platform_device *pdev; + const unsigned short *pin_req; + struct sport_register *tx_regs; + struct sport_register *rx_regs; + int tx_dma_chan; + int rx_dma_chan; + int tx_err_irq; + int rx_err_irq; + + void (*tx_callback)(void *data); + void *tx_data; + void (*rx_callback)(void *data); + void *rx_data; + + struct dmasg *tx_desc; + struct dmasg *rx_desc; + unsigned int tx_desc_size; + unsigned int rx_desc_size; + unsigned char *tx_buf; + unsigned char *rx_buf; + unsigned int tx_fragsize; + unsigned int rx_fragsize; + unsigned int tx_frags; + unsigned int rx_frags; + unsigned int wdsize; +}; + +struct sport_params { + u32 spctl; + u32 div; +}; + +struct sport_device *sport_create(struct platform_device *pdev); +void sport_delete(struct sport_device *sport); +int sport_set_tx_params(struct sport_device *sport, + struct sport_params *params); +int sport_set_rx_params(struct sport_device *sport, + struct sport_params *params); +void sport_tx_start(struct sport_device *sport); +void sport_rx_start(struct sport_device *sport); +void sport_tx_stop(struct sport_device *sport); +void sport_rx_stop(struct sport_device *sport); +void sport_set_tx_callback(struct sport_device *sport, + void (*tx_callback)(void *), void *tx_data); +void sport_set_rx_callback(struct sport_device *sport, + void (*rx_callback)(void *), void *rx_data); +int sport_config_tx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize); +int sport_config_rx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize); +unsigned long sport_curr_offset_tx(struct sport_device *sport); +unsigned long sport_curr_offset_rx(struct sport_device *sport); + + + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 59d8efaa17e9..9f8e8594aeb9 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -12,6 +12,7 @@ config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 + select SND_SOC_AB8500_CODEC if ABX500_CORE select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS select SND_SOC_AD1836 if SPI_MASTER select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI @@ -29,19 +30,26 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C + select SND_SOC_CS42L52 if I2C select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C + select SND_SOC_DA732X if I2C select SND_SOC_DFBMCS320 + select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C + select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_MC13783 if MFD_MC13XXX + select SND_SOC_ML26124 if I2C + select SND_SOC_OMAP_HDMI_CODEC if OMAP4_DSS_HDMI select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_SGTL5000 if I2C @@ -49,6 +57,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SPDIF select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI select SND_SOC_STA32X if I2C + select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER @@ -65,6 +74,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM2000 if I2C select SND_SOC_WM2200 if I2C select SND_SOC_WM5100 if I2C + select SND_SOC_WM5102 if MFD_WM5102 + select SND_SOC_WM5110 if MFD_WM5110 select SND_SOC_WM8350 if MFD_WM8350 select SND_SOC_WM8400 if MFD_WM8400 select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI @@ -121,11 +132,21 @@ config SND_SOC_ALL_CODECS config SND_SOC_88PM860X tristate +config SND_SOC_ARIZONA + tristate + default y if SND_SOC_WM5102=y + default y if SND_SOC_WM5110=y + default m if SND_SOC_WM5102=m + default m if SND_SOC_WM5110=m + config SND_SOC_WM_HUBS tristate default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y default m if SND_SOC_WM8993=m || SND_SOC_WM8994=m +config SND_SOC_AB8500_CODEC + tristate + config SND_SOC_AC97_CODEC tristate select SND_AC97_CODEC @@ -181,6 +202,9 @@ config SND_SOC_CQ0093VC config SND_SOC_CS42L51 tristate +config SND_SOC_CS42L52 + tristate + config SND_SOC_CS42L73 tristate @@ -211,12 +235,21 @@ config SND_SOC_L3 config SND_SOC_DA7210 tristate +config SND_SOC_DA732X + tristate + config SND_SOC_DFBMCS320 tristate config SND_SOC_DMIC tristate +config SND_SOC_ISABELLE + tristate + +config SND_SOC_LM49453 + tristate + config SND_SOC_MAX98088 tristate @@ -226,6 +259,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate +config SND_SOC_OMAP_HDMI_CODEC + tristate + config SND_SOC_PCM3008 tristate @@ -252,6 +288,9 @@ config SND_SOC_SSM2602 config SND_SOC_STA32X tristate +config SND_SOC_STA529 + tristate + config SND_SOC_STAC9766 tristate @@ -299,6 +338,12 @@ config SND_SOC_WM2200 config SND_SOC_WM5100 tristate +config SND_SOC_WM5102 + tristate + +config SND_SOC_WM5110 + tristate + config SND_SOC_WM8350 tristate @@ -435,5 +480,11 @@ config SND_SOC_MAX9768 config SND_SOC_MAX9877 tristate +config SND_SOC_MC13783 + tristate + +config SND_SOC_ML26124 + tristate + config SND_SOC_TPA6130A2 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6662eb0cdcc0..34148bb59c68 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,4 +1,5 @@ snd-soc-88pm860x-objs := 88pm860x-codec.o +snd-soc-ab8500-codec-objs := ab8500-codec.o snd-soc-ac97-objs := ac97.o snd-soc-ad1836-objs := ad1836.o snd-soc-ad193x-objs := ad193x.o @@ -13,22 +14,30 @@ snd-soc-ak4535-objs := ak4535.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o +snd-soc-arizona-objs := arizona.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o +snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o +snd-soc-da732x-objs := da732x.o snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o +snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o snd-soc-lm4857-objs := lm4857.o +snd-soc-lm49453-objs := lm49453.o snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o +snd-soc-mc13783-objs := mc13783.o +snd-soc-ml26124-objs := ml26124.o +snd-soc-omap-hdmi-codec-objs := omap-hdmi.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -36,9 +45,11 @@ snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o snd-soc-sigmadsp-objs := sigmadsp.o snd-soc-sn95031-objs := sn95031.o -snd-soc-spdif-objs := spdif_transciever.o +snd-soc-spdif-tx-objs := spdif_transciever.o +snd-soc-spdif-rx-objs := spdif_receiver.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o +snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o @@ -54,6 +65,8 @@ snd-soc-wm1250-ev1-objs := wm1250-ev1.o snd-soc-wm2000-objs := wm2000.o snd-soc-wm2200-objs := wm2200.o snd-soc-wm5100-objs := wm5100.o wm5100-tables.o +snd-soc-wm5102-objs := wm5102.o +snd-soc-wm5110-objs := wm5110.o snd-soc-wm8350-objs := wm8350.o snd-soc-wm8400-objs := wm8400.o snd-soc-wm8510-objs := wm8510.o @@ -103,6 +116,7 @@ snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o +obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o @@ -119,30 +133,39 @@ obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o +obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o +obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o +obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o +obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o +obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o -obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o +obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o +obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o +obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_OMAP_HDMI_CODEC) += snd-soc-omap-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o -obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o +obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o +obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o @@ -158,6 +181,8 @@ obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o obj-$(CONFIG_SND_SOC_WM2200) += snd-soc-wm2200.o obj-$(CONFIG_SND_SOC_WM5100) += snd-soc-wm5100.o +obj-$(CONFIG_SND_SOC_WM5102) += snd-soc-wm5102.o +obj-$(CONFIG_SND_SOC_WM5110) += snd-soc-wm5110.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c new file mode 100644 index 000000000000..3c795921c5f6 --- /dev/null +++ b/sound/soc/codecs/ab8500-codec.c @@ -0,0 +1,2522 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com>, + * Kristoffer Karlsson <kristoffer.karlsson@stericsson.com>, + * Roger Nilsson <roger.xr.nilsson@stericsson.com>, + * for ST-Ericsson. + * + * Based on the early work done by: + * Mikko J. Lehto <mikko.lehto@symbio.com>, + * Mikko Sarmanne <mikko.sarmanne@symbio.com>, + * Jarmo K. Kuronen <jarmo.kuronen@symbio.com>, + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/slab.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/platform_device.h> +#include <linux/mutex.h> +#include <linux/mfd/abx500/ab8500.h> +#include <linux/mfd/abx500.h> +#include <linux/mfd/abx500/ab8500-sysctrl.h> +#include <linux/mfd/abx500/ab8500-codec.h> +#include <linux/regulator/consumer.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "ab8500-codec.h" + +/* Macrocell value definitions */ +#define CLK_32K_OUT2_DISABLE 0x01 +#define INACTIVE_RESET_AUDIO 0x02 +#define ENABLE_AUDIO_CLK_TO_AUDIO_BLK 0x10 +#define ENABLE_VINTCORE12_SUPPLY 0x04 +#define GPIO27_DIR_OUTPUT 0x04 +#define GPIO29_DIR_OUTPUT 0x10 +#define GPIO31_DIR_OUTPUT 0x40 + +/* Macrocell register definitions */ +#define AB8500_CTRL3_REG 0x0200 +#define AB8500_GPIO_DIR4_REG 0x1013 + +/* Nr of FIR/IIR-coeff banks in ANC-block */ +#define AB8500_NR_OF_ANC_COEFF_BANKS 2 + +/* Minimum duration to keep ANC IIR Init bit high or +low before proceeding with the configuration sequence */ +#define AB8500_ANC_SM_DELAY 2000 + +#define AB8500_FILTER_CONTROL(xname, xcount, xmin, xmax) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = filter_control_info, \ + .get = filter_control_get, .put = filter_control_put, \ + .private_value = (unsigned long)&(struct filter_control) \ + {.count = xcount, .min = xmin, .max = xmax} } + +struct filter_control { + long min, max; + unsigned int count; + long value[128]; +}; + +/* Sidetone states */ +static const char * const enum_sid_state[] = { + "Unconfigured", + "Apply FIR", + "FIR is configured", +}; +enum sid_state { + SID_UNCONFIGURED = 0, + SID_APPLY_FIR = 1, + SID_FIR_CONFIGURED = 2, +}; + +static const char * const enum_anc_state[] = { + "Unconfigured", + "Apply FIR and IIR", + "FIR and IIR are configured", + "Apply FIR", + "FIR is configured", + "Apply IIR", + "IIR is configured" +}; +enum anc_state { + ANC_UNCONFIGURED = 0, + ANC_APPLY_FIR_IIR = 1, + ANC_FIR_IIR_CONFIGURED = 2, + ANC_APPLY_FIR = 3, + ANC_FIR_CONFIGURED = 4, + ANC_APPLY_IIR = 5, + ANC_IIR_CONFIGURED = 6 +}; + +/* Analog microphones */ +enum amic_idx { + AMIC_IDX_1A, + AMIC_IDX_1B, + AMIC_IDX_2 +}; + +struct ab8500_codec_drvdata_dbg { + struct regulator *vaud; + struct regulator *vamic1; + struct regulator *vamic2; + struct regulator *vdmic; +}; + +/* Private data for AB8500 device-driver */ +struct ab8500_codec_drvdata { + /* Sidetone */ + long *sid_fir_values; + enum sid_state sid_status; + + /* ANC */ + struct mutex anc_lock; + long *anc_fir_values; + long *anc_iir_values; + enum anc_state anc_status; +}; + +static inline const char *amic_micbias_str(enum amic_micbias micbias) +{ + switch (micbias) { + case AMIC_MICBIAS_VAMIC1: + return "VAMIC1"; + case AMIC_MICBIAS_VAMIC2: + return "VAMIC2"; + default: + return "Unknown"; + } +} + +static inline const char *amic_type_str(enum amic_type type) +{ + switch (type) { + case AMIC_TYPE_DIFFERENTIAL: + return "DIFFERENTIAL"; + case AMIC_TYPE_SINGLE_ENDED: + return "SINGLE ENDED"; + default: + return "Unknown"; + } +} + +/* + * Read'n'write functions + */ + +/* Read a register from the audio-bank of AB8500 */ +static unsigned int ab8500_codec_read_reg(struct snd_soc_codec *codec, + unsigned int reg) +{ + int status; + unsigned int value = 0; + + u8 value8; + status = abx500_get_register_interruptible(codec->dev, AB8500_AUDIO, + reg, &value8); + if (status < 0) { + dev_err(codec->dev, + "%s: ERROR: Register (0x%02x:0x%02x) read failed (%d).\n", + __func__, (u8)AB8500_AUDIO, (u8)reg, status); + } else { + dev_dbg(codec->dev, + "%s: Read 0x%02x from register 0x%02x:0x%02x\n", + __func__, value8, (u8)AB8500_AUDIO, (u8)reg); + value = (unsigned int)value8; + } + + return value; +} + +/* Write to a register in the audio-bank of AB8500 */ +static int ab8500_codec_write_reg(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + int status; + + status = abx500_set_register_interruptible(codec->dev, AB8500_AUDIO, + reg, value); + if (status < 0) + dev_err(codec->dev, + "%s: ERROR: Register (%02x:%02x) write failed (%d).\n", + __func__, (u8)AB8500_AUDIO, (u8)reg, status); + else + dev_dbg(codec->dev, + "%s: Wrote 0x%02x into register %02x:%02x\n", + __func__, (u8)value, (u8)AB8500_AUDIO, (u8)reg); + + return status; +} + +/* + * Controls - DAPM + */ + +/* Earpiece */ + +/* Earpiece source selector */ +static const char * const enum_ear_lineout_source[] = {"Headset Left", + "Speaker Left"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ear_lineout_source, AB8500_DMICFILTCONF, + AB8500_DMICFILTCONF_DA3TOEAR, enum_ear_lineout_source); +static const struct snd_kcontrol_new dapm_ear_lineout_source = + SOC_DAPM_ENUM("Earpiece or LineOut Mono Source", + dapm_enum_ear_lineout_source); + +/* LineOut */ + +/* LineOut source selector */ +static const char * const enum_lineout_source[] = {"Mono Path", "Stereo Path"}; +static SOC_ENUM_DOUBLE_DECL(dapm_enum_lineout_source, AB8500_ANACONF5, + AB8500_ANACONF5_HSLDACTOLOL, + AB8500_ANACONF5_HSRDACTOLOR, enum_lineout_source); +static const struct snd_kcontrol_new dapm_lineout_source[] = { + SOC_DAPM_ENUM("LineOut Source", dapm_enum_lineout_source), +}; + +/* Handsfree */ + +/* Speaker Left - ANC selector */ +static const char * const enum_HFx_sel[] = {"Audio Path", "ANC"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_HFl_sel, AB8500_DIGMULTCONF2, + AB8500_DIGMULTCONF2_HFLSEL, enum_HFx_sel); +static const struct snd_kcontrol_new dapm_HFl_select[] = { + SOC_DAPM_ENUM("Speaker Left Source", dapm_enum_HFl_sel), +}; + +/* Speaker Right - ANC selector */ +static SOC_ENUM_SINGLE_DECL(dapm_enum_HFr_sel, AB8500_DIGMULTCONF2, + AB8500_DIGMULTCONF2_HFRSEL, enum_HFx_sel); +static const struct snd_kcontrol_new dapm_HFr_select[] = { + SOC_DAPM_ENUM("Speaker Right Source", dapm_enum_HFr_sel), +}; + +/* Mic 1 */ + +/* Mic 1 - Mic 1a or 1b selector */ +static const char * const enum_mic1ab_sel[] = {"Mic 1b", "Mic 1a"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_mic1ab_sel, AB8500_ANACONF3, + AB8500_ANACONF3_MIC1SEL, enum_mic1ab_sel); +static const struct snd_kcontrol_new dapm_mic1ab_mux[] = { + SOC_DAPM_ENUM("Mic 1a or 1b Select", dapm_enum_mic1ab_sel), +}; + +/* Mic 1 - AD3 - Mic 1 or DMic 3 selector */ +static const char * const enum_ad3_sel[] = {"Mic 1", "DMic 3"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad3_sel, AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_AD3SEL, enum_ad3_sel); +static const struct snd_kcontrol_new dapm_ad3_select[] = { + SOC_DAPM_ENUM("AD3 Source Select", dapm_enum_ad3_sel), +}; + +/* Mic 1 - AD6 - Mic 1 or DMic 6 selector */ +static const char * const enum_ad6_sel[] = {"Mic 1", "DMic 6"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad6_sel, AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_AD6SEL, enum_ad6_sel); +static const struct snd_kcontrol_new dapm_ad6_select[] = { + SOC_DAPM_ENUM("AD6 Source Select", dapm_enum_ad6_sel), +}; + +/* Mic 2 */ + +/* Mic 2 - AD5 - Mic 2 or DMic 5 selector */ +static const char * const enum_ad5_sel[] = {"Mic 2", "DMic 5"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad5_sel, AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_AD5SEL, enum_ad5_sel); +static const struct snd_kcontrol_new dapm_ad5_select[] = { + SOC_DAPM_ENUM("AD5 Source Select", dapm_enum_ad5_sel), +}; + +/* LineIn */ + +/* LineIn left - AD1 - LineIn Left or DMic 1 selector */ +static const char * const enum_ad1_sel[] = {"LineIn Left", "DMic 1"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad1_sel, AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_AD1SEL, enum_ad1_sel); +static const struct snd_kcontrol_new dapm_ad1_select[] = { + SOC_DAPM_ENUM("AD1 Source Select", dapm_enum_ad1_sel), +}; + +/* LineIn right - Mic 2 or LineIn Right selector */ +static const char * const enum_mic2lr_sel[] = {"Mic 2", "LineIn Right"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_mic2lr_sel, AB8500_ANACONF3, + AB8500_ANACONF3_LINRSEL, enum_mic2lr_sel); +static const struct snd_kcontrol_new dapm_mic2lr_select[] = { + SOC_DAPM_ENUM("Mic 2 or LINR Select", dapm_enum_mic2lr_sel), +}; + +/* LineIn right - AD2 - LineIn Right or DMic2 selector */ +static const char * const enum_ad2_sel[] = {"LineIn Right", "DMic 2"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad2_sel, AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_AD2SEL, enum_ad2_sel); +static const struct snd_kcontrol_new dapm_ad2_select[] = { + SOC_DAPM_ENUM("AD2 Source Select", dapm_enum_ad2_sel), +}; + + +/* ANC */ + +static const char * const enum_anc_in_sel[] = {"Mic 1 / DMic 6", + "Mic 2 / DMic 5"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_anc_in_sel, AB8500_DMICFILTCONF, + AB8500_DMICFILTCONF_ANCINSEL, enum_anc_in_sel); +static const struct snd_kcontrol_new dapm_anc_in_select[] = { + SOC_DAPM_ENUM("ANC Source", dapm_enum_anc_in_sel), +}; + +/* ANC - Enable/Disable */ +static const struct snd_kcontrol_new dapm_anc_enable[] = { + SOC_DAPM_SINGLE("Switch", AB8500_ANCCONF1, + AB8500_ANCCONF1_ENANC, 0, 0), +}; + +/* ANC to Earpiece - Mute */ +static const struct snd_kcontrol_new dapm_anc_ear_mute[] = { + SOC_DAPM_SINGLE("Switch", AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_ANCSEL, 1, 0), +}; + + + +/* Sidetone left */ + +/* Sidetone left - Input selector */ +static const char * const enum_stfir1_in_sel[] = { + "LineIn Left", "LineIn Right", "Mic 1", "Headset Left" +}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_stfir1_in_sel, AB8500_DIGMULTCONF2, + AB8500_DIGMULTCONF2_FIRSID1SEL, enum_stfir1_in_sel); +static const struct snd_kcontrol_new dapm_stfir1_in_select[] = { + SOC_DAPM_ENUM("Sidetone Left Source", dapm_enum_stfir1_in_sel), +}; + +/* Sidetone right path */ + +/* Sidetone right - Input selector */ +static const char * const enum_stfir2_in_sel[] = { + "LineIn Right", "Mic 1", "DMic 4", "Headset Right" +}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_stfir2_in_sel, AB8500_DIGMULTCONF2, + AB8500_DIGMULTCONF2_FIRSID2SEL, enum_stfir2_in_sel); +static const struct snd_kcontrol_new dapm_stfir2_in_select[] = { + SOC_DAPM_ENUM("Sidetone Right Source", dapm_enum_stfir2_in_sel), +}; + +/* Vibra */ + +static const char * const enum_pwm2vibx[] = {"Audio Path", "PWM Generator"}; + +static SOC_ENUM_SINGLE_DECL(dapm_enum_pwm2vib1, AB8500_PWMGENCONF1, + AB8500_PWMGENCONF1_PWMTOVIB1, enum_pwm2vibx); + +static const struct snd_kcontrol_new dapm_pwm2vib1[] = { + SOC_DAPM_ENUM("Vibra 1 Controller", dapm_enum_pwm2vib1), +}; + +static SOC_ENUM_SINGLE_DECL(dapm_enum_pwm2vib2, AB8500_PWMGENCONF1, + AB8500_PWMGENCONF1_PWMTOVIB2, enum_pwm2vibx); + +static const struct snd_kcontrol_new dapm_pwm2vib2[] = { + SOC_DAPM_ENUM("Vibra 2 Controller", dapm_enum_pwm2vib2), +}; + +/* + * DAPM-widgets + */ + +static const struct snd_soc_dapm_widget ab8500_dapm_widgets[] = { + + /* Clocks */ + SND_SOC_DAPM_CLOCK_SUPPLY("audioclk"), + + /* Regulators */ + SND_SOC_DAPM_REGULATOR_SUPPLY("V-AUD", 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC1", 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC2", 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-DMIC", 0), + + /* Power */ + SND_SOC_DAPM_SUPPLY("Audio Power", + AB8500_POWERUP, AB8500_POWERUP_POWERUP, 0, + NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("Audio Analog Power", + AB8500_POWERUP, AB8500_POWERUP_ENANA, 0, + NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* Main supply node */ + SND_SOC_DAPM_SUPPLY("Main Supply", SND_SOC_NOPM, 0, 0, + NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* DA/AD */ + + SND_SOC_DAPM_INPUT("ADC Input"), + SND_SOC_DAPM_ADC("ADC", "ab8500_0c", SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("DAC Output"), + + SND_SOC_DAPM_AIF_IN("DA_IN1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN3", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN4", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN5", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN6", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT3", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT4", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT57", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT68", NULL, 0, SND_SOC_NOPM, 0, 0), + + /* Headset path */ + + SND_SOC_DAPM_SUPPLY("Charge Pump", AB8500_ANACONF5, + AB8500_ANACONF5_ENCPHS, 0, NULL, 0), + + SND_SOC_DAPM_DAC("DA1 Enable", "ab8500_0p", + AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA1, 0), + SND_SOC_DAPM_DAC("DA2 Enable", "ab8500_0p", + AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA2, 0), + + SND_SOC_DAPM_PGA("HSL Digital Volume", SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_PGA("HSR Digital Volume", SND_SOC_NOPM, 0, 0, + NULL, 0), + + SND_SOC_DAPM_DAC("HSL DAC", "ab8500_0p", + AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACHSL, 0), + SND_SOC_DAPM_DAC("HSR DAC", "ab8500_0p", + AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACHSR, 0), + SND_SOC_DAPM_MIXER("HSL DAC Mute", AB8500_MUTECONF, + AB8500_MUTECONF_MUTDACHSL, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("HSR DAC Mute", AB8500_MUTECONF, + AB8500_MUTECONF_MUTDACHSR, 1, + NULL, 0), + SND_SOC_DAPM_DAC("HSL DAC Driver", "ab8500_0p", + AB8500_ANACONF3, AB8500_ANACONF3_ENDRVHSL, 0), + SND_SOC_DAPM_DAC("HSR DAC Driver", "ab8500_0p", + AB8500_ANACONF3, AB8500_ANACONF3_ENDRVHSR, 0), + + SND_SOC_DAPM_MIXER("HSL Mute", + AB8500_MUTECONF, AB8500_MUTECONF_MUTHSL, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("HSR Mute", + AB8500_MUTECONF, AB8500_MUTECONF_MUTHSR, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("HSL Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENHSL, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("HSR Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENHSR, 0, + NULL, 0), + SND_SOC_DAPM_PGA("HSL Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_PGA("HSR Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + + SND_SOC_DAPM_OUTPUT("Headset Left"), + SND_SOC_DAPM_OUTPUT("Headset Right"), + + /* LineOut path */ + + SND_SOC_DAPM_MUX("LineOut Source", + SND_SOC_NOPM, 0, 0, dapm_lineout_source), + + SND_SOC_DAPM_MIXER("LOL Disable HFL", + AB8500_ANACONF4, AB8500_ANACONF4_ENHFL, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("LOR Disable HFR", + AB8500_ANACONF4, AB8500_ANACONF4_ENHFR, 1, + NULL, 0), + + SND_SOC_DAPM_MIXER("LOL Enable", + AB8500_ANACONF5, AB8500_ANACONF5_ENLOL, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("LOR Enable", + AB8500_ANACONF5, AB8500_ANACONF5_ENLOR, 0, + NULL, 0), + + SND_SOC_DAPM_OUTPUT("LineOut Left"), + SND_SOC_DAPM_OUTPUT("LineOut Right"), + + /* Earpiece path */ + + SND_SOC_DAPM_MUX("Earpiece or LineOut Mono Source", + SND_SOC_NOPM, 0, 0, &dapm_ear_lineout_source), + SND_SOC_DAPM_MIXER("EAR DAC", + AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACEAR, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("EAR Mute", + AB8500_MUTECONF, AB8500_MUTECONF_MUTEAR, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("EAR Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENEAR, 0, + NULL, 0), + + SND_SOC_DAPM_OUTPUT("Earpiece"), + + /* Handsfree path */ + + SND_SOC_DAPM_MIXER("DA3 Channel Volume", + AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA3, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DA4 Channel Volume", + AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA4, 0, + NULL, 0), + SND_SOC_DAPM_MUX("Speaker Left Source", + SND_SOC_NOPM, 0, 0, dapm_HFl_select), + SND_SOC_DAPM_MUX("Speaker Right Source", + SND_SOC_NOPM, 0, 0, dapm_HFr_select), + SND_SOC_DAPM_MIXER("HFL DAC", AB8500_DAPATHCONF, + AB8500_DAPATHCONF_ENDACHFL, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("HFR DAC", + AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACHFR, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DA4 or ANC path to HfR", + AB8500_DIGMULTCONF2, AB8500_DIGMULTCONF2_DATOHFREN, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DA3 or ANC path to HfL", + AB8500_DIGMULTCONF2, AB8500_DIGMULTCONF2_DATOHFLEN, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("HFL Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENHFL, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("HFR Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENHFR, 0, + NULL, 0), + + SND_SOC_DAPM_OUTPUT("Speaker Left"), + SND_SOC_DAPM_OUTPUT("Speaker Right"), + + /* Vibrator path */ + + SND_SOC_DAPM_INPUT("PWMGEN1"), + SND_SOC_DAPM_INPUT("PWMGEN2"), + + SND_SOC_DAPM_MIXER("DA5 Channel Volume", + AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA5, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DA6 Channel Volume", + AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA6, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("VIB1 DAC", + AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACVIB1, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("VIB2 DAC", + AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACVIB2, 0, + NULL, 0), + SND_SOC_DAPM_MUX("Vibra 1 Controller", + SND_SOC_NOPM, 0, 0, dapm_pwm2vib1), + SND_SOC_DAPM_MUX("Vibra 2 Controller", + SND_SOC_NOPM, 0, 0, dapm_pwm2vib2), + SND_SOC_DAPM_MIXER("VIB1 Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENVIB1, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("VIB2 Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENVIB2, 0, + NULL, 0), + + SND_SOC_DAPM_OUTPUT("Vibra 1"), + SND_SOC_DAPM_OUTPUT("Vibra 2"), + + /* Mic 1 */ + + SND_SOC_DAPM_INPUT("Mic 1"), + + SND_SOC_DAPM_MUX("Mic 1a or 1b Select", + SND_SOC_NOPM, 0, 0, dapm_mic1ab_mux), + SND_SOC_DAPM_MIXER("MIC1 Mute", + AB8500_ANACONF2, AB8500_ANACONF2_MUTMIC1, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("MIC1A V-AMICx Enable", + AB8500_ANACONF2, AB8500_ANACONF2_ENMIC1, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("MIC1B V-AMICx Enable", + AB8500_ANACONF2, AB8500_ANACONF2_ENMIC1, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("MIC1 ADC", + AB8500_ANACONF3, AB8500_ANACONF3_ENADCMIC, 0, + NULL, 0), + SND_SOC_DAPM_MUX("AD3 Source Select", + SND_SOC_NOPM, 0, 0, dapm_ad3_select), + SND_SOC_DAPM_MIXER("AD3 Channel Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD3 Enable", + AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD34, 0, + NULL, 0), + + /* Mic 2 */ + + SND_SOC_DAPM_INPUT("Mic 2"), + + SND_SOC_DAPM_MIXER("MIC2 Mute", + AB8500_ANACONF2, AB8500_ANACONF2_MUTMIC2, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("MIC2 V-AMICx Enable", AB8500_ANACONF2, + AB8500_ANACONF2_ENMIC2, 0, + NULL, 0), + + /* LineIn */ + + SND_SOC_DAPM_INPUT("LineIn Left"), + SND_SOC_DAPM_INPUT("LineIn Right"), + + SND_SOC_DAPM_MIXER("LINL Mute", + AB8500_ANACONF2, AB8500_ANACONF2_MUTLINL, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("LINR Mute", + AB8500_ANACONF2, AB8500_ANACONF2_MUTLINR, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("LINL Enable", AB8500_ANACONF2, + AB8500_ANACONF2_ENLINL, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("LINR Enable", AB8500_ANACONF2, + AB8500_ANACONF2_ENLINR, 0, + NULL, 0), + + /* LineIn Bypass path */ + SND_SOC_DAPM_MIXER("LINL to HSL Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("LINR to HSR Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + + /* LineIn, Mic 2 */ + SND_SOC_DAPM_MUX("Mic 2 or LINR Select", + SND_SOC_NOPM, 0, 0, dapm_mic2lr_select), + SND_SOC_DAPM_MIXER("LINL ADC", AB8500_ANACONF3, + AB8500_ANACONF3_ENADCLINL, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("LINR ADC", AB8500_ANACONF3, + AB8500_ANACONF3_ENADCLINR, 0, + NULL, 0), + SND_SOC_DAPM_MUX("AD1 Source Select", + SND_SOC_NOPM, 0, 0, dapm_ad1_select), + SND_SOC_DAPM_MUX("AD2 Source Select", + SND_SOC_NOPM, 0, 0, dapm_ad2_select), + SND_SOC_DAPM_MIXER("AD1 Channel Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD2 Channel Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + + SND_SOC_DAPM_MIXER("AD12 Enable", + AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD12, 0, + NULL, 0), + + /* HD Capture path */ + + SND_SOC_DAPM_MUX("AD5 Source Select", + SND_SOC_NOPM, 0, 0, dapm_ad5_select), + SND_SOC_DAPM_MUX("AD6 Source Select", + SND_SOC_NOPM, 0, 0, dapm_ad6_select), + SND_SOC_DAPM_MIXER("AD5 Channel Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD6 Channel Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD57 Enable", + AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD5768, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD68 Enable", + AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD5768, 0, + NULL, 0), + + /* Digital Microphone path */ + + SND_SOC_DAPM_INPUT("DMic 1"), + SND_SOC_DAPM_INPUT("DMic 2"), + SND_SOC_DAPM_INPUT("DMic 3"), + SND_SOC_DAPM_INPUT("DMic 4"), + SND_SOC_DAPM_INPUT("DMic 5"), + SND_SOC_DAPM_INPUT("DMic 6"), + + SND_SOC_DAPM_MIXER("DMIC1", + AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC1, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DMIC2", + AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC2, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DMIC3", + AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC3, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DMIC4", + AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC4, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DMIC5", + AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC5, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DMIC6", + AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC6, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD4 Channel Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD4 Enable", + AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD34, + 0, NULL, 0), + + /* Acoustical Noise Cancellation path */ + + SND_SOC_DAPM_INPUT("ANC Configure Input"), + SND_SOC_DAPM_OUTPUT("ANC Configure Output"), + + SND_SOC_DAPM_MUX("ANC Source", + SND_SOC_NOPM, 0, 0, + dapm_anc_in_select), + SND_SOC_DAPM_SWITCH("ANC", + SND_SOC_NOPM, 0, 0, + dapm_anc_enable), + SND_SOC_DAPM_SWITCH("ANC to Earpiece", + SND_SOC_NOPM, 0, 0, + dapm_anc_ear_mute), + + /* Sidetone Filter path */ + + SND_SOC_DAPM_MUX("Sidetone Left Source", + SND_SOC_NOPM, 0, 0, + dapm_stfir1_in_select), + SND_SOC_DAPM_MUX("Sidetone Right Source", + SND_SOC_NOPM, 0, 0, + dapm_stfir2_in_select), + SND_SOC_DAPM_MIXER("STFIR1 Control", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("STFIR2 Control", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("STFIR1 Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("STFIR2 Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), +}; + +/* + * DAPM-routes + */ +static const struct snd_soc_dapm_route ab8500_dapm_routes[] = { + /* Power AB8500 audio-block when AD/DA is active */ + {"Main Supply", NULL, "V-AUD"}, + {"Main Supply", NULL, "audioclk"}, + {"Main Supply", NULL, "Audio Power"}, + {"Main Supply", NULL, "Audio Analog Power"}, + + {"DAC", NULL, "ab8500_0p"}, + {"DAC", NULL, "Main Supply"}, + {"ADC", NULL, "ab8500_0c"}, + {"ADC", NULL, "Main Supply"}, + + /* ANC Configure */ + {"ANC Configure Input", NULL, "Main Supply"}, + {"ANC Configure Output", NULL, "ANC Configure Input"}, + + /* AD/DA */ + {"ADC", NULL, "ADC Input"}, + {"DAC Output", NULL, "DAC"}, + + /* Powerup charge pump if DA1/2 is in use */ + + {"DA_IN1", NULL, "ab8500_0p"}, + {"DA_IN1", NULL, "Charge Pump"}, + {"DA_IN2", NULL, "ab8500_0p"}, + {"DA_IN2", NULL, "Charge Pump"}, + + /* Headset path */ + + {"DA1 Enable", NULL, "DA_IN1"}, + {"DA2 Enable", NULL, "DA_IN2"}, + + {"HSL Digital Volume", NULL, "DA1 Enable"}, + {"HSR Digital Volume", NULL, "DA2 Enable"}, + + {"HSL DAC", NULL, "HSL Digital Volume"}, + {"HSR DAC", NULL, "HSR Digital Volume"}, + + {"HSL DAC Mute", NULL, "HSL DAC"}, + {"HSR DAC Mute", NULL, "HSR DAC"}, + + {"HSL DAC Driver", NULL, "HSL DAC Mute"}, + {"HSR DAC Driver", NULL, "HSR DAC Mute"}, + + {"HSL Mute", NULL, "HSL DAC Driver"}, + {"HSR Mute", NULL, "HSR DAC Driver"}, + + {"HSL Enable", NULL, "HSL Mute"}, + {"HSR Enable", NULL, "HSR Mute"}, + + {"HSL Volume", NULL, "HSL Enable"}, + {"HSR Volume", NULL, "HSR Enable"}, + + {"Headset Left", NULL, "HSL Volume"}, + {"Headset Right", NULL, "HSR Volume"}, + + /* HF or LineOut path */ + + {"DA_IN3", NULL, "ab8500_0p"}, + {"DA3 Channel Volume", NULL, "DA_IN3"}, + {"DA_IN4", NULL, "ab8500_0p"}, + {"DA4 Channel Volume", NULL, "DA_IN4"}, + + {"Speaker Left Source", "Audio Path", "DA3 Channel Volume"}, + {"Speaker Right Source", "Audio Path", "DA4 Channel Volume"}, + + {"DA3 or ANC path to HfL", NULL, "Speaker Left Source"}, + {"DA4 or ANC path to HfR", NULL, "Speaker Right Source"}, + + /* HF path */ + + {"HFL DAC", NULL, "DA3 or ANC path to HfL"}, + {"HFR DAC", NULL, "DA4 or ANC path to HfR"}, + + {"HFL Enable", NULL, "HFL DAC"}, + {"HFR Enable", NULL, "HFR DAC"}, + + {"Speaker Left", NULL, "HFL Enable"}, + {"Speaker Right", NULL, "HFR Enable"}, + + /* Earpiece path */ + + {"Earpiece or LineOut Mono Source", "Headset Left", + "HSL Digital Volume"}, + {"Earpiece or LineOut Mono Source", "Speaker Left", + "DA3 or ANC path to HfL"}, + + {"EAR DAC", NULL, "Earpiece or LineOut Mono Source"}, + + {"EAR Mute", NULL, "EAR DAC"}, + + {"EAR Enable", NULL, "EAR Mute"}, + + {"Earpiece", NULL, "EAR Enable"}, + + /* LineOut path stereo */ + + {"LineOut Source", "Stereo Path", "HSL DAC Driver"}, + {"LineOut Source", "Stereo Path", "HSR DAC Driver"}, + + /* LineOut path mono */ + + {"LineOut Source", "Mono Path", "EAR DAC"}, + + /* LineOut path */ + + {"LOL Disable HFL", NULL, "LineOut Source"}, + {"LOR Disable HFR", NULL, "LineOut Source"}, + + {"LOL Enable", NULL, "LOL Disable HFL"}, + {"LOR Enable", NULL, "LOR Disable HFR"}, + + {"LineOut Left", NULL, "LOL Enable"}, + {"LineOut Right", NULL, "LOR Enable"}, + + /* Vibrator path */ + + {"DA_IN5", NULL, "ab8500_0p"}, + {"DA5 Channel Volume", NULL, "DA_IN5"}, + {"DA_IN6", NULL, "ab8500_0p"}, + {"DA6 Channel Volume", NULL, "DA_IN6"}, + + {"VIB1 DAC", NULL, "DA5 Channel Volume"}, + {"VIB2 DAC", NULL, "DA6 Channel Volume"}, + + {"Vibra 1 Controller", "Audio Path", "VIB1 DAC"}, + {"Vibra 2 Controller", "Audio Path", "VIB2 DAC"}, + {"Vibra 1 Controller", "PWM Generator", "PWMGEN1"}, + {"Vibra 2 Controller", "PWM Generator", "PWMGEN2"}, + + {"VIB1 Enable", NULL, "Vibra 1 Controller"}, + {"VIB2 Enable", NULL, "Vibra 2 Controller"}, + + {"Vibra 1", NULL, "VIB1 Enable"}, + {"Vibra 2", NULL, "VIB2 Enable"}, + + + /* Mic 2 */ + + {"MIC2 V-AMICx Enable", NULL, "Mic 2"}, + + /* LineIn */ + {"LINL Mute", NULL, "LineIn Left"}, + {"LINR Mute", NULL, "LineIn Right"}, + + {"LINL Enable", NULL, "LINL Mute"}, + {"LINR Enable", NULL, "LINR Mute"}, + + /* LineIn, Mic 2 */ + {"Mic 2 or LINR Select", "LineIn Right", "LINR Enable"}, + {"Mic 2 or LINR Select", "Mic 2", "MIC2 V-AMICx Enable"}, + + {"LINL ADC", NULL, "LINL Enable"}, + {"LINR ADC", NULL, "Mic 2 or LINR Select"}, + + {"AD1 Source Select", "LineIn Left", "LINL ADC"}, + {"AD2 Source Select", "LineIn Right", "LINR ADC"}, + + {"AD1 Channel Volume", NULL, "AD1 Source Select"}, + {"AD2 Channel Volume", NULL, "AD2 Source Select"}, + + {"AD12 Enable", NULL, "AD1 Channel Volume"}, + {"AD12 Enable", NULL, "AD2 Channel Volume"}, + + {"AD_OUT1", NULL, "ab8500_0c"}, + {"AD_OUT1", NULL, "AD12 Enable"}, + {"AD_OUT2", NULL, "ab8500_0c"}, + {"AD_OUT2", NULL, "AD12 Enable"}, + + /* Mic 1 */ + + {"MIC1 Mute", NULL, "Mic 1"}, + + {"MIC1A V-AMICx Enable", NULL, "MIC1 Mute"}, + {"MIC1B V-AMICx Enable", NULL, "MIC1 Mute"}, + + {"Mic 1a or 1b Select", "Mic 1a", "MIC1A V-AMICx Enable"}, + {"Mic 1a or 1b Select", "Mic 1b", "MIC1B V-AMICx Enable"}, + + {"MIC1 ADC", NULL, "Mic 1a or 1b Select"}, + + {"AD3 Source Select", "Mic 1", "MIC1 ADC"}, + + {"AD3 Channel Volume", NULL, "AD3 Source Select"}, + + {"AD3 Enable", NULL, "AD3 Channel Volume"}, + + {"AD_OUT3", NULL, "ab8500_0c"}, + {"AD_OUT3", NULL, "AD3 Enable"}, + + /* HD Capture path */ + + {"AD5 Source Select", "Mic 2", "LINR ADC"}, + {"AD6 Source Select", "Mic 1", "MIC1 ADC"}, + + {"AD5 Channel Volume", NULL, "AD5 Source Select"}, + {"AD6 Channel Volume", NULL, "AD6 Source Select"}, + + {"AD57 Enable", NULL, "AD5 Channel Volume"}, + {"AD68 Enable", NULL, "AD6 Channel Volume"}, + + {"AD_OUT57", NULL, "ab8500_0c"}, + {"AD_OUT57", NULL, "AD57 Enable"}, + {"AD_OUT68", NULL, "ab8500_0c"}, + {"AD_OUT68", NULL, "AD68 Enable"}, + + /* Digital Microphone path */ + + {"DMic 1", NULL, "V-DMIC"}, + {"DMic 2", NULL, "V-DMIC"}, + {"DMic 3", NULL, "V-DMIC"}, + {"DMic 4", NULL, "V-DMIC"}, + {"DMic 5", NULL, "V-DMIC"}, + {"DMic 6", NULL, "V-DMIC"}, + + {"AD1 Source Select", NULL, "DMic 1"}, + {"AD2 Source Select", NULL, "DMic 2"}, + {"AD3 Source Select", NULL, "DMic 3"}, + {"AD5 Source Select", NULL, "DMic 5"}, + {"AD6 Source Select", NULL, "DMic 6"}, + + {"AD4 Channel Volume", NULL, "DMic 4"}, + {"AD4 Enable", NULL, "AD4 Channel Volume"}, + + {"AD_OUT4", NULL, "ab8500_0c"}, + {"AD_OUT4", NULL, "AD4 Enable"}, + + /* LineIn Bypass path */ + + {"LINL to HSL Volume", NULL, "LINL Enable"}, + {"LINR to HSR Volume", NULL, "LINR Enable"}, + + {"HSL DAC Driver", NULL, "LINL to HSL Volume"}, + {"HSR DAC Driver", NULL, "LINR to HSR Volume"}, + + /* ANC path (Acoustic Noise Cancellation) */ + + {"ANC Source", "Mic 2 / DMic 5", "AD5 Channel Volume"}, + {"ANC Source", "Mic 1 / DMic 6", "AD6 Channel Volume"}, + + {"ANC", "Switch", "ANC Source"}, + + {"Speaker Left Source", "ANC", "ANC"}, + {"Speaker Right Source", "ANC", "ANC"}, + {"ANC to Earpiece", "Switch", "ANC"}, + + {"HSL Digital Volume", NULL, "ANC to Earpiece"}, + + /* Sidetone Filter path */ + + {"Sidetone Left Source", "LineIn Left", "AD12 Enable"}, + {"Sidetone Left Source", "LineIn Right", "AD12 Enable"}, + {"Sidetone Left Source", "Mic 1", "AD3 Enable"}, + {"Sidetone Left Source", "Headset Left", "DA_IN1"}, + {"Sidetone Right Source", "LineIn Right", "AD12 Enable"}, + {"Sidetone Right Source", "Mic 1", "AD3 Enable"}, + {"Sidetone Right Source", "DMic 4", "AD4 Enable"}, + {"Sidetone Right Source", "Headset Right", "DA_IN2"}, + + {"STFIR1 Control", NULL, "Sidetone Left Source"}, + {"STFIR2 Control", NULL, "Sidetone Right Source"}, + + {"STFIR1 Volume", NULL, "STFIR1 Control"}, + {"STFIR2 Volume", NULL, "STFIR2 Control"}, + + {"DA1 Enable", NULL, "STFIR1 Volume"}, + {"DA2 Enable", NULL, "STFIR2 Volume"}, +}; + +static const struct snd_soc_dapm_route ab8500_dapm_routes_mic1a_vamicx[] = { + {"MIC1A V-AMICx Enable", NULL, "V-AMIC1"}, + {"MIC1A V-AMICx Enable", NULL, "V-AMIC2"}, +}; + +static const struct snd_soc_dapm_route ab8500_dapm_routes_mic1b_vamicx[] = { + {"MIC1B V-AMICx Enable", NULL, "V-AMIC1"}, + {"MIC1B V-AMICx Enable", NULL, "V-AMIC2"}, +}; + +static const struct snd_soc_dapm_route ab8500_dapm_routes_mic2_vamicx[] = { + {"MIC2 V-AMICx Enable", NULL, "V-AMIC1"}, + {"MIC2 V-AMICx Enable", NULL, "V-AMIC2"}, +}; + +/* ANC FIR-coefficients configuration sequence */ +static void anc_fir(struct snd_soc_codec *codec, + unsigned int bnk, unsigned int par, unsigned int val) +{ + if (par == 0 && bnk == 0) + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ANCFIRUPDATE), + BIT(AB8500_ANCCONF1_ANCFIRUPDATE)); + + snd_soc_write(codec, AB8500_ANCCONF5, val >> 8 & 0xff); + snd_soc_write(codec, AB8500_ANCCONF6, val & 0xff); + + if (par == AB8500_ANC_FIR_COEFFS - 1 && bnk == 1) + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ANCFIRUPDATE), 0); +} + +/* ANC IIR-coefficients configuration sequence */ +static void anc_iir(struct snd_soc_codec *codec, unsigned int bnk, + unsigned int par, unsigned int val) +{ + if (par == 0) { + if (bnk == 0) { + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ANCIIRINIT), + BIT(AB8500_ANCCONF1_ANCIIRINIT)); + usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY); + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ANCIIRINIT), 0); + usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY); + } else { + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ANCIIRUPDATE), + BIT(AB8500_ANCCONF1_ANCIIRUPDATE)); + } + } else if (par > 3) { + snd_soc_write(codec, AB8500_ANCCONF7, 0); + snd_soc_write(codec, AB8500_ANCCONF8, val >> 16 & 0xff); + } + + snd_soc_write(codec, AB8500_ANCCONF7, val >> 8 & 0xff); + snd_soc_write(codec, AB8500_ANCCONF8, val & 0xff); + + if (par == AB8500_ANC_IIR_COEFFS - 1 && bnk == 1) + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ANCIIRUPDATE), 0); +} + +/* ANC IIR-/FIR-coefficients configuration sequence */ +static void anc_configure(struct snd_soc_codec *codec, + bool apply_fir, bool apply_iir) +{ + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); + unsigned int bnk, par, val; + + dev_dbg(codec->dev, "%s: Enter.\n", __func__); + + if (apply_fir) + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ENANC), 0); + + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ENANC), BIT(AB8500_ANCCONF1_ENANC)); + + if (apply_fir) + for (bnk = 0; bnk < AB8500_NR_OF_ANC_COEFF_BANKS; bnk++) + for (par = 0; par < AB8500_ANC_FIR_COEFFS; par++) { + val = snd_soc_read(codec, + drvdata->anc_fir_values[par]); + anc_fir(codec, bnk, par, val); + } + + if (apply_iir) + for (bnk = 0; bnk < AB8500_NR_OF_ANC_COEFF_BANKS; bnk++) + for (par = 0; par < AB8500_ANC_IIR_COEFFS; par++) { + val = snd_soc_read(codec, + drvdata->anc_iir_values[par]); + anc_iir(codec, bnk, par, val); + } + + dev_dbg(codec->dev, "%s: Exit.\n", __func__); +} + +/* + * Control-events + */ + +static int sid_status_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); + + mutex_lock(&codec->mutex); + ucontrol->value.integer.value[0] = drvdata->sid_status; + mutex_unlock(&codec->mutex); + + return 0; +} + +/* Write sidetone FIR-coefficients configuration sequence */ +static int sid_status_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); + unsigned int param, sidconf, val; + int status = 1; + + dev_dbg(codec->dev, "%s: Enter\n", __func__); + + if (ucontrol->value.integer.value[0] != SID_APPLY_FIR) { + dev_err(codec->dev, + "%s: ERROR: This control supports '%s' only!\n", + __func__, enum_sid_state[SID_APPLY_FIR]); + return -EIO; + } + + mutex_lock(&codec->mutex); + + sidconf = snd_soc_read(codec, AB8500_SIDFIRCONF); + if (((sidconf & BIT(AB8500_SIDFIRCONF_FIRSIDBUSY)) != 0)) { + if ((sidconf & BIT(AB8500_SIDFIRCONF_ENFIRSIDS)) == 0) { + dev_err(codec->dev, "%s: Sidetone busy while off!\n", + __func__); + status = -EPERM; + } else { + status = -EBUSY; + } + goto out; + } + + snd_soc_write(codec, AB8500_SIDFIRADR, 0); + + for (param = 0; param < AB8500_SID_FIR_COEFFS; param++) { + val = snd_soc_read(codec, drvdata->sid_fir_values[param]); + snd_soc_write(codec, AB8500_SIDFIRCOEF1, val >> 8 & 0xff); + snd_soc_write(codec, AB8500_SIDFIRCOEF2, val & 0xff); + } + + snd_soc_update_bits(codec, AB8500_SIDFIRADR, + BIT(AB8500_SIDFIRADR_FIRSIDSET), + BIT(AB8500_SIDFIRADR_FIRSIDSET)); + snd_soc_update_bits(codec, AB8500_SIDFIRADR, + BIT(AB8500_SIDFIRADR_FIRSIDSET), 0); + + drvdata->sid_status = SID_FIR_CONFIGURED; + +out: + mutex_unlock(&codec->mutex); + + dev_dbg(codec->dev, "%s: Exit\n", __func__); + + return status; +} + +static int anc_status_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); + + mutex_lock(&codec->mutex); + ucontrol->value.integer.value[0] = drvdata->anc_status; + mutex_unlock(&codec->mutex); + + return 0; +} + +static int anc_status_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); + struct device *dev = codec->dev; + bool apply_fir, apply_iir; + int req, status; + + dev_dbg(dev, "%s: Enter.\n", __func__); + + mutex_lock(&drvdata->anc_lock); + + req = ucontrol->value.integer.value[0]; + if (req != ANC_APPLY_FIR_IIR && req != ANC_APPLY_FIR && + req != ANC_APPLY_IIR) { + dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n", + __func__, enum_anc_state[req]); + status = -EINVAL; + goto cleanup; + } + apply_fir = req == ANC_APPLY_FIR || req == ANC_APPLY_FIR_IIR; + apply_iir = req == ANC_APPLY_IIR || req == ANC_APPLY_FIR_IIR; + + status = snd_soc_dapm_force_enable_pin(&codec->dapm, + "ANC Configure Input"); + if (status < 0) { + dev_err(dev, + "%s: ERROR: Failed to enable power (status = %d)!\n", + __func__, status); + goto cleanup; + } + snd_soc_dapm_sync(&codec->dapm); + + mutex_lock(&codec->mutex); + anc_configure(codec, apply_fir, apply_iir); + mutex_unlock(&codec->mutex); + + if (apply_fir) { + if (drvdata->anc_status == ANC_IIR_CONFIGURED) + drvdata->anc_status = ANC_FIR_IIR_CONFIGURED; + else if (drvdata->anc_status != ANC_FIR_IIR_CONFIGURED) + drvdata->anc_status = ANC_FIR_CONFIGURED; + } + if (apply_iir) { + if (drvdata->anc_status == ANC_FIR_CONFIGURED) + drvdata->anc_status = ANC_FIR_IIR_CONFIGURED; + else if (drvdata->anc_status != ANC_FIR_IIR_CONFIGURED) + drvdata->anc_status = ANC_IIR_CONFIGURED; + } + + status = snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input"); + snd_soc_dapm_sync(&codec->dapm); + +cleanup: + mutex_unlock(&drvdata->anc_lock); + + if (status < 0) + dev_err(dev, "%s: Unable to configure ANC! (status = %d)\n", + __func__, status); + + dev_dbg(dev, "%s: Exit.\n", __func__); + + return (status < 0) ? status : 1; +} + +static int filter_control_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct filter_control *fc = + (struct filter_control *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = fc->count; + uinfo->value.integer.min = fc->min; + uinfo->value.integer.max = fc->max; + + return 0; +} + +static int filter_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct filter_control *fc = + (struct filter_control *)kcontrol->private_value; + unsigned int i; + + mutex_lock(&codec->mutex); + for (i = 0; i < fc->count; i++) + ucontrol->value.integer.value[i] = fc->value[i]; + mutex_unlock(&codec->mutex); + + return 0; +} + +static int filter_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct filter_control *fc = + (struct filter_control *)kcontrol->private_value; + unsigned int i; + + mutex_lock(&codec->mutex); + for (i = 0; i < fc->count; i++) + fc->value[i] = ucontrol->value.integer.value[i]; + mutex_unlock(&codec->mutex); + + return 0; +} + +/* + * Controls - Non-DAPM ASoC + */ + +static DECLARE_TLV_DB_SCALE(adx_dig_gain_tlv, -3200, 100, 1); +/* -32dB = Mute */ + +static DECLARE_TLV_DB_SCALE(dax_dig_gain_tlv, -6300, 100, 1); +/* -63dB = Mute */ + +static DECLARE_TLV_DB_SCALE(hs_ear_dig_gain_tlv, -100, 100, 1); +/* -1dB = Mute */ + +static const unsigned int hs_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 3, TLV_DB_SCALE_ITEM(-3200, 400, 0), + 4, 15, TLV_DB_SCALE_ITEM(-1800, 200, 0), +}; + +static DECLARE_TLV_DB_SCALE(mic_gain_tlv, 0, 100, 0); + +static DECLARE_TLV_DB_SCALE(lin_gain_tlv, -1000, 200, 0); + +static DECLARE_TLV_DB_SCALE(lin2hs_gain_tlv, -3800, 200, 1); +/* -38dB = Mute */ + +static const char * const enum_hsfadspeed[] = {"2ms", "0.5ms", "10.6ms", + "5ms"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_hsfadspeed, + AB8500_DIGMICCONF, AB8500_DIGMICCONF_HSFADSPEED, enum_hsfadspeed); + +static const char * const enum_envdetthre[] = { + "250mV", "300mV", "350mV", "400mV", + "450mV", "500mV", "550mV", "600mV", + "650mV", "700mV", "750mV", "800mV", + "850mV", "900mV", "950mV", "1.00V" }; +static SOC_ENUM_SINGLE_DECL(soc_enum_envdeththre, + AB8500_ENVCPCONF, AB8500_ENVCPCONF_ENVDETHTHRE, enum_envdetthre); +static SOC_ENUM_SINGLE_DECL(soc_enum_envdetlthre, + AB8500_ENVCPCONF, AB8500_ENVCPCONF_ENVDETLTHRE, enum_envdetthre); +static const char * const enum_envdettime[] = { + "26.6us", "53.2us", "106us", "213us", + "426us", "851us", "1.70ms", "3.40ms", + "6.81ms", "13.6ms", "27.2ms", "54.5ms", + "109ms", "218ms", "436ms", "872ms" }; +static SOC_ENUM_SINGLE_DECL(soc_enum_envdettime, + AB8500_SIGENVCONF, AB8500_SIGENVCONF_ENVDETTIME, enum_envdettime); + +static const char * const enum_sinc31[] = {"Sinc 3", "Sinc 1"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_hsesinc, AB8500_HSLEARDIGGAIN, + AB8500_HSLEARDIGGAIN_HSSINC1, enum_sinc31); + +static const char * const enum_fadespeed[] = {"1ms", "4ms", "8ms", "16ms"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_fadespeed, AB8500_HSRDIGGAIN, + AB8500_HSRDIGGAIN_FADESPEED, enum_fadespeed); + +/* Earpiece */ + +static const char * const enum_lowpow[] = {"Normal", "Low Power"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_eardaclowpow, AB8500_ANACONF1, + AB8500_ANACONF1_EARDACLOWPOW, enum_lowpow); +static SOC_ENUM_SINGLE_DECL(soc_enum_eardrvlowpow, AB8500_ANACONF1, + AB8500_ANACONF1_EARDRVLOWPOW, enum_lowpow); + +static const char * const enum_av_mode[] = {"Audio", "Voice"}; +static SOC_ENUM_DOUBLE_DECL(soc_enum_ad12voice, AB8500_ADFILTCONF, + AB8500_ADFILTCONF_AD1VOICE, AB8500_ADFILTCONF_AD2VOICE, enum_av_mode); +static SOC_ENUM_DOUBLE_DECL(soc_enum_ad34voice, AB8500_ADFILTCONF, + AB8500_ADFILTCONF_AD3VOICE, AB8500_ADFILTCONF_AD4VOICE, enum_av_mode); + +/* DA */ + +static SOC_ENUM_SINGLE_DECL(soc_enum_da12voice, + AB8500_DASLOTCONF1, AB8500_DASLOTCONF1_DA12VOICE, + enum_av_mode); +static SOC_ENUM_SINGLE_DECL(soc_enum_da34voice, + AB8500_DASLOTCONF3, AB8500_DASLOTCONF3_DA34VOICE, + enum_av_mode); +static SOC_ENUM_SINGLE_DECL(soc_enum_da56voice, + AB8500_DASLOTCONF5, AB8500_DASLOTCONF5_DA56VOICE, + enum_av_mode); + +static const char * const enum_da2hslr[] = {"Sidetone", "Audio Path"}; +static SOC_ENUM_DOUBLE_DECL(soc_enum_da2hslr, AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_DATOHSLEN, + AB8500_DIGMULTCONF1_DATOHSREN, enum_da2hslr); + +static const char * const enum_sinc53[] = {"Sinc 5", "Sinc 3"}; +static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic12sinc, AB8500_DMICFILTCONF, + AB8500_DMICFILTCONF_DMIC1SINC3, + AB8500_DMICFILTCONF_DMIC2SINC3, enum_sinc53); +static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic34sinc, AB8500_DMICFILTCONF, + AB8500_DMICFILTCONF_DMIC3SINC3, + AB8500_DMICFILTCONF_DMIC4SINC3, enum_sinc53); +static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic56sinc, AB8500_DMICFILTCONF, + AB8500_DMICFILTCONF_DMIC5SINC3, + AB8500_DMICFILTCONF_DMIC6SINC3, enum_sinc53); + +/* Digital interface - DA from slot mapping */ +static const char * const enum_da_from_slot_map[] = {"SLOT0", + "SLOT1", + "SLOT2", + "SLOT3", + "SLOT4", + "SLOT5", + "SLOT6", + "SLOT7", + "SLOT8", + "SLOT9", + "SLOT10", + "SLOT11", + "SLOT12", + "SLOT13", + "SLOT14", + "SLOT15", + "SLOT16", + "SLOT17", + "SLOT18", + "SLOT19", + "SLOT20", + "SLOT21", + "SLOT22", + "SLOT23", + "SLOT24", + "SLOT25", + "SLOT26", + "SLOT27", + "SLOT28", + "SLOT29", + "SLOT30", + "SLOT31"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_da1slotmap, + AB8500_DASLOTCONF1, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da2slotmap, + AB8500_DASLOTCONF2, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da3slotmap, + AB8500_DASLOTCONF3, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da4slotmap, + AB8500_DASLOTCONF4, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da5slotmap, + AB8500_DASLOTCONF5, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da6slotmap, + AB8500_DASLOTCONF6, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da7slotmap, + AB8500_DASLOTCONF7, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da8slotmap, + AB8500_DASLOTCONF8, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); + +/* Digital interface - AD to slot mapping */ +static const char * const enum_ad_to_slot_map[] = {"AD_OUT1", + "AD_OUT2", + "AD_OUT3", + "AD_OUT4", + "AD_OUT5", + "AD_OUT6", + "AD_OUT7", + "AD_OUT8", + "zeroes", + "tristate"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot0map, + AB8500_ADSLOTSEL1, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot1map, + AB8500_ADSLOTSEL1, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot2map, + AB8500_ADSLOTSEL2, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot3map, + AB8500_ADSLOTSEL2, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot4map, + AB8500_ADSLOTSEL3, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot5map, + AB8500_ADSLOTSEL3, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot6map, + AB8500_ADSLOTSEL4, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot7map, + AB8500_ADSLOTSEL4, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot8map, + AB8500_ADSLOTSEL5, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot9map, + AB8500_ADSLOTSEL5, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot10map, + AB8500_ADSLOTSEL6, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot11map, + AB8500_ADSLOTSEL6, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot12map, + AB8500_ADSLOTSEL7, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot13map, + AB8500_ADSLOTSEL7, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot14map, + AB8500_ADSLOTSEL8, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot15map, + AB8500_ADSLOTSEL8, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot16map, + AB8500_ADSLOTSEL9, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot17map, + AB8500_ADSLOTSEL9, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot18map, + AB8500_ADSLOTSEL10, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot19map, + AB8500_ADSLOTSEL10, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot20map, + AB8500_ADSLOTSEL11, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot21map, + AB8500_ADSLOTSEL11, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot22map, + AB8500_ADSLOTSEL12, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot23map, + AB8500_ADSLOTSEL12, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot24map, + AB8500_ADSLOTSEL13, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot25map, + AB8500_ADSLOTSEL13, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot26map, + AB8500_ADSLOTSEL14, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot27map, + AB8500_ADSLOTSEL14, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot28map, + AB8500_ADSLOTSEL15, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot29map, + AB8500_ADSLOTSEL15, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot30map, + AB8500_ADSLOTSEL16, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot31map, + AB8500_ADSLOTSEL16, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); + +/* Digital interface - Burst mode */ +static const char * const enum_mask[] = {"Unmasked", "Masked"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_bfifomask, + AB8500_FIFOCONF1, AB8500_FIFOCONF1_BFIFOMASK, + enum_mask); +static const char * const enum_bitclk0[] = {"19_2_MHz", "38_4_MHz"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_bfifo19m2, + AB8500_FIFOCONF1, AB8500_FIFOCONF1_BFIFO19M2, + enum_bitclk0); +static const char * const enum_slavemaster[] = {"Slave", "Master"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_bfifomast, + AB8500_FIFOCONF3, AB8500_FIFOCONF3_BFIFOMAST_SHIFT, + enum_slavemaster); + +/* Sidetone */ +static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_sidstate, enum_sid_state); + +/* ANC */ +static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_ancstate, enum_anc_state); + +static struct snd_kcontrol_new ab8500_ctrls[] = { + /* Charge pump */ + SOC_ENUM("Charge Pump High Threshold For Low Voltage", + soc_enum_envdeththre), + SOC_ENUM("Charge Pump Low Threshold For Low Voltage", + soc_enum_envdetlthre), + SOC_SINGLE("Charge Pump Envelope Detection Switch", + AB8500_SIGENVCONF, AB8500_SIGENVCONF_ENVDETCPEN, + 1, 0), + SOC_ENUM("Charge Pump Envelope Detection Decay Time", + soc_enum_envdettime), + + /* Headset */ + SOC_ENUM("Headset Mode", soc_enum_da12voice), + SOC_SINGLE("Headset High Pass Switch", + AB8500_ANACONF1, AB8500_ANACONF1_HSHPEN, + 1, 0), + SOC_SINGLE("Headset Low Power Switch", + AB8500_ANACONF1, AB8500_ANACONF1_HSLOWPOW, + 1, 0), + SOC_SINGLE("Headset DAC Low Power Switch", + AB8500_ANACONF1, AB8500_ANACONF1_DACLOWPOW1, + 1, 0), + SOC_SINGLE("Headset DAC Drv Low Power Switch", + AB8500_ANACONF1, AB8500_ANACONF1_DACLOWPOW0, + 1, 0), + SOC_ENUM("Headset Fade Speed", soc_enum_hsfadspeed), + SOC_ENUM("Headset Source", soc_enum_da2hslr), + SOC_ENUM("Headset Filter", soc_enum_hsesinc), + SOC_DOUBLE_R_TLV("Headset Master Volume", + AB8500_DADIGGAIN1, AB8500_DADIGGAIN2, + 0, AB8500_DADIGGAINX_DAXGAIN_MAX, 1, dax_dig_gain_tlv), + SOC_DOUBLE_R_TLV("Headset Digital Volume", + AB8500_HSLEARDIGGAIN, AB8500_HSRDIGGAIN, + 0, AB8500_HSLEARDIGGAIN_HSLDGAIN_MAX, 1, hs_ear_dig_gain_tlv), + SOC_DOUBLE_TLV("Headset Volume", + AB8500_ANAGAIN3, + AB8500_ANAGAIN3_HSLGAIN, AB8500_ANAGAIN3_HSRGAIN, + AB8500_ANAGAIN3_HSXGAIN_MAX, 1, hs_gain_tlv), + + /* Earpiece */ + SOC_ENUM("Earpiece DAC Mode", + soc_enum_eardaclowpow), + SOC_ENUM("Earpiece DAC Drv Mode", + soc_enum_eardrvlowpow), + + /* HandsFree */ + SOC_ENUM("HF Mode", soc_enum_da34voice), + SOC_SINGLE("HF and Headset Swap Switch", + AB8500_DASLOTCONF1, AB8500_DASLOTCONF1_SWAPDA12_34, + 1, 0), + SOC_DOUBLE("HF Low EMI Mode Switch", + AB8500_CLASSDCONF1, + AB8500_CLASSDCONF1_HFLSWAPEN, AB8500_CLASSDCONF1_HFRSWAPEN, + 1, 0), + SOC_DOUBLE("HF FIR Bypass Switch", + AB8500_CLASSDCONF2, + AB8500_CLASSDCONF2_FIRBYP0, AB8500_CLASSDCONF2_FIRBYP1, + 1, 0), + SOC_DOUBLE("HF High Volume Switch", + AB8500_CLASSDCONF2, + AB8500_CLASSDCONF2_HIGHVOLEN0, AB8500_CLASSDCONF2_HIGHVOLEN1, + 1, 0), + SOC_SINGLE("HF L and R Bridge Switch", + AB8500_CLASSDCONF1, AB8500_CLASSDCONF1_PARLHF, + 1, 0), + SOC_DOUBLE_R_TLV("HF Master Volume", + AB8500_DADIGGAIN3, AB8500_DADIGGAIN4, + 0, AB8500_DADIGGAINX_DAXGAIN_MAX, 1, dax_dig_gain_tlv), + + /* Vibra */ + SOC_DOUBLE("Vibra High Volume Switch", + AB8500_CLASSDCONF2, + AB8500_CLASSDCONF2_HIGHVOLEN2, AB8500_CLASSDCONF2_HIGHVOLEN3, + 1, 0), + SOC_DOUBLE("Vibra Low EMI Mode Switch", + AB8500_CLASSDCONF1, + AB8500_CLASSDCONF1_VIB1SWAPEN, AB8500_CLASSDCONF1_VIB2SWAPEN, + 1, 0), + SOC_DOUBLE("Vibra FIR Bypass Switch", + AB8500_CLASSDCONF2, + AB8500_CLASSDCONF2_FIRBYP2, AB8500_CLASSDCONF2_FIRBYP3, + 1, 0), + SOC_ENUM("Vibra Mode", soc_enum_da56voice), + SOC_DOUBLE_R("Vibra PWM Duty Cycle N", + AB8500_PWMGENCONF3, AB8500_PWMGENCONF5, + AB8500_PWMGENCONFX_PWMVIBXDUTCYC, + AB8500_PWMGENCONFX_PWMVIBXDUTCYC_MAX, 0), + SOC_DOUBLE_R("Vibra PWM Duty Cycle P", + AB8500_PWMGENCONF2, AB8500_PWMGENCONF4, + AB8500_PWMGENCONFX_PWMVIBXDUTCYC, + AB8500_PWMGENCONFX_PWMVIBXDUTCYC_MAX, 0), + SOC_SINGLE("Vibra 1 and 2 Bridge Switch", + AB8500_CLASSDCONF1, AB8500_CLASSDCONF1_PARLVIB, + 1, 0), + SOC_DOUBLE_R_TLV("Vibra Master Volume", + AB8500_DADIGGAIN5, AB8500_DADIGGAIN6, + 0, AB8500_DADIGGAINX_DAXGAIN_MAX, 1, dax_dig_gain_tlv), + + /* HandsFree, Vibra */ + SOC_SINGLE("ClassD High Pass Volume", + AB8500_CLASSDCONF3, AB8500_CLASSDCONF3_DITHHPGAIN, + AB8500_CLASSDCONF3_DITHHPGAIN_MAX, 0), + SOC_SINGLE("ClassD White Volume", + AB8500_CLASSDCONF3, AB8500_CLASSDCONF3_DITHWGAIN, + AB8500_CLASSDCONF3_DITHWGAIN_MAX, 0), + + /* Mic 1, Mic 2, LineIn */ + SOC_DOUBLE_R_TLV("Mic Master Volume", + AB8500_ADDIGGAIN3, AB8500_ADDIGGAIN4, + 0, AB8500_ADDIGGAINX_ADXGAIN_MAX, 1, adx_dig_gain_tlv), + + /* Mic 1 */ + SOC_SINGLE_TLV("Mic 1", + AB8500_ANAGAIN1, + AB8500_ANAGAINX_MICXGAIN, + AB8500_ANAGAINX_MICXGAIN_MAX, 0, mic_gain_tlv), + SOC_SINGLE("Mic 1 Low Power Switch", + AB8500_ANAGAIN1, AB8500_ANAGAINX_LOWPOWMICX, + 1, 0), + + /* Mic 2 */ + SOC_DOUBLE("Mic High Pass Switch", + AB8500_ADFILTCONF, + AB8500_ADFILTCONF_AD3NH, AB8500_ADFILTCONF_AD4NH, + 1, 1), + SOC_ENUM("Mic Mode", soc_enum_ad34voice), + SOC_ENUM("Mic Filter", soc_enum_dmic34sinc), + SOC_SINGLE_TLV("Mic 2", + AB8500_ANAGAIN2, + AB8500_ANAGAINX_MICXGAIN, + AB8500_ANAGAINX_MICXGAIN_MAX, 0, mic_gain_tlv), + SOC_SINGLE("Mic 2 Low Power Switch", + AB8500_ANAGAIN2, AB8500_ANAGAINX_LOWPOWMICX, + 1, 0), + + /* LineIn */ + SOC_DOUBLE("LineIn High Pass Switch", + AB8500_ADFILTCONF, + AB8500_ADFILTCONF_AD1NH, AB8500_ADFILTCONF_AD2NH, + 1, 1), + SOC_ENUM("LineIn Filter", soc_enum_dmic12sinc), + SOC_ENUM("LineIn Mode", soc_enum_ad12voice), + SOC_DOUBLE_R_TLV("LineIn Master Volume", + AB8500_ADDIGGAIN1, AB8500_ADDIGGAIN2, + 0, AB8500_ADDIGGAINX_ADXGAIN_MAX, 1, adx_dig_gain_tlv), + SOC_DOUBLE_TLV("LineIn", + AB8500_ANAGAIN4, + AB8500_ANAGAIN4_LINLGAIN, AB8500_ANAGAIN4_LINRGAIN, + AB8500_ANAGAIN4_LINXGAIN_MAX, 0, lin_gain_tlv), + SOC_DOUBLE_R_TLV("LineIn to Headset Volume", + AB8500_DIGLINHSLGAIN, AB8500_DIGLINHSRGAIN, + AB8500_DIGLINHSXGAIN_LINTOHSXGAIN, + AB8500_DIGLINHSXGAIN_LINTOHSXGAIN_MAX, + 1, lin2hs_gain_tlv), + + /* DMic */ + SOC_ENUM("DMic Filter", soc_enum_dmic56sinc), + SOC_DOUBLE_R_TLV("DMic Master Volume", + AB8500_ADDIGGAIN5, AB8500_ADDIGGAIN6, + 0, AB8500_ADDIGGAINX_ADXGAIN_MAX, 1, adx_dig_gain_tlv), + + /* Digital gains */ + SOC_ENUM("Digital Gain Fade Speed", soc_enum_fadespeed), + + /* Analog loopback */ + SOC_DOUBLE_R_TLV("Analog Loopback Volume", + AB8500_ADDIGLOOPGAIN1, AB8500_ADDIGLOOPGAIN2, + 0, AB8500_ADDIGLOOPGAINX_ADXLBGAIN_MAX, 1, dax_dig_gain_tlv), + + /* Digital interface - DA from slot mapping */ + SOC_ENUM("Digital Interface DA 1 From Slot Map", soc_enum_da1slotmap), + SOC_ENUM("Digital Interface DA 2 From Slot Map", soc_enum_da2slotmap), + SOC_ENUM("Digital Interface DA 3 From Slot Map", soc_enum_da3slotmap), + SOC_ENUM("Digital Interface DA 4 From Slot Map", soc_enum_da4slotmap), + SOC_ENUM("Digital Interface DA 5 From Slot Map", soc_enum_da5slotmap), + SOC_ENUM("Digital Interface DA 6 From Slot Map", soc_enum_da6slotmap), + SOC_ENUM("Digital Interface DA 7 From Slot Map", soc_enum_da7slotmap), + SOC_ENUM("Digital Interface DA 8 From Slot Map", soc_enum_da8slotmap), + + /* Digital interface - AD to slot mapping */ + SOC_ENUM("Digital Interface AD To Slot 0 Map", soc_enum_adslot0map), + SOC_ENUM("Digital Interface AD To Slot 1 Map", soc_enum_adslot1map), + SOC_ENUM("Digital Interface AD To Slot 2 Map", soc_enum_adslot2map), + SOC_ENUM("Digital Interface AD To Slot 3 Map", soc_enum_adslot3map), + SOC_ENUM("Digital Interface AD To Slot 4 Map", soc_enum_adslot4map), + SOC_ENUM("Digital Interface AD To Slot 5 Map", soc_enum_adslot5map), + SOC_ENUM("Digital Interface AD To Slot 6 Map", soc_enum_adslot6map), + SOC_ENUM("Digital Interface AD To Slot 7 Map", soc_enum_adslot7map), + SOC_ENUM("Digital Interface AD To Slot 8 Map", soc_enum_adslot8map), + SOC_ENUM("Digital Interface AD To Slot 9 Map", soc_enum_adslot9map), + SOC_ENUM("Digital Interface AD To Slot 10 Map", soc_enum_adslot10map), + SOC_ENUM("Digital Interface AD To Slot 11 Map", soc_enum_adslot11map), + SOC_ENUM("Digital Interface AD To Slot 12 Map", soc_enum_adslot12map), + SOC_ENUM("Digital Interface AD To Slot 13 Map", soc_enum_adslot13map), + SOC_ENUM("Digital Interface AD To Slot 14 Map", soc_enum_adslot14map), + SOC_ENUM("Digital Interface AD To Slot 15 Map", soc_enum_adslot15map), + SOC_ENUM("Digital Interface AD To Slot 16 Map", soc_enum_adslot16map), + SOC_ENUM("Digital Interface AD To Slot 17 Map", soc_enum_adslot17map), + SOC_ENUM("Digital Interface AD To Slot 18 Map", soc_enum_adslot18map), + SOC_ENUM("Digital Interface AD To Slot 19 Map", soc_enum_adslot19map), + SOC_ENUM("Digital Interface AD To Slot 20 Map", soc_enum_adslot20map), + SOC_ENUM("Digital Interface AD To Slot 21 Map", soc_enum_adslot21map), + SOC_ENUM("Digital Interface AD To Slot 22 Map", soc_enum_adslot22map), + SOC_ENUM("Digital Interface AD To Slot 23 Map", soc_enum_adslot23map), + SOC_ENUM("Digital Interface AD To Slot 24 Map", soc_enum_adslot24map), + SOC_ENUM("Digital Interface AD To Slot 25 Map", soc_enum_adslot25map), + SOC_ENUM("Digital Interface AD To Slot 26 Map", soc_enum_adslot26map), + SOC_ENUM("Digital Interface AD To Slot 27 Map", soc_enum_adslot27map), + SOC_ENUM("Digital Interface AD To Slot 28 Map", soc_enum_adslot28map), + SOC_ENUM("Digital Interface AD To Slot 29 Map", soc_enum_adslot29map), + SOC_ENUM("Digital Interface AD To Slot 30 Map", soc_enum_adslot30map), + SOC_ENUM("Digital Interface AD To Slot 31 Map", soc_enum_adslot31map), + + /* Digital interface - Loopback */ + SOC_SINGLE("Digital Interface AD 1 Loopback Switch", + AB8500_DASLOTCONF1, AB8500_DASLOTCONF1_DAI7TOADO1, + 1, 0), + SOC_SINGLE("Digital Interface AD 2 Loopback Switch", + AB8500_DASLOTCONF2, AB8500_DASLOTCONF2_DAI8TOADO2, + 1, 0), + SOC_SINGLE("Digital Interface AD 3 Loopback Switch", + AB8500_DASLOTCONF3, AB8500_DASLOTCONF3_DAI7TOADO3, + 1, 0), + SOC_SINGLE("Digital Interface AD 4 Loopback Switch", + AB8500_DASLOTCONF4, AB8500_DASLOTCONF4_DAI8TOADO4, + 1, 0), + SOC_SINGLE("Digital Interface AD 5 Loopback Switch", + AB8500_DASLOTCONF5, AB8500_DASLOTCONF5_DAI7TOADO5, + 1, 0), + SOC_SINGLE("Digital Interface AD 6 Loopback Switch", + AB8500_DASLOTCONF6, AB8500_DASLOTCONF6_DAI8TOADO6, + 1, 0), + SOC_SINGLE("Digital Interface AD 7 Loopback Switch", + AB8500_DASLOTCONF7, AB8500_DASLOTCONF7_DAI8TOADO7, + 1, 0), + SOC_SINGLE("Digital Interface AD 8 Loopback Switch", + AB8500_DASLOTCONF8, AB8500_DASLOTCONF8_DAI7TOADO8, + 1, 0), + + /* Digital interface - Burst FIFO */ + SOC_SINGLE("Digital Interface 0 FIFO Enable Switch", + AB8500_DIGIFCONF3, AB8500_DIGIFCONF3_IF0BFIFOEN, + 1, 0), + SOC_ENUM("Burst FIFO Mask", soc_enum_bfifomask), + SOC_ENUM("Burst FIFO Bit-clock Frequency", soc_enum_bfifo19m2), + SOC_SINGLE("Burst FIFO Threshold", + AB8500_FIFOCONF1, AB8500_FIFOCONF1_BFIFOINT_SHIFT, + AB8500_FIFOCONF1_BFIFOINT_MAX, 0), + SOC_SINGLE("Burst FIFO Length", + AB8500_FIFOCONF2, AB8500_FIFOCONF2_BFIFOTX_SHIFT, + AB8500_FIFOCONF2_BFIFOTX_MAX, 0), + SOC_SINGLE("Burst FIFO EOS Extra Slots", + AB8500_FIFOCONF3, AB8500_FIFOCONF3_BFIFOEXSL_SHIFT, + AB8500_FIFOCONF3_BFIFOEXSL_MAX, 0), + SOC_SINGLE("Burst FIFO FS Extra Bit-clocks", + AB8500_FIFOCONF3, AB8500_FIFOCONF3_PREBITCLK0_SHIFT, + AB8500_FIFOCONF3_PREBITCLK0_MAX, 0), + SOC_ENUM("Burst FIFO Interface Mode", soc_enum_bfifomast), + + SOC_SINGLE("Burst FIFO Interface Switch", + AB8500_FIFOCONF3, AB8500_FIFOCONF3_BFIFORUN_SHIFT, + 1, 0), + SOC_SINGLE("Burst FIFO Switch Frame Number", + AB8500_FIFOCONF4, AB8500_FIFOCONF4_BFIFOFRAMSW_SHIFT, + AB8500_FIFOCONF4_BFIFOFRAMSW_MAX, 0), + SOC_SINGLE("Burst FIFO Wake Up Delay", + AB8500_FIFOCONF5, AB8500_FIFOCONF5_BFIFOWAKEUP_SHIFT, + AB8500_FIFOCONF5_BFIFOWAKEUP_MAX, 0), + SOC_SINGLE("Burst FIFO Samples In FIFO", + AB8500_FIFOCONF6, AB8500_FIFOCONF6_BFIFOSAMPLE_SHIFT, + AB8500_FIFOCONF6_BFIFOSAMPLE_MAX, 0), + + /* ANC */ + SOC_ENUM_EXT("ANC Status", soc_enum_ancstate, + anc_status_control_get, anc_status_control_put), + SOC_SINGLE_XR_SX("ANC Warp Delay Shift", + AB8500_ANCCONF2, 1, AB8500_ANCCONF2_SHIFT, + AB8500_ANCCONF2_MIN, AB8500_ANCCONF2_MAX, 0), + SOC_SINGLE_XR_SX("ANC FIR Output Shift", + AB8500_ANCCONF3, 1, AB8500_ANCCONF3_SHIFT, + AB8500_ANCCONF3_MIN, AB8500_ANCCONF3_MAX, 0), + SOC_SINGLE_XR_SX("ANC IIR Output Shift", + AB8500_ANCCONF4, 1, AB8500_ANCCONF4_SHIFT, + AB8500_ANCCONF4_MIN, AB8500_ANCCONF4_MAX, 0), + SOC_SINGLE_XR_SX("ANC Warp Delay", + AB8500_ANCCONF9, 2, AB8500_ANC_WARP_DELAY_SHIFT, + AB8500_ANC_WARP_DELAY_MIN, AB8500_ANC_WARP_DELAY_MAX, 0), + + /* Sidetone */ + SOC_ENUM_EXT("Sidetone Status", soc_enum_sidstate, + sid_status_control_get, sid_status_control_put), + SOC_SINGLE_STROBE("Sidetone Reset", + AB8500_SIDFIRADR, AB8500_SIDFIRADR_FIRSIDSET, 0), +}; + +static struct snd_kcontrol_new ab8500_filter_controls[] = { + AB8500_FILTER_CONTROL("ANC FIR Coefficients", AB8500_ANC_FIR_COEFFS, + AB8500_ANC_FIR_COEFF_MIN, AB8500_ANC_FIR_COEFF_MAX), + AB8500_FILTER_CONTROL("ANC IIR Coefficients", AB8500_ANC_IIR_COEFFS, + AB8500_ANC_IIR_COEFF_MIN, AB8500_ANC_IIR_COEFF_MAX), + AB8500_FILTER_CONTROL("Sidetone FIR Coefficients", + AB8500_SID_FIR_COEFFS, AB8500_SID_FIR_COEFF_MIN, + AB8500_SID_FIR_COEFF_MAX) +}; +enum ab8500_filter { + AB8500_FILTER_ANC_FIR = 0, + AB8500_FILTER_ANC_IIR = 1, + AB8500_FILTER_SID_FIR = 2, +}; + +/* + * Extended interface for codec-driver + */ + +static int ab8500_audio_init_audioblock(struct snd_soc_codec *codec) +{ + int status; + + dev_dbg(codec->dev, "%s: Enter.\n", __func__); + + /* Reset audio-registers and disable 32kHz-clock output 2 */ + status = ab8500_sysctrl_write(AB8500_STW4500CTRL3, + AB8500_STW4500CTRL3_CLK32KOUT2DIS | + AB8500_STW4500CTRL3_RESETAUDN, + AB8500_STW4500CTRL3_RESETAUDN); + if (status < 0) + return status; + + return 0; +} + +static int ab8500_audio_setup_mics(struct snd_soc_codec *codec, + struct amic_settings *amics) +{ + u8 value8; + unsigned int value; + int status; + const struct snd_soc_dapm_route *route; + + dev_dbg(codec->dev, "%s: Enter.\n", __func__); + + /* Set DMic-clocks to outputs */ + status = abx500_get_register_interruptible(codec->dev, (u8)AB8500_MISC, + (u8)AB8500_GPIO_DIR4_REG, + &value8); + if (status < 0) + return status; + value = value8 | GPIO27_DIR_OUTPUT | GPIO29_DIR_OUTPUT | + GPIO31_DIR_OUTPUT; + status = abx500_set_register_interruptible(codec->dev, + (u8)AB8500_MISC, + (u8)AB8500_GPIO_DIR4_REG, + value); + if (status < 0) + return status; + + /* Attach regulators to AMic DAPM-paths */ + dev_dbg(codec->dev, "%s: Mic 1a regulator: %s\n", __func__, + amic_micbias_str(amics->mic1a_micbias)); + route = &ab8500_dapm_routes_mic1a_vamicx[amics->mic1a_micbias]; + status = snd_soc_dapm_add_routes(&codec->dapm, route, 1); + dev_dbg(codec->dev, "%s: Mic 1b regulator: %s\n", __func__, + amic_micbias_str(amics->mic1b_micbias)); + route = &ab8500_dapm_routes_mic1b_vamicx[amics->mic1b_micbias]; + status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1); + dev_dbg(codec->dev, "%s: Mic 2 regulator: %s\n", __func__, + amic_micbias_str(amics->mic2_micbias)); + route = &ab8500_dapm_routes_mic2_vamicx[amics->mic2_micbias]; + status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1); + if (status < 0) { + dev_err(codec->dev, + "%s: Failed to add AMic-regulator DAPM-routes (%d).\n", + __func__, status); + return status; + } + + /* Set AMic-configuration */ + dev_dbg(codec->dev, "%s: Mic 1 mic-type: %s\n", __func__, + amic_type_str(amics->mic1_type)); + snd_soc_update_bits(codec, AB8500_ANAGAIN1, AB8500_ANAGAINX_ENSEMICX, + amics->mic1_type == AMIC_TYPE_DIFFERENTIAL ? + 0 : AB8500_ANAGAINX_ENSEMICX); + dev_dbg(codec->dev, "%s: Mic 2 mic-type: %s\n", __func__, + amic_type_str(amics->mic2_type)); + snd_soc_update_bits(codec, AB8500_ANAGAIN2, AB8500_ANAGAINX_ENSEMICX, + amics->mic2_type == AMIC_TYPE_DIFFERENTIAL ? + 0 : AB8500_ANAGAINX_ENSEMICX); + + return 0; +} +EXPORT_SYMBOL_GPL(ab8500_audio_setup_mics); + +static int ab8500_audio_set_ear_cmv(struct snd_soc_codec *codec, + enum ear_cm_voltage ear_cmv) +{ + char *cmv_str; + + switch (ear_cmv) { + case EAR_CMV_0_95V: + cmv_str = "0.95V"; + break; + case EAR_CMV_1_10V: + cmv_str = "1.10V"; + break; + case EAR_CMV_1_27V: + cmv_str = "1.27V"; + break; + case EAR_CMV_1_58V: + cmv_str = "1.58V"; + break; + default: + dev_err(codec->dev, + "%s: Unknown earpiece CM-voltage (%d)!\n", + __func__, (int)ear_cmv); + return -EINVAL; + } + dev_dbg(codec->dev, "%s: Earpiece CM-voltage: %s\n", __func__, + cmv_str); + snd_soc_update_bits(codec, AB8500_ANACONF1, AB8500_ANACONF1_EARSELCM, + ear_cmv); + + return 0; +} +EXPORT_SYMBOL_GPL(ab8500_audio_set_ear_cmv); + +static int ab8500_audio_set_bit_delay(struct snd_soc_dai *dai, + unsigned int delay) +{ + unsigned int mask, val; + struct snd_soc_codec *codec = dai->codec; + + mask = BIT(AB8500_DIGIFCONF2_IF0DEL); + val = 0; + + switch (delay) { + case 0: + break; + case 1: + val |= BIT(AB8500_DIGIFCONF2_IF0DEL); + break; + default: + dev_err(dai->codec->dev, + "%s: ERROR: Unsupported bit-delay (0x%x)!\n", + __func__, delay); + return -EINVAL; + } + + dev_dbg(dai->codec->dev, "%s: IF0 Bit-delay: %d bits.\n", + __func__, delay); + snd_soc_update_bits(codec, AB8500_DIGIFCONF2, mask, val); + + return 0; +} + +/* Gates clocking according format mask */ +static int ab8500_codec_set_dai_clock_gate(struct snd_soc_codec *codec, + unsigned int fmt) +{ + unsigned int mask; + unsigned int val; + + mask = BIT(AB8500_DIGIFCONF1_ENMASTGEN) | + BIT(AB8500_DIGIFCONF1_ENFSBITCLK0); + + val = BIT(AB8500_DIGIFCONF1_ENMASTGEN); + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: /* continuous clock */ + dev_dbg(codec->dev, "%s: IF0 Clock is continuous.\n", + __func__); + val |= BIT(AB8500_DIGIFCONF1_ENFSBITCLK0); + break; + case SND_SOC_DAIFMT_GATED: /* clock is gated */ + dev_dbg(codec->dev, "%s: IF0 Clock is gated.\n", + __func__); + break; + default: + dev_err(codec->dev, + "%s: ERROR: Unsupported clock mask (0x%x)!\n", + __func__, fmt & SND_SOC_DAIFMT_CLOCK_MASK); + return -EINVAL; + } + + snd_soc_update_bits(codec, AB8500_DIGIFCONF1, mask, val); + + return 0; +} + +static int ab8500_codec_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + unsigned int mask; + unsigned int val; + struct snd_soc_codec *codec = dai->codec; + int status; + + dev_dbg(codec->dev, "%s: Enter (fmt = 0x%x)\n", __func__, fmt); + + mask = BIT(AB8500_DIGIFCONF3_IF1DATOIF0AD) | + BIT(AB8500_DIGIFCONF3_IF1CLKTOIF0CLK) | + BIT(AB8500_DIGIFCONF3_IF0BFIFOEN) | + BIT(AB8500_DIGIFCONF3_IF0MASTER); + val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & FRM master */ + dev_dbg(dai->codec->dev, + "%s: IF0 Master-mode: AB8500 master.\n", __func__); + val |= BIT(AB8500_DIGIFCONF3_IF0MASTER); + break; + case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & FRM slave */ + dev_dbg(dai->codec->dev, + "%s: IF0 Master-mode: AB8500 slave.\n", __func__); + break; + case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & FRM master */ + case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */ + dev_err(dai->codec->dev, + "%s: ERROR: The device is either a master or a slave.\n", + __func__); + default: + dev_err(dai->codec->dev, + "%s: ERROR: Unsupporter master mask 0x%x\n", + __func__, fmt & SND_SOC_DAIFMT_MASTER_MASK); + return -EINVAL; + break; + } + + snd_soc_update_bits(codec, AB8500_DIGIFCONF3, mask, val); + + /* Set clock gating */ + status = ab8500_codec_set_dai_clock_gate(codec, fmt); + if (status) { + dev_err(dai->codec->dev, + "%s: ERRROR: Failed to set clock gate (%d).\n", + __func__, status); + return status; + } + + /* Setting data transfer format */ + + mask = BIT(AB8500_DIGIFCONF2_IF0FORMAT0) | + BIT(AB8500_DIGIFCONF2_IF0FORMAT1) | + BIT(AB8500_DIGIFCONF2_FSYNC0P) | + BIT(AB8500_DIGIFCONF2_BITCLK0P); + val = 0; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: /* I2S mode */ + dev_dbg(dai->codec->dev, "%s: IF0 Protocol: I2S\n", __func__); + val |= BIT(AB8500_DIGIFCONF2_IF0FORMAT1); + ab8500_audio_set_bit_delay(dai, 0); + break; + + case SND_SOC_DAIFMT_DSP_A: /* L data MSB after FRM LRC */ + dev_dbg(dai->codec->dev, + "%s: IF0 Protocol: DSP A (TDM)\n", __func__); + val |= BIT(AB8500_DIGIFCONF2_IF0FORMAT0); + ab8500_audio_set_bit_delay(dai, 1); + break; + + case SND_SOC_DAIFMT_DSP_B: /* L data MSB during FRM LRC */ + dev_dbg(dai->codec->dev, + "%s: IF0 Protocol: DSP B (TDM)\n", __func__); + val |= BIT(AB8500_DIGIFCONF2_IF0FORMAT0); + ab8500_audio_set_bit_delay(dai, 0); + break; + + default: + dev_err(dai->codec->dev, + "%s: ERROR: Unsupported format (0x%x)!\n", + __func__, fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: /* normal bit clock + frame */ + dev_dbg(dai->codec->dev, + "%s: IF0: Normal bit clock, normal frame\n", + __func__); + break; + case SND_SOC_DAIFMT_NB_IF: /* normal BCLK + inv FRM */ + dev_dbg(dai->codec->dev, + "%s: IF0: Normal bit clock, inverted frame\n", + __func__); + val |= BIT(AB8500_DIGIFCONF2_FSYNC0P); + break; + case SND_SOC_DAIFMT_IB_NF: /* invert BCLK + nor FRM */ + dev_dbg(dai->codec->dev, + "%s: IF0: Inverted bit clock, normal frame\n", + __func__); + val |= BIT(AB8500_DIGIFCONF2_BITCLK0P); + break; + case SND_SOC_DAIFMT_IB_IF: /* invert BCLK + FRM */ + dev_dbg(dai->codec->dev, + "%s: IF0: Inverted bit clock, inverted frame\n", + __func__); + val |= BIT(AB8500_DIGIFCONF2_FSYNC0P); + val |= BIT(AB8500_DIGIFCONF2_BITCLK0P); + break; + default: + dev_err(dai->codec->dev, + "%s: ERROR: Unsupported INV mask 0x%x\n", + __func__, fmt & SND_SOC_DAIFMT_INV_MASK); + return -EINVAL; + } + + snd_soc_update_bits(codec, AB8500_DIGIFCONF2, mask, val); + + return 0; +} + +static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val, mask, slots_active; + + mask = BIT(AB8500_DIGIFCONF2_IF0WL0) | + BIT(AB8500_DIGIFCONF2_IF0WL1); + val = 0; + + switch (slot_width) { + case 16: + break; + case 20: + val |= BIT(AB8500_DIGIFCONF2_IF0WL0); + break; + case 24: + val |= BIT(AB8500_DIGIFCONF2_IF0WL1); + break; + case 32: + val |= BIT(AB8500_DIGIFCONF2_IF0WL1) | + BIT(AB8500_DIGIFCONF2_IF0WL0); + break; + default: + dev_err(dai->codec->dev, "%s: Unsupported slot-width 0x%x\n", + __func__, slot_width); + return -EINVAL; + } + + dev_dbg(dai->codec->dev, "%s: IF0 slot-width: %d bits.\n", + __func__, slot_width); + snd_soc_update_bits(codec, AB8500_DIGIFCONF2, mask, val); + + /* Setup TDM clocking according to slot count */ + dev_dbg(dai->codec->dev, "%s: Slots, total: %d\n", __func__, slots); + mask = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS0) | + BIT(AB8500_DIGIFCONF1_IF0BITCLKOS1); + switch (slots) { + case 2: + val = AB8500_MASK_NONE; + break; + case 4: + val = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS0); + break; + case 8: + val = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS1); + break; + case 16: + val = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS0) | + BIT(AB8500_DIGIFCONF1_IF0BITCLKOS1); + break; + default: + dev_err(dai->codec->dev, + "%s: ERROR: Unsupported number of slots (%d)!\n", + __func__, slots); + return -EINVAL; + } + snd_soc_update_bits(codec, AB8500_DIGIFCONF1, mask, val); + + /* Setup TDM DA according to active tx slots */ + mask = AB8500_DASLOTCONFX_SLTODAX_MASK; + slots_active = hweight32(tx_mask); + dev_dbg(dai->codec->dev, "%s: Slots, active, TX: %d\n", __func__, + slots_active); + switch (slots_active) { + case 0: + break; + case 1: + /* Slot 9 -> DA_IN1 & DA_IN3 */ + snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, 11); + snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, 11); + snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, 11); + snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, 11); + break; + case 2: + /* Slot 9 -> DA_IN1 & DA_IN3, Slot 11 -> DA_IN2 & DA_IN4 */ + snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, 9); + snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, 9); + snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, 11); + snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, 11); + + break; + case 8: + dev_dbg(dai->codec->dev, + "%s: In 8-channel mode DA-from-slot mapping is set manually.", + __func__); + break; + default: + dev_err(dai->codec->dev, + "%s: Unsupported number of active TX-slots (%d)!\n", + __func__, slots_active); + return -EINVAL; + } + + /* Setup TDM AD according to active RX-slots */ + slots_active = hweight32(rx_mask); + dev_dbg(dai->codec->dev, "%s: Slots, active, RX: %d\n", __func__, + slots_active); + switch (slots_active) { + case 0: + break; + case 1: + /* AD_OUT3 -> slot 0 & 1 */ + snd_soc_update_bits(codec, AB8500_ADSLOTSEL1, AB8500_MASK_ALL, + AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN | + AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD); + break; + case 2: + /* AD_OUT3 -> slot 0, AD_OUT2 -> slot 1 */ + snd_soc_update_bits(codec, + AB8500_ADSLOTSEL1, + AB8500_MASK_ALL, + AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN | + AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD); + break; + case 8: + dev_dbg(dai->codec->dev, + "%s: In 8-channel mode AD-to-slot mapping is set manually.", + __func__); + break; + default: + dev_err(dai->codec->dev, + "%s: Unsupported number of active RX-slots (%d)!\n", + __func__, slots_active); + return -EINVAL; + } + + return 0; +} + +struct snd_soc_dai_driver ab8500_codec_dai[] = { + { + .name = "ab8500-codec-dai.0", + .id = 0, + .playback = { + .stream_name = "ab8500_0p", + .channels_min = 1, + .channels_max = 8, + .rates = AB8500_SUPPORTED_RATE, + .formats = AB8500_SUPPORTED_FMT, + }, + .ops = (struct snd_soc_dai_ops[]) { + { + .set_tdm_slot = ab8500_codec_set_dai_tdm_slot, + .set_fmt = ab8500_codec_set_dai_fmt, + } + }, + .symmetric_rates = 1 + }, + { + .name = "ab8500-codec-dai.1", + .id = 1, + .capture = { + .stream_name = "ab8500_0c", + .channels_min = 1, + .channels_max = 8, + .rates = AB8500_SUPPORTED_RATE, + .formats = AB8500_SUPPORTED_FMT, + }, + .ops = (struct snd_soc_dai_ops[]) { + { + .set_tdm_slot = ab8500_codec_set_dai_tdm_slot, + .set_fmt = ab8500_codec_set_dai_fmt, + } + }, + .symmetric_rates = 1 + } +}; + +static int ab8500_codec_probe(struct snd_soc_codec *codec) +{ + struct device *dev = codec->dev; + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev); + struct ab8500_platform_data *pdata; + struct filter_control *fc; + int status; + + dev_dbg(dev, "%s: Enter.\n", __func__); + + /* Setup AB8500 according to board-settings */ + pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent); + status = ab8500_audio_setup_mics(codec, &pdata->codec->amics); + if (status < 0) { + pr_err("%s: Failed to setup mics (%d)!\n", __func__, status); + return status; + } + status = ab8500_audio_set_ear_cmv(codec, pdata->codec->ear_cmv); + if (status < 0) { + pr_err("%s: Failed to set earpiece CM-voltage (%d)!\n", + __func__, status); + return status; + } + + status = ab8500_audio_init_audioblock(codec); + if (status < 0) { + dev_err(dev, "%s: failed to init audio-block (%d)!\n", + __func__, status); + return status; + } + + /* Override HW-defaults */ + ab8500_codec_write_reg(codec, + AB8500_ANACONF5, + BIT(AB8500_ANACONF5_HSAUTOEN)); + ab8500_codec_write_reg(codec, + AB8500_SHORTCIRCONF, + BIT(AB8500_SHORTCIRCONF_HSZCDDIS)); + + /* Add filter controls */ + status = snd_soc_add_codec_controls(codec, ab8500_filter_controls, + ARRAY_SIZE(ab8500_filter_controls)); + if (status < 0) { + dev_err(dev, + "%s: failed to add ab8500 filter controls (%d).\n", + __func__, status); + return status; + } + fc = (struct filter_control *) + &ab8500_filter_controls[AB8500_FILTER_ANC_FIR].private_value; + drvdata->anc_fir_values = (long *)fc->value; + fc = (struct filter_control *) + &ab8500_filter_controls[AB8500_FILTER_ANC_IIR].private_value; + drvdata->anc_iir_values = (long *)fc->value; + fc = (struct filter_control *) + &ab8500_filter_controls[AB8500_FILTER_SID_FIR].private_value; + drvdata->sid_fir_values = (long *)fc->value; + + (void)snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input"); + + mutex_init(&drvdata->anc_lock); + + return status; +} + +static struct snd_soc_codec_driver ab8500_codec_driver = { + .probe = ab8500_codec_probe, + .read = ab8500_codec_read_reg, + .write = ab8500_codec_write_reg, + .reg_word_size = sizeof(u8), + .controls = ab8500_ctrls, + .num_controls = ARRAY_SIZE(ab8500_ctrls), + .dapm_widgets = ab8500_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ab8500_dapm_widgets), + .dapm_routes = ab8500_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ab8500_dapm_routes), +}; + +static int __devinit ab8500_codec_driver_probe(struct platform_device *pdev) +{ + int status; + struct ab8500_codec_drvdata *drvdata; + + dev_dbg(&pdev->dev, "%s: Enter.\n", __func__); + + /* Create driver private-data struct */ + drvdata = devm_kzalloc(&pdev->dev, sizeof(struct ab8500_codec_drvdata), + GFP_KERNEL); + drvdata->sid_status = SID_UNCONFIGURED; + drvdata->anc_status = ANC_UNCONFIGURED; + dev_set_drvdata(&pdev->dev, drvdata); + + dev_dbg(&pdev->dev, "%s: Register codec.\n", __func__); + status = snd_soc_register_codec(&pdev->dev, &ab8500_codec_driver, + ab8500_codec_dai, + ARRAY_SIZE(ab8500_codec_dai)); + if (status < 0) + dev_err(&pdev->dev, + "%s: Error: Failed to register codec (%d).\n", + __func__, status); + + return status; +} + +static int __devexit ab8500_codec_driver_remove(struct platform_device *pdev) +{ + dev_info(&pdev->dev, "%s Enter.\n", __func__); + + snd_soc_unregister_codec(&pdev->dev); + + return 0; +} + +static struct platform_driver ab8500_codec_platform_driver = { + .driver = { + .name = "ab8500-codec", + .owner = THIS_MODULE, + }, + .probe = ab8500_codec_driver_probe, + .remove = __devexit_p(ab8500_codec_driver_remove), + .suspend = NULL, + .resume = NULL, +}; +module_platform_driver(ab8500_codec_platform_driver); + +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/ab8500-codec.h b/sound/soc/codecs/ab8500-codec.h new file mode 100644 index 000000000000..114f69a0c629 --- /dev/null +++ b/sound/soc/codecs/ab8500-codec.h @@ -0,0 +1,590 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com>, + * Kristoffer Karlsson <kristoffer.karlsson@stericsson.com>, + * Roger Nilsson <roger.xr.nilsson@stericsson.com>, + * for ST-Ericsson. + * + * Based on the early work done by: + * Mikko J. Lehto <mikko.lehto@symbio.com>, + * Mikko Sarmanne <mikko.sarmanne@symbio.com>, + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#ifndef AB8500_CODEC_REGISTERS_H +#define AB8500_CODEC_REGISTERS_H + +#define AB8500_SUPPORTED_RATE (SNDRV_PCM_RATE_48000) +#define AB8500_SUPPORTED_FMT (SNDRV_PCM_FMTBIT_S16_LE) + +/* AB8500 audio bank (0x0d) register definitions */ + +#define AB8500_POWERUP 0x00 +#define AB8500_AUDSWRESET 0x01 +#define AB8500_ADPATHENA 0x02 +#define AB8500_DAPATHENA 0x03 +#define AB8500_ANACONF1 0x04 +#define AB8500_ANACONF2 0x05 +#define AB8500_DIGMICCONF 0x06 +#define AB8500_ANACONF3 0x07 +#define AB8500_ANACONF4 0x08 +#define AB8500_DAPATHCONF 0x09 +#define AB8500_MUTECONF 0x0A +#define AB8500_SHORTCIRCONF 0x0B +#define AB8500_ANACONF5 0x0C +#define AB8500_ENVCPCONF 0x0D +#define AB8500_SIGENVCONF 0x0E +#define AB8500_PWMGENCONF1 0x0F +#define AB8500_PWMGENCONF2 0x10 +#define AB8500_PWMGENCONF3 0x11 +#define AB8500_PWMGENCONF4 0x12 +#define AB8500_PWMGENCONF5 0x13 +#define AB8500_ANAGAIN1 0x14 +#define AB8500_ANAGAIN2 0x15 +#define AB8500_ANAGAIN3 0x16 +#define AB8500_ANAGAIN4 0x17 +#define AB8500_DIGLINHSLGAIN 0x18 +#define AB8500_DIGLINHSRGAIN 0x19 +#define AB8500_ADFILTCONF 0x1A +#define AB8500_DIGIFCONF1 0x1B +#define AB8500_DIGIFCONF2 0x1C +#define AB8500_DIGIFCONF3 0x1D +#define AB8500_DIGIFCONF4 0x1E +#define AB8500_ADSLOTSEL1 0x1F +#define AB8500_ADSLOTSEL2 0x20 +#define AB8500_ADSLOTSEL3 0x21 +#define AB8500_ADSLOTSEL4 0x22 +#define AB8500_ADSLOTSEL5 0x23 +#define AB8500_ADSLOTSEL6 0x24 +#define AB8500_ADSLOTSEL7 0x25 +#define AB8500_ADSLOTSEL8 0x26 +#define AB8500_ADSLOTSEL9 0x27 +#define AB8500_ADSLOTSEL10 0x28 +#define AB8500_ADSLOTSEL11 0x29 +#define AB8500_ADSLOTSEL12 0x2A +#define AB8500_ADSLOTSEL13 0x2B +#define AB8500_ADSLOTSEL14 0x2C +#define AB8500_ADSLOTSEL15 0x2D +#define AB8500_ADSLOTSEL16 0x2E +#define AB8500_ADSLOTHIZCTRL1 0x2F +#define AB8500_ADSLOTHIZCTRL2 0x30 +#define AB8500_ADSLOTHIZCTRL3 0x31 +#define AB8500_ADSLOTHIZCTRL4 0x32 +#define AB8500_DASLOTCONF1 0x33 +#define AB8500_DASLOTCONF2 0x34 +#define AB8500_DASLOTCONF3 0x35 +#define AB8500_DASLOTCONF4 0x36 +#define AB8500_DASLOTCONF5 0x37 +#define AB8500_DASLOTCONF6 0x38 +#define AB8500_DASLOTCONF7 0x39 +#define AB8500_DASLOTCONF8 0x3A +#define AB8500_CLASSDCONF1 0x3B +#define AB8500_CLASSDCONF2 0x3C +#define AB8500_CLASSDCONF3 0x3D +#define AB8500_DMICFILTCONF 0x3E +#define AB8500_DIGMULTCONF1 0x3F +#define AB8500_DIGMULTCONF2 0x40 +#define AB8500_ADDIGGAIN1 0x41 +#define AB8500_ADDIGGAIN2 0x42 +#define AB8500_ADDIGGAIN3 0x43 +#define AB8500_ADDIGGAIN4 0x44 +#define AB8500_ADDIGGAIN5 0x45 +#define AB8500_ADDIGGAIN6 0x46 +#define AB8500_DADIGGAIN1 0x47 +#define AB8500_DADIGGAIN2 0x48 +#define AB8500_DADIGGAIN3 0x49 +#define AB8500_DADIGGAIN4 0x4A +#define AB8500_DADIGGAIN5 0x4B +#define AB8500_DADIGGAIN6 0x4C +#define AB8500_ADDIGLOOPGAIN1 0x4D +#define AB8500_ADDIGLOOPGAIN2 0x4E +#define AB8500_HSLEARDIGGAIN 0x4F +#define AB8500_HSRDIGGAIN 0x50 +#define AB8500_SIDFIRGAIN1 0x51 +#define AB8500_SIDFIRGAIN2 0x52 +#define AB8500_ANCCONF1 0x53 +#define AB8500_ANCCONF2 0x54 +#define AB8500_ANCCONF3 0x55 +#define AB8500_ANCCONF4 0x56 +#define AB8500_ANCCONF5 0x57 +#define AB8500_ANCCONF6 0x58 +#define AB8500_ANCCONF7 0x59 +#define AB8500_ANCCONF8 0x5A +#define AB8500_ANCCONF9 0x5B +#define AB8500_ANCCONF10 0x5C +#define AB8500_ANCCONF11 0x5D +#define AB8500_ANCCONF12 0x5E +#define AB8500_ANCCONF13 0x5F +#define AB8500_ANCCONF14 0x60 +#define AB8500_SIDFIRADR 0x61 +#define AB8500_SIDFIRCOEF1 0x62 +#define AB8500_SIDFIRCOEF2 0x63 +#define AB8500_SIDFIRCONF 0x64 +#define AB8500_AUDINTMASK1 0x65 +#define AB8500_AUDINTSOURCE1 0x66 +#define AB8500_AUDINTMASK2 0x67 +#define AB8500_AUDINTSOURCE2 0x68 +#define AB8500_FIFOCONF1 0x69 +#define AB8500_FIFOCONF2 0x6A +#define AB8500_FIFOCONF3 0x6B +#define AB8500_FIFOCONF4 0x6C +#define AB8500_FIFOCONF5 0x6D +#define AB8500_FIFOCONF6 0x6E +#define AB8500_AUDREV 0x6F + +#define AB8500_FIRST_REG AB8500_POWERUP +#define AB8500_LAST_REG AB8500_AUDREV +#define AB8500_CACHEREGNUM (AB8500_LAST_REG + 1) + +#define AB8500_MASK_ALL 0xFF +#define AB8500_MASK_NONE 0x00 + +/* AB8500_POWERUP */ +#define AB8500_POWERUP_POWERUP 7 +#define AB8500_POWERUP_ENANA 3 + +/* AB8500_AUDSWRESET */ +#define AB8500_AUDSWRESET_SWRESET 7 + +/* AB8500_ADPATHENA */ +#define AB8500_ADPATHENA_ENAD12 7 +#define AB8500_ADPATHENA_ENAD34 5 +#define AB8500_ADPATHENA_ENAD5768 3 + +/* AB8500_DAPATHENA */ +#define AB8500_DAPATHENA_ENDA1 7 +#define AB8500_DAPATHENA_ENDA2 6 +#define AB8500_DAPATHENA_ENDA3 5 +#define AB8500_DAPATHENA_ENDA4 4 +#define AB8500_DAPATHENA_ENDA5 3 +#define AB8500_DAPATHENA_ENDA6 2 + +/* AB8500_ANACONF1 */ +#define AB8500_ANACONF1_HSLOWPOW 7 +#define AB8500_ANACONF1_DACLOWPOW1 6 +#define AB8500_ANACONF1_DACLOWPOW0 5 +#define AB8500_ANACONF1_EARDACLOWPOW 4 +#define AB8500_ANACONF1_EARSELCM 2 +#define AB8500_ANACONF1_HSHPEN 1 +#define AB8500_ANACONF1_EARDRVLOWPOW 0 + +/* AB8500_ANACONF2 */ +#define AB8500_ANACONF2_ENMIC1 7 +#define AB8500_ANACONF2_ENMIC2 6 +#define AB8500_ANACONF2_ENLINL 5 +#define AB8500_ANACONF2_ENLINR 4 +#define AB8500_ANACONF2_MUTMIC1 3 +#define AB8500_ANACONF2_MUTMIC2 2 +#define AB8500_ANACONF2_MUTLINL 1 +#define AB8500_ANACONF2_MUTLINR 0 + +/* AB8500_DIGMICCONF */ +#define AB8500_DIGMICCONF_ENDMIC1 7 +#define AB8500_DIGMICCONF_ENDMIC2 6 +#define AB8500_DIGMICCONF_ENDMIC3 5 +#define AB8500_DIGMICCONF_ENDMIC4 4 +#define AB8500_DIGMICCONF_ENDMIC5 3 +#define AB8500_DIGMICCONF_ENDMIC6 2 +#define AB8500_DIGMICCONF_HSFADSPEED 0 + +/* AB8500_ANACONF3 */ +#define AB8500_ANACONF3_MIC1SEL 7 +#define AB8500_ANACONF3_LINRSEL 6 +#define AB8500_ANACONF3_ENDRVHSL 5 +#define AB8500_ANACONF3_ENDRVHSR 4 +#define AB8500_ANACONF3_ENADCMIC 2 +#define AB8500_ANACONF3_ENADCLINL 1 +#define AB8500_ANACONF3_ENADCLINR 0 + +/* AB8500_ANACONF4 */ +#define AB8500_ANACONF4_DISPDVSS 7 +#define AB8500_ANACONF4_ENEAR 6 +#define AB8500_ANACONF4_ENHSL 5 +#define AB8500_ANACONF4_ENHSR 4 +#define AB8500_ANACONF4_ENHFL 3 +#define AB8500_ANACONF4_ENHFR 2 +#define AB8500_ANACONF4_ENVIB1 1 +#define AB8500_ANACONF4_ENVIB2 0 + +/* AB8500_DAPATHCONF */ +#define AB8500_DAPATHCONF_ENDACEAR 6 +#define AB8500_DAPATHCONF_ENDACHSL 5 +#define AB8500_DAPATHCONF_ENDACHSR 4 +#define AB8500_DAPATHCONF_ENDACHFL 3 +#define AB8500_DAPATHCONF_ENDACHFR 2 +#define AB8500_DAPATHCONF_ENDACVIB1 1 +#define AB8500_DAPATHCONF_ENDACVIB2 0 + +/* AB8500_MUTECONF */ +#define AB8500_MUTECONF_MUTEAR 6 +#define AB8500_MUTECONF_MUTHSL 5 +#define AB8500_MUTECONF_MUTHSR 4 +#define AB8500_MUTECONF_MUTDACEAR 2 +#define AB8500_MUTECONF_MUTDACHSL 1 +#define AB8500_MUTECONF_MUTDACHSR 0 + +/* AB8500_SHORTCIRCONF */ +#define AB8500_SHORTCIRCONF_ENSHORTPWD 7 +#define AB8500_SHORTCIRCONF_EARSHORTDIS 6 +#define AB8500_SHORTCIRCONF_HSSHORTDIS 5 +#define AB8500_SHORTCIRCONF_HSPULLDEN 4 +#define AB8500_SHORTCIRCONF_HSOSCEN 2 +#define AB8500_SHORTCIRCONF_HSFADDIS 1 +#define AB8500_SHORTCIRCONF_HSZCDDIS 0 +/* Zero cross should be disabled */ + +/* AB8500_ANACONF5 */ +#define AB8500_ANACONF5_ENCPHS 7 +#define AB8500_ANACONF5_HSLDACTOLOL 5 +#define AB8500_ANACONF5_HSRDACTOLOR 4 +#define AB8500_ANACONF5_ENLOL 3 +#define AB8500_ANACONF5_ENLOR 2 +#define AB8500_ANACONF5_HSAUTOEN 0 + +/* AB8500_ENVCPCONF */ +#define AB8500_ENVCPCONF_ENVDETHTHRE 4 +#define AB8500_ENVCPCONF_ENVDETLTHRE 0 +#define AB8500_ENVCPCONF_ENVDETHTHRE_MAX 0x0F +#define AB8500_ENVCPCONF_ENVDETLTHRE_MAX 0x0F + +/* AB8500_SIGENVCONF */ +#define AB8500_SIGENVCONF_CPLVEN 5 +#define AB8500_SIGENVCONF_ENVDETCPEN 4 +#define AB8500_SIGENVCONF_ENVDETTIME 0 +#define AB8500_SIGENVCONF_ENVDETTIME_MAX 0x0F + +/* AB8500_PWMGENCONF1 */ +#define AB8500_PWMGENCONF1_PWMTOVIB1 7 +#define AB8500_PWMGENCONF1_PWMTOVIB2 6 +#define AB8500_PWMGENCONF1_PWM1CTRL 5 +#define AB8500_PWMGENCONF1_PWM2CTRL 4 +#define AB8500_PWMGENCONF1_PWM1NCTRL 3 +#define AB8500_PWMGENCONF1_PWM1PCTRL 2 +#define AB8500_PWMGENCONF1_PWM2NCTRL 1 +#define AB8500_PWMGENCONF1_PWM2PCTRL 0 + +/* AB8500_PWMGENCONF2 */ +/* AB8500_PWMGENCONF3 */ +/* AB8500_PWMGENCONF4 */ +/* AB8500_PWMGENCONF5 */ +#define AB8500_PWMGENCONFX_PWMVIBXPOL 7 +#define AB8500_PWMGENCONFX_PWMVIBXDUTCYC 0 +#define AB8500_PWMGENCONFX_PWMVIBXDUTCYC_MAX 0x64 + +/* AB8500_ANAGAIN1 */ +/* AB8500_ANAGAIN2 */ +#define AB8500_ANAGAINX_ENSEMICX 7 +#define AB8500_ANAGAINX_LOWPOWMICX 6 +#define AB8500_ANAGAINX_MICXGAIN 0 +#define AB8500_ANAGAINX_MICXGAIN_MAX 0x1F + +/* AB8500_ANAGAIN3 */ +#define AB8500_ANAGAIN3_HSLGAIN 4 +#define AB8500_ANAGAIN3_HSRGAIN 0 +#define AB8500_ANAGAIN3_HSXGAIN_MAX 0x0F + +/* AB8500_ANAGAIN4 */ +#define AB8500_ANAGAIN4_LINLGAIN 4 +#define AB8500_ANAGAIN4_LINRGAIN 0 +#define AB8500_ANAGAIN4_LINXGAIN_MAX 0x0F + +/* AB8500_DIGLINHSLGAIN */ +/* AB8500_DIGLINHSRGAIN */ +#define AB8500_DIGLINHSXGAIN_LINTOHSXGAIN 0 +#define AB8500_DIGLINHSXGAIN_LINTOHSXGAIN_MAX 0x13 + +/* AB8500_ADFILTCONF */ +#define AB8500_ADFILTCONF_AD1NH 7 +#define AB8500_ADFILTCONF_AD2NH 6 +#define AB8500_ADFILTCONF_AD3NH 5 +#define AB8500_ADFILTCONF_AD4NH 4 +#define AB8500_ADFILTCONF_AD1VOICE 3 +#define AB8500_ADFILTCONF_AD2VOICE 2 +#define AB8500_ADFILTCONF_AD3VOICE 1 +#define AB8500_ADFILTCONF_AD4VOICE 0 + +/* AB8500_DIGIFCONF1 */ +#define AB8500_DIGIFCONF1_ENMASTGEN 7 +#define AB8500_DIGIFCONF1_IF1BITCLKOS1 6 +#define AB8500_DIGIFCONF1_IF1BITCLKOS0 5 +#define AB8500_DIGIFCONF1_ENFSBITCLK1 4 +#define AB8500_DIGIFCONF1_IF0BITCLKOS1 2 +#define AB8500_DIGIFCONF1_IF0BITCLKOS0 1 +#define AB8500_DIGIFCONF1_ENFSBITCLK0 0 + +/* AB8500_DIGIFCONF2 */ +#define AB8500_DIGIFCONF2_FSYNC0P 6 +#define AB8500_DIGIFCONF2_BITCLK0P 5 +#define AB8500_DIGIFCONF2_IF0DEL 4 +#define AB8500_DIGIFCONF2_IF0FORMAT1 3 +#define AB8500_DIGIFCONF2_IF0FORMAT0 2 +#define AB8500_DIGIFCONF2_IF0WL1 1 +#define AB8500_DIGIFCONF2_IF0WL0 0 + +/* AB8500_DIGIFCONF3 */ +#define AB8500_DIGIFCONF3_IF0DATOIF1AD 7 +#define AB8500_DIGIFCONF3_IF0CLKTOIF1CLK 6 +#define AB8500_DIGIFCONF3_IF1MASTER 5 +#define AB8500_DIGIFCONF3_IF1DATOIF0AD 3 +#define AB8500_DIGIFCONF3_IF1CLKTOIF0CLK 2 +#define AB8500_DIGIFCONF3_IF0MASTER 1 +#define AB8500_DIGIFCONF3_IF0BFIFOEN 0 + +/* AB8500_DIGIFCONF4 */ +#define AB8500_DIGIFCONF4_FSYNC1P 6 +#define AB8500_DIGIFCONF4_BITCLK1P 5 +#define AB8500_DIGIFCONF4_IF1DEL 4 +#define AB8500_DIGIFCONF4_IF1FORMAT1 3 +#define AB8500_DIGIFCONF4_IF1FORMAT0 2 +#define AB8500_DIGIFCONF4_IF1WL1 1 +#define AB8500_DIGIFCONF4_IF1WL0 0 + +/* AB8500_ADSLOTSELX */ +#define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_ODD 0x00 +#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD 0x01 +#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD 0x02 +#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_ODD 0x03 +#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_ODD 0x04 +#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_ODD 0x05 +#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_ODD 0x06 +#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_ODD 0x07 +#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_ODD 0x08 +#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_ODD 0x0F +#define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_EVEN 0x00 +#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_EVEN 0x10 +#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN 0x20 +#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_EVEN 0x30 +#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_EVEN 0x40 +#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_EVEN 0x50 +#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_EVEN 0x60 +#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_EVEN 0x70 +#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_EVEN 0x80 +#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_EVEN 0xF0 +#define AB8500_ADSLOTSELX_EVEN_SHIFT 0 +#define AB8500_ADSLOTSELX_ODD_SHIFT 4 + +/* AB8500_ADSLOTHIZCTRL1 */ +/* AB8500_ADSLOTHIZCTRL2 */ +/* AB8500_ADSLOTHIZCTRL3 */ +/* AB8500_ADSLOTHIZCTRL4 */ +/* AB8500_DASLOTCONF1 */ +#define AB8500_DASLOTCONF1_DA12VOICE 7 +#define AB8500_DASLOTCONF1_SWAPDA12_34 6 +#define AB8500_DASLOTCONF1_DAI7TOADO1 5 + +/* AB8500_DASLOTCONF2 */ +#define AB8500_DASLOTCONF2_DAI8TOADO2 5 + +/* AB8500_DASLOTCONF3 */ +#define AB8500_DASLOTCONF3_DA34VOICE 7 +#define AB8500_DASLOTCONF3_DAI7TOADO3 5 + +/* AB8500_DASLOTCONF4 */ +#define AB8500_DASLOTCONF4_DAI8TOADO4 5 + +/* AB8500_DASLOTCONF5 */ +#define AB8500_DASLOTCONF5_DA56VOICE 7 +#define AB8500_DASLOTCONF5_DAI7TOADO5 5 + +/* AB8500_DASLOTCONF6 */ +#define AB8500_DASLOTCONF6_DAI8TOADO6 5 + +/* AB8500_DASLOTCONF7 */ +#define AB8500_DASLOTCONF7_DAI8TOADO7 5 + +/* AB8500_DASLOTCONF8 */ +#define AB8500_DASLOTCONF8_DAI7TOADO8 5 + +#define AB8500_DASLOTCONFX_SLTODAX_SHIFT 0 +#define AB8500_DASLOTCONFX_SLTODAX_MASK 0x1F + +/* AB8500_CLASSDCONF1 */ +#define AB8500_CLASSDCONF1_PARLHF 7 +#define AB8500_CLASSDCONF1_PARLVIB 6 +#define AB8500_CLASSDCONF1_VIB1SWAPEN 3 +#define AB8500_CLASSDCONF1_VIB2SWAPEN 2 +#define AB8500_CLASSDCONF1_HFLSWAPEN 1 +#define AB8500_CLASSDCONF1_HFRSWAPEN 0 + +/* AB8500_CLASSDCONF2 */ +#define AB8500_CLASSDCONF2_FIRBYP3 7 +#define AB8500_CLASSDCONF2_FIRBYP2 6 +#define AB8500_CLASSDCONF2_FIRBYP1 5 +#define AB8500_CLASSDCONF2_FIRBYP0 4 +#define AB8500_CLASSDCONF2_HIGHVOLEN3 3 +#define AB8500_CLASSDCONF2_HIGHVOLEN2 2 +#define AB8500_CLASSDCONF2_HIGHVOLEN1 1 +#define AB8500_CLASSDCONF2_HIGHVOLEN0 0 + +/* AB8500_CLASSDCONF3 */ +#define AB8500_CLASSDCONF3_DITHHPGAIN 4 +#define AB8500_CLASSDCONF3_DITHHPGAIN_MAX 0x0A +#define AB8500_CLASSDCONF3_DITHWGAIN 0 +#define AB8500_CLASSDCONF3_DITHWGAIN_MAX 0x0A + +/* AB8500_DMICFILTCONF */ +#define AB8500_DMICFILTCONF_ANCINSEL 7 +#define AB8500_DMICFILTCONF_DA3TOEAR 6 +#define AB8500_DMICFILTCONF_DMIC1SINC3 5 +#define AB8500_DMICFILTCONF_DMIC2SINC3 4 +#define AB8500_DMICFILTCONF_DMIC3SINC3 3 +#define AB8500_DMICFILTCONF_DMIC4SINC3 2 +#define AB8500_DMICFILTCONF_DMIC5SINC3 1 +#define AB8500_DMICFILTCONF_DMIC6SINC3 0 + +/* AB8500_DIGMULTCONF1 */ +#define AB8500_DIGMULTCONF1_DATOHSLEN 7 +#define AB8500_DIGMULTCONF1_DATOHSREN 6 +#define AB8500_DIGMULTCONF1_AD1SEL 5 +#define AB8500_DIGMULTCONF1_AD2SEL 4 +#define AB8500_DIGMULTCONF1_AD3SEL 3 +#define AB8500_DIGMULTCONF1_AD5SEL 2 +#define AB8500_DIGMULTCONF1_AD6SEL 1 +#define AB8500_DIGMULTCONF1_ANCSEL 0 + +/* AB8500_DIGMULTCONF2 */ +#define AB8500_DIGMULTCONF2_DATOHFREN 7 +#define AB8500_DIGMULTCONF2_DATOHFLEN 6 +#define AB8500_DIGMULTCONF2_HFRSEL 5 +#define AB8500_DIGMULTCONF2_HFLSEL 4 +#define AB8500_DIGMULTCONF2_FIRSID1SEL 2 +#define AB8500_DIGMULTCONF2_FIRSID2SEL 0 + +/* AB8500_ADDIGGAIN1 */ +/* AB8500_ADDIGGAIN2 */ +/* AB8500_ADDIGGAIN3 */ +/* AB8500_ADDIGGAIN4 */ +/* AB8500_ADDIGGAIN5 */ +/* AB8500_ADDIGGAIN6 */ +#define AB8500_ADDIGGAINX_FADEDISADX 6 +#define AB8500_ADDIGGAINX_ADXGAIN_MAX 0x3F + +/* AB8500_DADIGGAIN1 */ +/* AB8500_DADIGGAIN2 */ +/* AB8500_DADIGGAIN3 */ +/* AB8500_DADIGGAIN4 */ +/* AB8500_DADIGGAIN5 */ +/* AB8500_DADIGGAIN6 */ +#define AB8500_DADIGGAINX_FADEDISDAX 6 +#define AB8500_DADIGGAINX_DAXGAIN_MAX 0x3F + +/* AB8500_ADDIGLOOPGAIN1 */ +/* AB8500_ADDIGLOOPGAIN2 */ +#define AB8500_ADDIGLOOPGAINX_FADEDISADXL 6 +#define AB8500_ADDIGLOOPGAINX_ADXLBGAIN_MAX 0x3F + +/* AB8500_HSLEARDIGGAIN */ +#define AB8500_HSLEARDIGGAIN_HSSINC1 7 +#define AB8500_HSLEARDIGGAIN_FADEDISHSL 4 +#define AB8500_HSLEARDIGGAIN_HSLDGAIN_MAX 0x09 + +/* AB8500_HSRDIGGAIN */ +#define AB8500_HSRDIGGAIN_FADESPEED 6 +#define AB8500_HSRDIGGAIN_FADEDISHSR 4 +#define AB8500_HSRDIGGAIN_HSRDGAIN_MAX 0x09 + +/* AB8500_SIDFIRGAIN1 */ +/* AB8500_SIDFIRGAIN2 */ +#define AB8500_SIDFIRGAINX_FIRSIDXGAIN_MAX 0x1F + +/* AB8500_ANCCONF1 */ +#define AB8500_ANCCONF1_ANCIIRUPDATE 3 +#define AB8500_ANCCONF1_ENANC 2 +#define AB8500_ANCCONF1_ANCIIRINIT 1 +#define AB8500_ANCCONF1_ANCFIRUPDATE 0 + +/* AB8500_ANCCONF2 */ +#define AB8500_ANCCONF2_SHIFT 5 +#define AB8500_ANCCONF2_MIN -0x10 +#define AB8500_ANCCONF2_MAX 0xF + +/* AB8500_ANCCONF3 */ +#define AB8500_ANCCONF3_SHIFT 5 +#define AB8500_ANCCONF3_MIN -0x10 +#define AB8500_ANCCONF3_MAX 0xF + +/* AB8500_ANCCONF4 */ +#define AB8500_ANCCONF4_SHIFT 5 +#define AB8500_ANCCONF4_MIN -0x10 +#define AB8500_ANCCONF4_MAX 0xF + +/* AB8500_ANC_FIR_COEFFS */ +#define AB8500_ANC_FIR_COEFF_MIN -0x8000 +#define AB8500_ANC_FIR_COEFF_MAX 0x7FFF +#define AB8500_ANC_FIR_COEFFS 15 + +/* AB8500_ANC_IIR_COEFFS */ +#define AB8500_ANC_IIR_COEFF_MIN -0x800000 +#define AB8500_ANC_IIR_COEFF_MAX 0x7FFFFF +#define AB8500_ANC_IIR_COEFFS 24 +/* AB8500_ANC_WARP_DELAY */ +#define AB8500_ANC_WARP_DELAY_SHIFT 16 +#define AB8500_ANC_WARP_DELAY_MIN 0x0000 +#define AB8500_ANC_WARP_DELAY_MAX 0xFFFF + +/* AB8500_ANCCONF11 */ +/* AB8500_ANCCONF12 */ +/* AB8500_ANCCONF13 */ +/* AB8500_ANCCONF14 */ + +/* AB8500_SIDFIRADR */ +#define AB8500_SIDFIRADR_FIRSIDSET 7 +#define AB8500_SIDFIRADR_ADDRESS_SHIFT 0 +#define AB8500_SIDFIRADR_ADDRESS_MAX 0x7F + +/* AB8500_SIDFIRCOEF1 */ +/* AB8500_SIDFIRCOEF2 */ +#define AB8500_SID_FIR_COEFF_MIN 0 +#define AB8500_SID_FIR_COEFF_MAX 0xFFFF +#define AB8500_SID_FIR_COEFFS 128 + +/* AB8500_SIDFIRCONF */ +#define AB8500_SIDFIRCONF_ENFIRSIDS 2 +#define AB8500_SIDFIRCONF_FIRSIDSTOIF1 1 +#define AB8500_SIDFIRCONF_FIRSIDBUSY 0 + +/* AB8500_AUDINTMASK1 */ +/* AB8500_AUDINTSOURCE1 */ +/* AB8500_AUDINTMASK2 */ +/* AB8500_AUDINTSOURCE2 */ + +/* AB8500_FIFOCONF1 */ +#define AB8500_FIFOCONF1_BFIFOMASK 0x80 +#define AB8500_FIFOCONF1_BFIFO19M2 0x40 +#define AB8500_FIFOCONF1_BFIFOINT_SHIFT 0 +#define AB8500_FIFOCONF1_BFIFOINT_MAX 0x3F + +/* AB8500_FIFOCONF2 */ +#define AB8500_FIFOCONF2_BFIFOTX_SHIFT 0 +#define AB8500_FIFOCONF2_BFIFOTX_MAX 0xFF + +/* AB8500_FIFOCONF3 */ +#define AB8500_FIFOCONF3_BFIFOEXSL_SHIFT 5 +#define AB8500_FIFOCONF3_BFIFOEXSL_MAX 0x5 +#define AB8500_FIFOCONF3_PREBITCLK0_SHIFT 2 +#define AB8500_FIFOCONF3_PREBITCLK0_MAX 0x7 +#define AB8500_FIFOCONF3_BFIFOMAST_SHIFT 1 +#define AB8500_FIFOCONF3_BFIFORUN_SHIFT 0 + +/* AB8500_FIFOCONF4 */ +#define AB8500_FIFOCONF4_BFIFOFRAMSW_SHIFT 0 +#define AB8500_FIFOCONF4_BFIFOFRAMSW_MAX 0xFF + +/* AB8500_FIFOCONF5 */ +#define AB8500_FIFOCONF5_BFIFOWAKEUP_SHIFT 0 +#define AB8500_FIFOCONF5_BFIFOWAKEUP_MAX 0xFF + +/* AB8500_FIFOCONF6 */ +#define AB8500_FIFOCONF6_BFIFOSAMPLE_SHIFT 0 +#define AB8500_FIFOCONF6_BFIFOSAMPLE_MAX 0xFF + +/* AB8500_AUDREV */ + +#endif diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 1bbad4c16d28..ea06b834a7de 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -26,13 +26,11 @@ static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; - return snd_ac97_set_rate(codec->ac97, reg, runtime->rate); + return snd_ac97_set_rate(codec->ac97, reg, substream->runtime->rate); } #define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ @@ -93,11 +91,6 @@ static int ac97_soc_probe(struct snd_soc_codec *codec) return 0; } -static int ac97_soc_remove(struct snd_soc_codec *codec) -{ - return 0; -} - #ifdef CONFIG_PM static int ac97_soc_suspend(struct snd_soc_codec *codec) { @@ -121,7 +114,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ac97 = { .write = ac97_write, .read = ac97_read, .probe = ac97_soc_probe, - .remove = ac97_soc_remove, .suspend = ac97_soc_suspend, .resume = ac97_soc_resume, }; diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 12e3b4118557..c67b50d8b317 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -162,9 +162,7 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { int word_len = 0; - - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; /* bit size */ switch (params_format(params)) { diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index a4a6bef2c0bb..13e62be4f990 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -245,9 +245,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { int word_len = 0, master_rate = 0; - - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); /* bit size */ diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 78e9ce48bb99..3d50fc8646b6 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -258,8 +258,7 @@ static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec, static int adau1701_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; snd_pcm_format_t format; unsigned int val; diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index ceb96ecf5588..31d4483245d0 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -88,8 +88,7 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int val = 0; /* set the IEC958 bits: consumer mode, no copyright bit */ diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 838ae8b22b50..618fdc30f73e 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -262,8 +262,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); u8 mode2 = snd_soc_read(codec, AK4535_MODE2) & ~(0x3 << 5); int rate = params_rate(params), fs = 256; diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index c4d165a4bddf..543a12f471be 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -296,8 +296,7 @@ static int ak4641_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); int rate = params_rate(params), fs = 256; u8 mode2; @@ -517,67 +516,24 @@ static int ak4641_resume(struct snd_soc_codec *codec) static int ak4641_probe(struct snd_soc_codec *codec) { - struct ak4641_platform_data *pdata = codec->dev->platform_data; int ret; - - if (pdata) { - if (gpio_is_valid(pdata->gpio_power)) { - ret = gpio_request_one(pdata->gpio_power, - GPIOF_OUT_INIT_LOW, "ak4641 power"); - if (ret) - goto err_out; - } - if (gpio_is_valid(pdata->gpio_npdn)) { - ret = gpio_request_one(pdata->gpio_npdn, - GPIOF_OUT_INIT_LOW, "ak4641 npdn"); - if (ret) - goto err_gpio; - - udelay(1); /* > 150 ns */ - gpio_set_value(pdata->gpio_npdn, 1); - } - } - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - goto err_register; + return ret; } /* power on device */ ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; - -err_register: - if (pdata) { - if (gpio_is_valid(pdata->gpio_power)) - gpio_set_value(pdata->gpio_power, 0); - if (gpio_is_valid(pdata->gpio_npdn)) - gpio_free(pdata->gpio_npdn); - } -err_gpio: - if (pdata && gpio_is_valid(pdata->gpio_power)) - gpio_free(pdata->gpio_power); -err_out: - return ret; } static int ak4641_remove(struct snd_soc_codec *codec) { - struct ak4641_platform_data *pdata = codec->dev->platform_data; - ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (pdata) { - if (gpio_is_valid(pdata->gpio_power)) { - gpio_set_value(pdata->gpio_power, 0); - gpio_free(pdata->gpio_power); - } - if (gpio_is_valid(pdata->gpio_npdn)) - gpio_free(pdata->gpio_npdn); - } return 0; } @@ -604,6 +560,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4641 = { static int __devinit ak4641_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct ak4641_platform_data *pdata = i2c->dev.platform_data; struct ak4641_priv *ak4641; int ret; @@ -612,16 +569,62 @@ static int __devinit ak4641_i2c_probe(struct i2c_client *i2c, if (!ak4641) return -ENOMEM; + if (pdata) { + if (gpio_is_valid(pdata->gpio_power)) { + ret = gpio_request_one(pdata->gpio_power, + GPIOF_OUT_INIT_LOW, "ak4641 power"); + if (ret) + goto err_out; + } + if (gpio_is_valid(pdata->gpio_npdn)) { + ret = gpio_request_one(pdata->gpio_npdn, + GPIOF_OUT_INIT_LOW, "ak4641 npdn"); + if (ret) + goto err_gpio; + + udelay(1); /* > 150 ns */ + gpio_set_value(pdata->gpio_npdn, 1); + } + } + i2c_set_clientdata(i2c, ak4641); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ak4641, ak4641_dai, ARRAY_SIZE(ak4641_dai)); + if (ret != 0) + goto err_gpio2; + + return 0; + +err_gpio2: + if (pdata) { + if (gpio_is_valid(pdata->gpio_power)) + gpio_set_value(pdata->gpio_power, 0); + if (gpio_is_valid(pdata->gpio_npdn)) + gpio_free(pdata->gpio_npdn); + } +err_gpio: + if (pdata && gpio_is_valid(pdata->gpio_power)) + gpio_free(pdata->gpio_power); +err_out: return ret; } static int __devexit ak4641_i2c_remove(struct i2c_client *i2c) { + struct ak4641_platform_data *pdata = i2c->dev.platform_data; + snd_soc_unregister_codec(&i2c->dev); + + if (pdata) { + if (gpio_is_valid(pdata->gpio_power)) { + gpio_set_value(pdata->gpio_power, 0); + gpio_free(pdata->gpio_power); + } + if (gpio_is_valid(pdata->gpio_npdn)) + gpio_free(pdata->gpio_npdn); + } + return 0; } @@ -641,23 +644,7 @@ static struct i2c_driver ak4641_i2c_driver = { .id_table = ak4641_i2c_id, }; -static int __init ak4641_modinit(void) -{ - int ret; - - ret = i2c_add_driver(&ak4641_i2c_driver); - if (ret != 0) - pr_err("Failed to register AK4641 I2C driver: %d\n", ret); - - return ret; -} -module_init(ak4641_modinit); - -static void __exit ak4641_exit(void) -{ - i2c_del_driver(&ak4641_i2c_driver); -} -module_exit(ak4641_exit); +module_i2c_driver(ak4641_i2c_driver); MODULE_DESCRIPTION("SoC AK4641 driver"); MODULE_AUTHOR("Harald Welte <laforge@gnufiish.org>"); diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index d47b62ddb210..1960478ce6bb 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -705,8 +705,7 @@ static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai, static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); int coeff, rate; u16 iface; @@ -1084,25 +1083,7 @@ static struct i2c_driver alc5623_i2c_driver = { .id_table = alc5623_i2c_table, }; -static int __init alc5623_modinit(void) -{ - int ret; - - ret = i2c_add_driver(&alc5623_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "%s: can't add i2c driver", __func__); - return ret; - } - - return ret; -} -module_init(alc5623_modinit); - -static void __exit alc5623_modexit(void) -{ - i2c_del_driver(&alc5623_i2c_driver); -} -module_exit(alc5623_modexit); +module_i2c_driver(alc5623_i2c_driver); MODULE_DESCRIPTION("ASoC alc5621/2/3 driver"); MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>"); diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index e2111e0ccad7..7dd02420b36d 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -861,8 +861,7 @@ static int alc5632_set_dai_fmt(struct snd_soc_dai *codec_dai, static int alc5632_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int coeff, rate; u16 iface; @@ -1131,7 +1130,7 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, alc5632); - alc5632->regmap = regmap_init_i2c(client, &alc5632_regmap); + alc5632->regmap = devm_regmap_init_i2c(client, &alc5632_regmap); if (IS_ERR(alc5632->regmap)) { ret = PTR_ERR(alc5632->regmap); dev_err(&client->dev, "regmap_init() failed: %d\n", ret); @@ -1143,7 +1142,6 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, if (ret1 != 0 || ret2 != 0) { dev_err(&client->dev, "Failed to read chip ID: ret1=%d, ret2=%d\n", ret1, ret2); - regmap_exit(alc5632->regmap); return -EIO; } @@ -1152,14 +1150,12 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, if ((vid1 != 0x10EC) || (vid2 != id->driver_data)) { dev_err(&client->dev, "Device is not a ALC5632: VID1=0x%x, VID2=0x%x\n", vid1, vid2); - regmap_exit(alc5632->regmap); return -EINVAL; } ret = alc5632_reset(alc5632->regmap); if (ret < 0) { dev_err(&client->dev, "Failed to issue reset\n"); - regmap_exit(alc5632->regmap); return ret; } @@ -1177,7 +1173,6 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, if (ret < 0) { dev_err(&client->dev, "Failed to register codec: %d\n", ret); - regmap_exit(alc5632->regmap); return ret; } @@ -1186,9 +1181,7 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, static __devexit int alc5632_i2c_remove(struct i2c_client *client) { - struct alc5632_priv *alc5632 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); - regmap_exit(alc5632->regmap); return 0; } @@ -1209,25 +1202,7 @@ static struct i2c_driver alc5632_i2c_driver = { .id_table = alc5632_i2c_table, }; -static int __init alc5632_modinit(void) -{ - int ret; - - ret = i2c_add_driver(&alc5632_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "%s: can't add i2c driver", __func__); - return ret; - } - - return ret; -} -module_init(alc5632_modinit); - -static void __exit alc5632_modexit(void) -{ - i2c_del_driver(&alc5632_i2c_driver); -} -module_exit(alc5632_modexit); +module_i2c_driver(alc5632_i2c_driver); MODULE_DESCRIPTION("ASoC ALC5632 driver"); MODULE_AUTHOR("Leon Romanovsky <leon@leon.nu>"); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c new file mode 100644 index 000000000000..5c9cacaf2d52 --- /dev/null +++ b/sound/soc/codecs/arizona.c @@ -0,0 +1,937 @@ +/* + * arizona.c - Wolfson Arizona class device shared support + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/gcd.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> + +#include <linux/mfd/arizona/core.h> +#include <linux/mfd/arizona/registers.h> + +#include "arizona.h" + +#define ARIZONA_AIF_BCLK_CTRL 0x00 +#define ARIZONA_AIF_TX_PIN_CTRL 0x01 +#define ARIZONA_AIF_RX_PIN_CTRL 0x02 +#define ARIZONA_AIF_RATE_CTRL 0x03 +#define ARIZONA_AIF_FORMAT 0x04 +#define ARIZONA_AIF_TX_BCLK_RATE 0x05 +#define ARIZONA_AIF_RX_BCLK_RATE 0x06 +#define ARIZONA_AIF_FRAME_CTRL_1 0x07 +#define ARIZONA_AIF_FRAME_CTRL_2 0x08 +#define ARIZONA_AIF_FRAME_CTRL_3 0x09 +#define ARIZONA_AIF_FRAME_CTRL_4 0x0A +#define ARIZONA_AIF_FRAME_CTRL_5 0x0B +#define ARIZONA_AIF_FRAME_CTRL_6 0x0C +#define ARIZONA_AIF_FRAME_CTRL_7 0x0D +#define ARIZONA_AIF_FRAME_CTRL_8 0x0E +#define ARIZONA_AIF_FRAME_CTRL_9 0x0F +#define ARIZONA_AIF_FRAME_CTRL_10 0x10 +#define ARIZONA_AIF_FRAME_CTRL_11 0x11 +#define ARIZONA_AIF_FRAME_CTRL_12 0x12 +#define ARIZONA_AIF_FRAME_CTRL_13 0x13 +#define ARIZONA_AIF_FRAME_CTRL_14 0x14 +#define ARIZONA_AIF_FRAME_CTRL_15 0x15 +#define ARIZONA_AIF_FRAME_CTRL_16 0x16 +#define ARIZONA_AIF_FRAME_CTRL_17 0x17 +#define ARIZONA_AIF_FRAME_CTRL_18 0x18 +#define ARIZONA_AIF_TX_ENABLES 0x19 +#define ARIZONA_AIF_RX_ENABLES 0x1A +#define ARIZONA_AIF_FORCE_WRITE 0x1B + +#define arizona_fll_err(_fll, fmt, ...) \ + dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) +#define arizona_fll_warn(_fll, fmt, ...) \ + dev_warn(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) +#define arizona_fll_dbg(_fll, fmt, ...) \ + dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) + +#define arizona_aif_err(_dai, fmt, ...) \ + dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) +#define arizona_aif_warn(_dai, fmt, ...) \ + dev_warn(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) +#define arizona_aif_dbg(_dai, fmt, ...) \ + dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) + +const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { + "None", + "Tone Generator 1", + "Tone Generator 2", + "Haptics", + "AEC", + "Mic Mute Mixer", + "Noise Generator", + "IN1L", + "IN1R", + "IN2L", + "IN2R", + "IN3L", + "IN3R", + "IN4L", + "IN4R", + "AIF1RX1", + "AIF1RX2", + "AIF1RX3", + "AIF1RX4", + "AIF1RX5", + "AIF1RX6", + "AIF1RX7", + "AIF1RX8", + "AIF2RX1", + "AIF2RX2", + "AIF3RX1", + "AIF3RX2", + "SLIMRX1", + "SLIMRX2", + "SLIMRX3", + "SLIMRX4", + "SLIMRX5", + "SLIMRX6", + "SLIMRX7", + "SLIMRX8", + "EQ1", + "EQ2", + "EQ3", + "EQ4", + "DRC1L", + "DRC1R", + "DRC2L", + "DRC2R", + "LHPF1", + "LHPF2", + "LHPF3", + "LHPF4", + "DSP1.1", + "DSP1.2", + "DSP1.3", + "DSP1.4", + "DSP1.5", + "DSP1.6", + "ASRC1L", + "ASRC1R", + "ASRC2L", + "ASRC2R", +}; +EXPORT_SYMBOL_GPL(arizona_mixer_texts); + +int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = { + 0x00, /* None */ + 0x04, /* Tone */ + 0x05, + 0x06, /* Haptics */ + 0x08, /* AEC */ + 0x0c, /* Noise mixer */ + 0x0d, /* Comfort noise */ + 0x10, /* IN1L */ + 0x11, + 0x12, + 0x13, + 0x14, + 0x15, + 0x16, + 0x17, + 0x20, /* AIF1RX1 */ + 0x21, + 0x22, + 0x23, + 0x24, + 0x25, + 0x26, + 0x27, + 0x28, /* AIF2RX1 */ + 0x29, + 0x30, /* AIF3RX1 */ + 0x31, + 0x38, /* SLIMRX1 */ + 0x39, + 0x3a, + 0x3b, + 0x3c, + 0x3d, + 0x3e, + 0x3f, + 0x50, /* EQ1 */ + 0x51, + 0x52, + 0x53, + 0x58, /* DRC1L */ + 0x59, + 0x5a, + 0x5b, + 0x60, /* LHPF1 */ + 0x61, + 0x62, + 0x63, + 0x68, /* DSP1.1 */ + 0x69, + 0x6a, + 0x6b, + 0x6c, + 0x6d, + 0x90, /* ASRC1L */ + 0x91, + 0x92, + 0x93, +}; +EXPORT_SYMBOL_GPL(arizona_mixer_values); + +const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0); +EXPORT_SYMBOL_GPL(arizona_mixer_tlv); + +static const char *arizona_lhpf_mode_text[] = { + "Low-pass", "High-pass" +}; + +const struct soc_enum arizona_lhpf1_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF1_1, ARIZONA_LHPF1_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf1_mode); + +const struct soc_enum arizona_lhpf2_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF2_1, ARIZONA_LHPF2_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf2_mode); + +const struct soc_enum arizona_lhpf3_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF3_1, ARIZONA_LHPF3_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf3_mode); + +const struct soc_enum arizona_lhpf4_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF4_1, ARIZONA_LHPF4_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf4_mode); + +int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, + int event) +{ + return 0; +} +EXPORT_SYMBOL_GPL(arizona_in_ev); + +int arizona_out_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + return 0; +} +EXPORT_SYMBOL_GPL(arizona_out_ev); + +int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + char *name; + unsigned int reg; + unsigned int mask = ARIZONA_SYSCLK_FREQ_MASK | ARIZONA_SYSCLK_SRC_MASK; + unsigned int val = source << ARIZONA_SYSCLK_SRC_SHIFT; + unsigned int *clk; + + switch (clk_id) { + case ARIZONA_CLK_SYSCLK: + name = "SYSCLK"; + reg = ARIZONA_SYSTEM_CLOCK_1; + clk = &priv->sysclk; + mask |= ARIZONA_SYSCLK_FRAC; + break; + case ARIZONA_CLK_ASYNCCLK: + name = "ASYNCCLK"; + reg = ARIZONA_ASYNC_CLOCK_1; + clk = &priv->asyncclk; + break; + default: + return -EINVAL; + } + + switch (freq) { + case 5644800: + case 6144000: + break; + case 11289600: + case 12288000: + val |= 1 << ARIZONA_SYSCLK_FREQ_SHIFT; + break; + case 22579200: + case 24576000: + val |= 2 << ARIZONA_SYSCLK_FREQ_SHIFT; + break; + case 45158400: + case 49152000: + val |= 3 << ARIZONA_SYSCLK_FREQ_SHIFT; + break; + default: + return -EINVAL; + } + + *clk = freq; + + if (freq % 6144000) + val |= ARIZONA_SYSCLK_FRAC; + + dev_dbg(arizona->dev, "%s set to %uHz", name, freq); + + return regmap_update_bits(arizona->regmap, reg, mask, val); +} +EXPORT_SYMBOL_GPL(arizona_set_sysclk); + +static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + int lrclk, bclk, mode, base; + + base = dai->driver->base; + + lrclk = 0; + bclk = 0; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + mode = 0; + break; + case SND_SOC_DAIFMT_DSP_B: + mode = 1; + break; + case SND_SOC_DAIFMT_I2S: + mode = 2; + break; + case SND_SOC_DAIFMT_LEFT_J: + mode = 3; + break; + default: + arizona_aif_err(dai, "Unsupported DAI format %d\n", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBS_CFM: + lrclk |= ARIZONA_AIF1TX_LRCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + bclk |= ARIZONA_AIF1_BCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFM: + bclk |= ARIZONA_AIF1_BCLK_MSTR; + lrclk |= ARIZONA_AIF1TX_LRCLK_MSTR; + break; + default: + arizona_aif_err(dai, "Unsupported master mode %d\n", + fmt & SND_SOC_DAIFMT_MASTER_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + bclk |= ARIZONA_AIF1_BCLK_INV; + lrclk |= ARIZONA_AIF1TX_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + bclk |= ARIZONA_AIF1_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + lrclk |= ARIZONA_AIF1TX_LRCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_INV | ARIZONA_AIF1_BCLK_MSTR, + bclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_PIN_CTRL, + ARIZONA_AIF1TX_LRCLK_INV | + ARIZONA_AIF1TX_LRCLK_MSTR, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_PIN_CTRL, + ARIZONA_AIF1RX_LRCLK_INV | + ARIZONA_AIF1RX_LRCLK_MSTR, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_FORMAT, + ARIZONA_AIF1_FMT_MASK, mode); + + return 0; +} + +static const int arizona_48k_bclk_rates[] = { + -1, + 48000, + 64000, + 96000, + 128000, + 192000, + 256000, + 384000, + 512000, + 768000, + 1024000, + 1536000, + 2048000, + 3072000, + 4096000, + 6144000, + 8192000, + 12288000, + 24576000, +}; + +static const unsigned int arizona_48k_rates[] = { + 12000, + 24000, + 48000, + 96000, + 192000, + 384000, + 768000, + 4000, + 8000, + 16000, + 32000, + 64000, + 128000, + 256000, + 512000, +}; + +static const struct snd_pcm_hw_constraint_list arizona_48k_constraint = { + .count = ARRAY_SIZE(arizona_48k_rates), + .list = arizona_48k_rates, +}; + +static const int arizona_44k1_bclk_rates[] = { + -1, + 44100, + 58800, + 88200, + 117600, + 177640, + 235200, + 352800, + 470400, + 705600, + 940800, + 1411200, + 1881600, + 2882400, + 3763200, + 5644800, + 7526400, + 11289600, + 22579200, +}; + +static const unsigned int arizona_44k1_rates[] = { + 11025, + 22050, + 44100, + 88200, + 176400, + 352800, + 705600, +}; + +static const struct snd_pcm_hw_constraint_list arizona_44k1_constraint = { + .count = ARRAY_SIZE(arizona_44k1_rates), + .list = arizona_44k1_rates, +}; + +static int arizona_sr_vals[] = { + 0, + 12000, + 24000, + 48000, + 96000, + 192000, + 384000, + 768000, + 0, + 11025, + 22050, + 44100, + 88200, + 176400, + 352800, + 705600, + 4000, + 8000, + 16000, + 32000, + 64000, + 128000, + 256000, + 512000, +}; + +static int arizona_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; + const struct snd_pcm_hw_constraint_list *constraint; + unsigned int base_rate; + + switch (dai_priv->clk) { + case ARIZONA_CLK_SYSCLK: + base_rate = priv->sysclk; + break; + case ARIZONA_CLK_ASYNCCLK: + base_rate = priv->asyncclk; + break; + default: + return 0; + } + + if (base_rate % 8000) + constraint = &arizona_44k1_constraint; + else + constraint = &arizona_48k_constraint; + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + constraint); +} + +static int arizona_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; + int base = dai->driver->base; + const int *rates; + int i; + int bclk, lrclk, wl, frame, sr_val; + + if (params_rate(params) % 8000) + rates = &arizona_44k1_bclk_rates[0]; + else + rates = &arizona_48k_bclk_rates[0]; + + for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) { + if (rates[i] >= snd_soc_params_to_bclk(params) && + rates[i] % params_rate(params) == 0) { + bclk = i; + break; + } + } + if (i == ARRAY_SIZE(arizona_44k1_bclk_rates)) { + arizona_aif_err(dai, "Unsupported sample rate %dHz\n", + params_rate(params)); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(arizona_sr_vals); i++) + if (arizona_sr_vals[i] == params_rate(params)) + break; + if (i == ARRAY_SIZE(arizona_sr_vals)) { + arizona_aif_err(dai, "Unsupported sample rate %dHz\n", + params_rate(params)); + return -EINVAL; + } + sr_val = i; + + lrclk = snd_soc_params_to_bclk(params) / params_rate(params); + + arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n", + rates[bclk], rates[bclk] / lrclk); + + wl = snd_pcm_format_width(params_format(params)); + frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; + + /* + * We will need to be more flexible than this in future, + * currently we use a single sample rate for SYSCLK. + */ + switch (dai_priv->clk) { + case ARIZONA_CLK_SYSCLK: + snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, + ARIZONA_SAMPLE_RATE_1_MASK, sr_val); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, 0); + break; + case ARIZONA_CLK_ASYNCCLK: + snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1, + ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, 8); + break; + default: + arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk); + return -EINVAL; + } + + snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_BCLK_RATE, + ARIZONA_AIF1TX_BCPF_MASK, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_BCLK_RATE, + ARIZONA_AIF1RX_BCPF_MASK, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_1, + ARIZONA_AIF1TX_WL_MASK | + ARIZONA_AIF1TX_SLOT_LEN_MASK, frame); + snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_2, + ARIZONA_AIF1RX_WL_MASK | + ARIZONA_AIF1RX_SLOT_LEN_MASK, frame); + + return 0; +} + +static const char *arizona_dai_clk_str(int clk_id) +{ + switch (clk_id) { + case ARIZONA_CLK_SYSCLK: + return "SYSCLK"; + case ARIZONA_CLK_ASYNCCLK: + return "ASYNCCLK"; + default: + return "Unknown clock"; + } +} + +static int arizona_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; + struct snd_soc_dapm_route routes[2]; + + switch (clk_id) { + case ARIZONA_CLK_SYSCLK: + case ARIZONA_CLK_ASYNCCLK: + break; + default: + return -EINVAL; + } + + if (clk_id == dai_priv->clk) + return 0; + + if (dai->active) { + dev_err(codec->dev, "Can't change clock on active DAI %d\n", + dai->id); + return -EBUSY; + } + + memset(&routes, 0, sizeof(routes)); + routes[0].sink = dai->driver->capture.stream_name; + routes[1].sink = dai->driver->playback.stream_name; + + routes[0].source = arizona_dai_clk_str(dai_priv->clk); + routes[1].source = arizona_dai_clk_str(dai_priv->clk); + snd_soc_dapm_del_routes(&codec->dapm, routes, ARRAY_SIZE(routes)); + + routes[0].source = arizona_dai_clk_str(clk_id); + routes[1].source = arizona_dai_clk_str(clk_id); + snd_soc_dapm_add_routes(&codec->dapm, routes, ARRAY_SIZE(routes)); + + return snd_soc_dapm_sync(&codec->dapm); +} + +const struct snd_soc_dai_ops arizona_dai_ops = { + .startup = arizona_startup, + .set_fmt = arizona_set_fmt, + .hw_params = arizona_hw_params, + .set_sysclk = arizona_dai_set_sysclk, +}; +EXPORT_SYMBOL_GPL(arizona_dai_ops); + +int arizona_init_dai(struct arizona_priv *priv, int id) +{ + struct arizona_dai_priv *dai_priv = &priv->dai[id]; + + dai_priv->clk = ARIZONA_CLK_SYSCLK; + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_dai); + +static irqreturn_t arizona_fll_lock(int irq, void *data) +{ + struct arizona_fll *fll = data; + + arizona_fll_dbg(fll, "Locked\n"); + + complete(&fll->lock); + + return IRQ_HANDLED; +} + +static irqreturn_t arizona_fll_clock_ok(int irq, void *data) +{ + struct arizona_fll *fll = data; + + arizona_fll_dbg(fll, "clock OK\n"); + + complete(&fll->ok); + + return IRQ_HANDLED; +} + +static struct { + unsigned int min; + unsigned int max; + u16 fratio; + int ratio; +} fll_fratios[] = { + { 0, 64000, 4, 16 }, + { 64000, 128000, 3, 8 }, + { 128000, 256000, 2, 4 }, + { 256000, 1000000, 1, 2 }, + { 1000000, 13500000, 0, 1 }, +}; + +struct arizona_fll_cfg { + int n; + int theta; + int lambda; + int refdiv; + int outdiv; + int fratio; +}; + +static int arizona_calc_fll(struct arizona_fll *fll, + struct arizona_fll_cfg *cfg, + unsigned int Fref, + unsigned int Fout) +{ + unsigned int target, div, gcd_fll; + int i, ratio; + + arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, Fout); + + /* Fref must be <=13.5MHz */ + div = 1; + cfg->refdiv = 0; + while ((Fref / div) > 13500000) { + div *= 2; + cfg->refdiv++; + + if (div > 8) { + arizona_fll_err(fll, + "Can't scale %dMHz in to <=13.5MHz\n", + Fref); + return -EINVAL; + } + } + + /* Apply the division for our remaining calculations */ + Fref /= div; + + /* Fvco should be over the targt; don't check the upper bound */ + div = 1; + while (Fout * div < 90000000 * fll->vco_mult) { + div++; + if (div > 7) { + arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n", + Fout); + return -EINVAL; + } + } + target = Fout * div / fll->vco_mult; + cfg->outdiv = div; + + arizona_fll_dbg(fll, "Fvco=%dHz\n", target); + + /* Find an appropraite FLL_FRATIO and factor it out of the target */ + for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { + if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { + cfg->fratio = fll_fratios[i].fratio; + ratio = fll_fratios[i].ratio; + break; + } + } + if (i == ARRAY_SIZE(fll_fratios)) { + arizona_fll_err(fll, "Unable to find FRATIO for Fref=%uHz\n", + Fref); + return -EINVAL; + } + + cfg->n = target / (ratio * Fref); + + if (target % Fref) { + gcd_fll = gcd(target, ratio * Fref); + arizona_fll_dbg(fll, "GCD=%u\n", gcd_fll); + + cfg->theta = (target - (cfg->n * ratio * Fref)) + / gcd_fll; + cfg->lambda = (ratio * Fref) / gcd_fll; + } else { + cfg->theta = 0; + cfg->lambda = 0; + } + + arizona_fll_dbg(fll, "N=%x THETA=%x LAMBDA=%x\n", + cfg->n, cfg->theta, cfg->lambda); + arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n", + cfg->fratio, cfg->fratio, cfg->outdiv, cfg->refdiv); + + return 0; + +} + +static void arizona_apply_fll(struct arizona *arizona, unsigned int base, + struct arizona_fll_cfg *cfg, int source) +{ + regmap_update_bits(arizona->regmap, base + 3, + ARIZONA_FLL1_THETA_MASK, cfg->theta); + regmap_update_bits(arizona->regmap, base + 4, + ARIZONA_FLL1_LAMBDA_MASK, cfg->lambda); + regmap_update_bits(arizona->regmap, base + 5, + ARIZONA_FLL1_FRATIO_MASK, + cfg->fratio << ARIZONA_FLL1_FRATIO_SHIFT); + regmap_update_bits(arizona->regmap, base + 6, + ARIZONA_FLL1_CLK_REF_DIV_MASK | + ARIZONA_FLL1_CLK_REF_SRC_MASK, + cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT | + source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT); + + regmap_update_bits(arizona->regmap, base + 2, + ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK, + ARIZONA_FLL1_CTRL_UPD | cfg->n); +} + +int arizona_set_fll(struct arizona_fll *fll, int source, + unsigned int Fref, unsigned int Fout) +{ + struct arizona *arizona = fll->arizona; + struct arizona_fll_cfg cfg, sync; + unsigned int reg, val; + int syncsrc; + bool ena; + int ret; + + ret = regmap_read(arizona->regmap, fll->base + 1, ®); + if (ret != 0) { + arizona_fll_err(fll, "Failed to read current state: %d\n", + ret); + return ret; + } + ena = reg & ARIZONA_FLL1_ENA; + + if (Fout) { + /* Do we have a 32kHz reference? */ + regmap_read(arizona->regmap, ARIZONA_CLOCK_32K_1, &val); + switch (val & ARIZONA_CLK_32K_SRC_MASK) { + case ARIZONA_CLK_SRC_MCLK1: + case ARIZONA_CLK_SRC_MCLK2: + syncsrc = val & ARIZONA_CLK_32K_SRC_MASK; + break; + default: + syncsrc = -1; + } + + if (source == syncsrc) + syncsrc = -1; + + if (syncsrc >= 0) { + ret = arizona_calc_fll(fll, &sync, Fref, Fout); + if (ret != 0) + return ret; + + ret = arizona_calc_fll(fll, &cfg, 32768, Fout); + if (ret != 0) + return ret; + } else { + ret = arizona_calc_fll(fll, &cfg, Fref, Fout); + if (ret != 0) + return ret; + } + } else { + regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_ENA, 0); + regmap_update_bits(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, 0); + + if (ena) + pm_runtime_put_autosuspend(arizona->dev); + + return 0; + } + + regmap_update_bits(arizona->regmap, fll->base + 5, + ARIZONA_FLL1_OUTDIV_MASK, + cfg.outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + + if (syncsrc >= 0) { + arizona_apply_fll(arizona, fll->base, &cfg, syncsrc); + arizona_apply_fll(arizona, fll->base + 0x10, &sync, source); + } else { + arizona_apply_fll(arizona, fll->base, &cfg, source); + } + + if (!ena) + pm_runtime_get(arizona->dev); + + /* Clear any pending completions */ + try_wait_for_completion(&fll->ok); + + regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); + if (syncsrc >= 0) + regmap_update_bits(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, + ARIZONA_FLL1_SYNC_ENA); + + ret = wait_for_completion_timeout(&fll->ok, + msecs_to_jiffies(25)); + if (ret == 0) + arizona_fll_warn(fll, "Timed out waiting for lock\n"); + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_set_fll); + +int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, + int ok_irq, struct arizona_fll *fll) +{ + int ret; + + init_completion(&fll->lock); + init_completion(&fll->ok); + + fll->id = id; + fll->base = base; + fll->arizona = arizona; + + snprintf(fll->lock_name, sizeof(fll->lock_name), "FLL%d lock", id); + snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name), + "FLL%d clock OK", id); + + ret = arizona_request_irq(arizona, lock_irq, fll->lock_name, + arizona_fll_lock, fll); + if (ret != 0) { + dev_err(arizona->dev, "Failed to get FLL%d lock IRQ: %d\n", + id, ret); + } + + ret = arizona_request_irq(arizona, ok_irq, fll->clock_ok_name, + arizona_fll_clock_ok, fll); + if (ret != 0) { + dev_err(arizona->dev, "Failed to get FLL%d clock OK IRQ: %d\n", + id, ret); + } + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_fll); + +MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support"); +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h new file mode 100644 index 000000000000..59caca8865e8 --- /dev/null +++ b/sound/soc/codecs/arizona.h @@ -0,0 +1,159 @@ +/* + * arizona.h - Wolfson Arizona class device shared support + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _ASOC_ARIZONA_H +#define _ASOC_ARIZONA_H + +#include <linux/completion.h> + +#include <sound/soc.h> + +#define ARIZONA_CLK_SYSCLK 1 +#define ARIZONA_CLK_ASYNCCLK 2 + +#define ARIZONA_CLK_SRC_MCLK1 0x0 +#define ARIZONA_CLK_SRC_MCLK2 0x1 +#define ARIZONA_CLK_SRC_FLL1 0x4 +#define ARIZONA_CLK_SRC_FLL2 0x5 +#define ARIZONA_CLK_SRC_AIF1BCLK 0x8 +#define ARIZONA_CLK_SRC_AIF2BCLK 0x9 +#define ARIZONA_CLK_SRC_AIF3BCLK 0xa + +#define ARIZONA_FLL_SRC_MCLK1 0 +#define ARIZONA_FLL_SRC_MCLK2 1 +#define ARIZONA_FLL_SRC_SLIMCLK 2 +#define ARIZONA_FLL_SRC_FLL1 3 +#define ARIZONA_FLL_SRC_FLL2 4 +#define ARIZONA_FLL_SRC_AIF1BCLK 5 +#define ARIZONA_FLL_SRC_AIF2BCLK 6 +#define ARIZONA_FLL_SRC_AIF3BCLK 7 +#define ARIZONA_FLL_SRC_AIF1LRCLK 8 +#define ARIZONA_FLL_SRC_AIF2LRCLK 9 +#define ARIZONA_FLL_SRC_AIF3LRCLK 10 + +#define ARIZONA_MIXER_VOL_MASK 0x00FE +#define ARIZONA_MIXER_VOL_SHIFT 1 +#define ARIZONA_MIXER_VOL_WIDTH 7 + +#define ARIZONA_MAX_DAI 3 + +struct arizona; + +struct arizona_dai_priv { + int clk; +}; + +struct arizona_priv { + struct arizona *arizona; + int sysclk; + int asyncclk; + struct arizona_dai_priv dai[ARIZONA_MAX_DAI]; +}; + +#define ARIZONA_NUM_MIXER_INPUTS 57 + +extern const unsigned int arizona_mixer_tlv[]; +extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS]; +extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; + +#define ARIZONA_MIXER_CONTROLS(name, base) \ + SOC_SINGLE_RANGE_TLV(name " Input 1 Volume", base + 1, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv), \ + SOC_SINGLE_RANGE_TLV(name " Input 2 Volume", base + 3, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv), \ + SOC_SINGLE_RANGE_TLV(name " Input 3 Volume", base + 5, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv), \ + SOC_SINGLE_RANGE_TLV(name " Input 4 Volume", base + 7, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv) + +#define ARIZONA_MUX_ENUM_DECL(name, reg) \ + SOC_VALUE_ENUM_SINGLE_DECL(name, reg, 0, 0xff, \ + arizona_mixer_texts, arizona_mixer_values) + +#define ARIZONA_MUX_CTL_DECL(name) \ + const struct snd_kcontrol_new name##_mux = \ + SOC_DAPM_VALUE_ENUM("Route", name##_enum) + +#define ARIZONA_MIXER_ENUMS(name, base_reg) \ + static ARIZONA_MUX_ENUM_DECL(name##_in1_enum, base_reg); \ + static ARIZONA_MUX_ENUM_DECL(name##_in2_enum, base_reg + 2); \ + static ARIZONA_MUX_ENUM_DECL(name##_in3_enum, base_reg + 4); \ + static ARIZONA_MUX_ENUM_DECL(name##_in4_enum, base_reg + 6); \ + static ARIZONA_MUX_CTL_DECL(name##_in1); \ + static ARIZONA_MUX_CTL_DECL(name##_in2); \ + static ARIZONA_MUX_CTL_DECL(name##_in3); \ + static ARIZONA_MUX_CTL_DECL(name##_in4) + +#define ARIZONA_MUX(name, ctrl) \ + SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl) + +#define ARIZONA_MIXER_WIDGETS(name, name_str) \ + ARIZONA_MUX(name_str " Input 1", &name##_in1_mux), \ + ARIZONA_MUX(name_str " Input 2", &name##_in2_mux), \ + ARIZONA_MUX(name_str " Input 3", &name##_in3_mux), \ + ARIZONA_MUX(name_str " Input 4", &name##_in4_mux), \ + SND_SOC_DAPM_MIXER(name_str " Mixer", SND_SOC_NOPM, 0, 0, NULL, 0) + +#define ARIZONA_MIXER_ROUTES(widget, name) \ + { widget, NULL, name " Mixer" }, \ + { name " Mixer", NULL, name " Input 1" }, \ + { name " Mixer", NULL, name " Input 2" }, \ + { name " Mixer", NULL, name " Input 3" }, \ + { name " Mixer", NULL, name " Input 4" }, \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 1"), \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 2"), \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 3"), \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 4") + +extern const struct soc_enum arizona_lhpf1_mode; +extern const struct soc_enum arizona_lhpf2_mode; +extern const struct soc_enum arizona_lhpf3_mode; +extern const struct soc_enum arizona_lhpf4_mode; + +extern int arizona_in_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event); +extern int arizona_out_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event); + +extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir); + +extern const struct snd_soc_dai_ops arizona_dai_ops; + +#define ARIZONA_FLL_NAME_LEN 20 + +struct arizona_fll { + struct arizona *arizona; + int id; + unsigned int base; + unsigned int vco_mult; + struct completion lock; + struct completion ok; + + char lock_name[ARIZONA_FLL_NAME_LEN]; + char clock_ok_name[ARIZONA_FLL_NAME_LEN]; +}; + +extern int arizona_init_fll(struct arizona *arizona, int id, int base, + int lock_irq, int ok_irq, struct arizona_fll *fll); +extern int arizona_set_fll(struct arizona_fll *fll, int source, + unsigned int Fref, unsigned int Fout); + +extern int arizona_init_dai(struct arizona_priv *priv, int dai); + +#endif diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 1d672f528662..047917f0b8ae 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -307,8 +307,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); int ret; unsigned int i; @@ -600,10 +599,12 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec) static int cs4270_soc_resume(struct snd_soc_codec *codec) { struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); - int reg; + int reg, ret; - regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), - cs4270->supplies); + ret = regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + if (ret != 0) + return ret; /* In case the device was put to hard reset during sleep, we need to * wait 500ns here before any I2C communication. */ diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index bf7141280a74..9eb01d7d58a3 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -318,8 +318,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); int i, ret; unsigned int ratio, val; diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index a8bf588e8740..091d0193f507 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -141,15 +141,15 @@ static const struct soc_enum cs42l51_chan_mix = static const struct snd_kcontrol_new cs42l51_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, - 7, 0xffffff99, 0x18, adc_pcm_tlv), + 6, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("PCM Playback Switch", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1), SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", CS42L51_AOUTA_VOL, CS42L51_AOUTB_VOL, - 8, 0xffffff19, 0x18, aout_tlv), + 0, 0x34, 0xE4, aout_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, - 7, 0xffffff99, 0x18, adc_pcm_tlv), + 6, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("ADC Mixer Switch", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), @@ -356,8 +356,7 @@ static int cs42l51_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); int ret; unsigned int i; diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c new file mode 100644 index 000000000000..628daf6a1d97 --- /dev/null +++ b/sound/soc/codecs/cs42l52.c @@ -0,0 +1,1284 @@ +/* + * cs42l52.c -- CS42L52 ALSA SoC audio driver + * + * Copyright 2012 CirrusLogic, Inc. + * + * Author: Georgi Vlaev <joe@nucleusys.com> + * Author: Brian Austin <brian.austin@cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/input.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs42l52.h> +#include "cs42l52.h" + +struct sp_config { + u8 spc, format, spfs; + u32 srate; +}; + +struct cs42l52_private { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct device *dev; + struct sp_config config; + struct cs42l52_platform_data pdata; + u32 sysclk; + u8 mclksel; + u32 mclk; + u8 flags; +#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) + struct input_dev *beep; + struct work_struct beep_work; + int beep_rate; +#endif +}; + +static const struct reg_default cs42l52_reg_defaults[] = { + { CS42L52_PWRCTL1, 0x9F }, /* r02 PWRCTL 1 */ + { CS42L52_PWRCTL2, 0x07 }, /* r03 PWRCTL 2 */ + { CS42L52_PWRCTL3, 0xFF }, /* r04 PWRCTL 3 */ + { CS42L52_CLK_CTL, 0xA0 }, /* r05 Clocking Ctl */ + { CS42L52_IFACE_CTL1, 0x00 }, /* r06 Interface Ctl 1 */ + { CS42L52_ADC_PGA_A, 0x80 }, /* r08 Input A Select */ + { CS42L52_ADC_PGA_B, 0x80 }, /* r09 Input B Select */ + { CS42L52_ANALOG_HPF_CTL, 0xA5 }, /* r0A Analog HPF Ctl */ + { CS42L52_ADC_HPF_FREQ, 0x00 }, /* r0B ADC HPF Corner Freq */ + { CS42L52_ADC_MISC_CTL, 0x00 }, /* r0C Misc. ADC Ctl */ + { CS42L52_PB_CTL1, 0x60 }, /* r0D Playback Ctl 1 */ + { CS42L52_MISC_CTL, 0x02 }, /* r0E Misc. Ctl */ + { CS42L52_PB_CTL2, 0x00 }, /* r0F Playback Ctl 2 */ + { CS42L52_MICA_CTL, 0x00 }, /* r10 MICA Amp Ctl */ + { CS42L52_MICB_CTL, 0x00 }, /* r11 MICB Amp Ctl */ + { CS42L52_PGAA_CTL, 0x00 }, /* r12 PGAA Vol, Misc. */ + { CS42L52_PGAB_CTL, 0x00 }, /* r13 PGAB Vol, Misc. */ + { CS42L52_PASSTHRUA_VOL, 0x00 }, /* r14 Bypass A Vol */ + { CS42L52_PASSTHRUB_VOL, 0x00 }, /* r15 Bypass B Vol */ + { CS42L52_ADCA_VOL, 0x00 }, /* r16 ADCA Volume */ + { CS42L52_ADCB_VOL, 0x00 }, /* r17 ADCB Volume */ + { CS42L52_ADCA_MIXER_VOL, 0x80 }, /* r18 ADCA Mixer Volume */ + { CS42L52_ADCB_MIXER_VOL, 0x80 }, /* r19 ADCB Mixer Volume */ + { CS42L52_PCMA_MIXER_VOL, 0x00 }, /* r1A PCMA Mixer Volume */ + { CS42L52_PCMB_MIXER_VOL, 0x00 }, /* r1B PCMB Mixer Volume */ + { CS42L52_BEEP_FREQ, 0x00 }, /* r1C Beep Freq on Time */ + { CS42L52_BEEP_VOL, 0x00 }, /* r1D Beep Volume off Time */ + { CS42L52_BEEP_TONE_CTL, 0x00 }, /* r1E Beep Tone Cfg. */ + { CS42L52_TONE_CTL, 0x00 }, /* r1F Tone Ctl */ + { CS42L52_MASTERA_VOL, 0x88 }, /* r20 Master A Volume */ + { CS42L52_MASTERB_VOL, 0x00 }, /* r21 Master B Volume */ + { CS42L52_HPA_VOL, 0x00 }, /* r22 Headphone A Volume */ + { CS42L52_HPB_VOL, 0x00 }, /* r23 Headphone B Volume */ + { CS42L52_SPKA_VOL, 0x00 }, /* r24 Speaker A Volume */ + { CS42L52_SPKB_VOL, 0x00 }, /* r25 Speaker B Volume */ + { CS42L52_ADC_PCM_MIXER, 0x00 }, /* r26 Channel Mixer and Swap */ + { CS42L52_LIMITER_CTL1, 0x00 }, /* r27 Limit Ctl 1 Thresholds */ + { CS42L52_LIMITER_CTL2, 0x7F }, /* r28 Limit Ctl 2 Release Rate */ + { CS42L52_LIMITER_AT_RATE, 0xC0 }, /* r29 Limiter Attack Rate */ + { CS42L52_ALC_CTL, 0x00 }, /* r2A ALC Ctl 1 Attack Rate */ + { CS42L52_ALC_RATE, 0x3F }, /* r2B ALC Release Rate */ + { CS42L52_ALC_THRESHOLD, 0x3f }, /* r2C ALC Thresholds */ + { CS42L52_NOISE_GATE_CTL, 0x00 }, /* r2D Noise Gate Ctl */ + { CS42L52_CLK_STATUS, 0x00 }, /* r2E Overflow and Clock Status */ + { CS42L52_BATT_COMPEN, 0x00 }, /* r2F battery Compensation */ + { CS42L52_BATT_LEVEL, 0x00 }, /* r30 VP Battery Level */ + { CS42L52_SPK_STATUS, 0x00 }, /* r31 Speaker Status */ + { CS42L52_TEM_CTL, 0x3B }, /* r32 Temp Ctl */ + { CS42L52_THE_FOLDBACK, 0x00 }, /* r33 Foldback */ +}; + +static bool cs42l52_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L52_CHIP: + case CS42L52_PWRCTL1: + case CS42L52_PWRCTL2: + case CS42L52_PWRCTL3: + case CS42L52_CLK_CTL: + case CS42L52_IFACE_CTL1: + case CS42L52_IFACE_CTL2: + case CS42L52_ADC_PGA_A: + case CS42L52_ADC_PGA_B: + case CS42L52_ANALOG_HPF_CTL: + case CS42L52_ADC_HPF_FREQ: + case CS42L52_ADC_MISC_CTL: + case CS42L52_PB_CTL1: + case CS42L52_MISC_CTL: + case CS42L52_PB_CTL2: + case CS42L52_MICA_CTL: + case CS42L52_MICB_CTL: + case CS42L52_PGAA_CTL: + case CS42L52_PGAB_CTL: + case CS42L52_PASSTHRUA_VOL: + case CS42L52_PASSTHRUB_VOL: + case CS42L52_ADCA_VOL: + case CS42L52_ADCB_VOL: + case CS42L52_ADCA_MIXER_VOL: + case CS42L52_ADCB_MIXER_VOL: + case CS42L52_PCMA_MIXER_VOL: + case CS42L52_PCMB_MIXER_VOL: + case CS42L52_BEEP_FREQ: + case CS42L52_BEEP_VOL: + case CS42L52_BEEP_TONE_CTL: + case CS42L52_TONE_CTL: + case CS42L52_MASTERA_VOL: + case CS42L52_MASTERB_VOL: + case CS42L52_HPA_VOL: + case CS42L52_HPB_VOL: + case CS42L52_SPKA_VOL: + case CS42L52_SPKB_VOL: + case CS42L52_ADC_PCM_MIXER: + case CS42L52_LIMITER_CTL1: + case CS42L52_LIMITER_CTL2: + case CS42L52_LIMITER_AT_RATE: + case CS42L52_ALC_CTL: + case CS42L52_ALC_RATE: + case CS42L52_ALC_THRESHOLD: + case CS42L52_NOISE_GATE_CTL: + case CS42L52_CLK_STATUS: + case CS42L52_BATT_COMPEN: + case CS42L52_BATT_LEVEL: + case CS42L52_SPK_STATUS: + case CS42L52_TEM_CTL: + case CS42L52_THE_FOLDBACK: + case CS42L52_CHARGE_PUMP: + return true; + default: + return false; + } +} + +static bool cs42l52_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L52_IFACE_CTL2: + case CS42L52_CLK_STATUS: + case CS42L52_BATT_LEVEL: + case CS42L52_SPK_STATUS: + case CS42L52_CHARGE_PUMP: + return 1; + default: + return 0; + } +} + +static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0); + +static DECLARE_TLV_DB_SCALE(hpd_tlv, -9600, 50, 1); + +static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0); + +static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); + +static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); + +static const unsigned int limiter_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), + 3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0), +}; + +static const char * const cs42l52_adca_text[] = { + "Input1A", "Input2A", "Input3A", "Input4A", "PGA Input Left"}; + +static const char * const cs42l52_adcb_text[] = { + "Input1B", "Input2B", "Input3B", "Input4B", "PGA Input Right"}; + +static const struct soc_enum adca_enum = + SOC_ENUM_SINGLE(CS42L52_ADC_PGA_A, 5, + ARRAY_SIZE(cs42l52_adca_text), cs42l52_adca_text); + +static const struct soc_enum adcb_enum = + SOC_ENUM_SINGLE(CS42L52_ADC_PGA_B, 5, + ARRAY_SIZE(cs42l52_adcb_text), cs42l52_adcb_text); + +static const struct snd_kcontrol_new adca_mux = + SOC_DAPM_ENUM("Left ADC Input Capture Mux", adca_enum); + +static const struct snd_kcontrol_new adcb_mux = + SOC_DAPM_ENUM("Right ADC Input Capture Mux", adcb_enum); + +static const char * const mic_bias_level_text[] = { + "0.5 +VA", "0.6 +VA", "0.7 +VA", + "0.8 +VA", "0.83 +VA", "0.91 +VA" +}; + +static const struct soc_enum mic_bias_level_enum = + SOC_ENUM_SINGLE(CS42L52_IFACE_CTL1, 0, + ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text); + +static const char * const cs42l52_mic_text[] = { "Single", "Differential" }; + +static const struct soc_enum mica_enum = + SOC_ENUM_SINGLE(CS42L52_MICA_CTL, 5, + ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text); + +static const struct soc_enum micb_enum = + SOC_ENUM_SINGLE(CS42L52_MICB_CTL, 5, + ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text); + +static const struct snd_kcontrol_new mica_mux = + SOC_DAPM_ENUM("Left Mic Input Capture Mux", mica_enum); + +static const struct snd_kcontrol_new micb_mux = + SOC_DAPM_ENUM("Right Mic Input Capture Mux", micb_enum); + +static const char * const digital_output_mux_text[] = {"ADC", "DSP"}; + +static const struct soc_enum digital_output_mux_enum = + SOC_ENUM_SINGLE(CS42L52_ADC_MISC_CTL, 6, + ARRAY_SIZE(digital_output_mux_text), + digital_output_mux_text); + +static const struct snd_kcontrol_new digital_output_mux = + SOC_DAPM_ENUM("Digital Output Mux", digital_output_mux_enum); + +static const char * const hp_gain_num_text[] = { + "0.3959", "0.4571", "0.5111", "0.6047", + "0.7099", "0.8399", "1.000", "1.1430" +}; + +static const struct soc_enum hp_gain_enum = + SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 4, + ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text); + +static const char * const beep_pitch_text[] = { + "C4", "C5", "D5", "E5", "F5", "G5", "A5", "B5", + "C6", "D6", "E6", "F6", "G6", "A6", "B6", "C7" +}; + +static const struct soc_enum beep_pitch_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 4, + ARRAY_SIZE(beep_pitch_text), beep_pitch_text); + +static const char * const beep_ontime_text[] = { + "86 ms", "430 ms", "780 ms", "1.20 s", "1.50 s", + "1.80 s", "2.20 s", "2.50 s", "2.80 s", "3.20 s", + "3.50 s", "3.80 s", "4.20 s", "4.50 s", "4.80 s", "5.20 s" +}; + +static const struct soc_enum beep_ontime_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 0, + ARRAY_SIZE(beep_ontime_text), beep_ontime_text); + +static const char * const beep_offtime_text[] = { + "1.23 s", "2.58 s", "3.90 s", "5.20 s", + "6.60 s", "8.05 s", "9.35 s", "10.80 s" +}; + +static const struct soc_enum beep_offtime_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_VOL, 5, + ARRAY_SIZE(beep_offtime_text), beep_offtime_text); + +static const char * const beep_config_text[] = { + "Off", "Single", "Multiple", "Continuous" +}; + +static const struct soc_enum beep_config_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 6, + ARRAY_SIZE(beep_config_text), beep_config_text); + +static const char * const beep_bass_text[] = { + "50 Hz", "100 Hz", "200 Hz", "250 Hz" +}; + +static const struct soc_enum beep_bass_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 1, + ARRAY_SIZE(beep_bass_text), beep_bass_text); + +static const char * const beep_treble_text[] = { + "5 kHz", "7 kHz", "10 kHz", " 15 kHz" +}; + +static const struct soc_enum beep_treble_enum = + SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 3, + ARRAY_SIZE(beep_treble_text), beep_treble_text); + +static const char * const ng_threshold_text[] = { + "-34dB", "-37dB", "-40dB", "-43dB", + "-46dB", "-52dB", "-58dB", "-64dB" +}; + +static const struct soc_enum ng_threshold_enum = + SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 2, + ARRAY_SIZE(ng_threshold_text), ng_threshold_text); + +static const char * const cs42l52_ng_delay_text[] = { + "50ms", "100ms", "150ms", "200ms"}; + +static const struct soc_enum ng_delay_enum = + SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 0, + ARRAY_SIZE(cs42l52_ng_delay_text), cs42l52_ng_delay_text); + +static const char * const cs42l52_ng_type_text[] = { + "Apply Specific", "Apply All" +}; + +static const struct soc_enum ng_type_enum = + SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 6, + ARRAY_SIZE(cs42l52_ng_type_text), cs42l52_ng_type_text); + +static const char * const left_swap_text[] = { + "Left", "LR 2", "Right"}; + +static const char * const right_swap_text[] = { + "Right", "LR 2", "Left"}; + +static const unsigned int swap_values[] = { 0, 1, 3 }; + +static const struct soc_enum adca_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 1, + ARRAY_SIZE(left_swap_text), + left_swap_text, + swap_values); + +static const struct snd_kcontrol_new adca_mixer = + SOC_DAPM_ENUM("Route", adca_swap_enum); + +static const struct soc_enum pcma_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 1, + ARRAY_SIZE(left_swap_text), + left_swap_text, + swap_values); + +static const struct snd_kcontrol_new pcma_mixer = + SOC_DAPM_ENUM("Route", pcma_swap_enum); + +static const struct soc_enum adcb_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 1, + ARRAY_SIZE(right_swap_text), + right_swap_text, + swap_values); + +static const struct snd_kcontrol_new adcb_mixer = + SOC_DAPM_ENUM("Route", adcb_swap_enum); + +static const struct soc_enum pcmb_swap_enum = + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 1, + ARRAY_SIZE(right_swap_text), + right_swap_text, + swap_values); + +static const struct snd_kcontrol_new pcmb_mixer = + SOC_DAPM_ENUM("Route", pcmb_swap_enum); + + +static const struct snd_kcontrol_new passthrul_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_MISC_CTL, 6, 1, 0); + +static const struct snd_kcontrol_new passthrur_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_MISC_CTL, 7, 1, 0); + +static const struct snd_kcontrol_new spkl_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 0, 1, 1); + +static const struct snd_kcontrol_new spkr_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 2, 1, 1); + +static const struct snd_kcontrol_new hpl_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 4, 1, 1); + +static const struct snd_kcontrol_new hpr_ctl = + SOC_DAPM_SINGLE("Switch", CS42L52_PWRCTL3, 6, 1, 1); + +static const struct snd_kcontrol_new cs42l52_snd_controls[] = { + + SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L52_MASTERA_VOL, + CS42L52_MASTERB_VOL, 0, 0x34, 0xE4, hl_tlv), + + SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL, + CS42L52_HPB_VOL, 0, 0x34, 0xCC, hpd_tlv), + + SOC_ENUM("Headphone Analog Gain", hp_gain_enum), + + SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, + CS42L52_SPKB_VOL, 7, 0x1, 0xff, hl_tlv), + + SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, + CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv), + + SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), + + SOC_DOUBLE_R_TLV("MIC Gain Volume", CS42L52_MICA_CTL, + CS42L52_MICB_CTL, 0, 0x10, 0, mic_tlv), + + SOC_ENUM("MIC Bias Level", mic_bias_level_enum), + + SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL, + CS42L52_ADCB_VOL, 7, 0x80, 0xA0, ipd_tlv), + SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", + CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, + 6, 0x7f, 0x19, ipd_tlv), + + SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), + + SOC_DOUBLE_R("ADC Mixer Switch", CS42L52_ADCA_MIXER_VOL, + CS42L52_ADCB_MIXER_VOL, 7, 1, 1), + + SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL, + CS42L52_PGAB_CTL, 0, 0x28, 0x30, pga_tlv), + + SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", + CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, + 6, 0x7f, 0x19, hl_tlv), + SOC_DOUBLE_R("PCM Mixer Switch", + CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1), + + SOC_ENUM("Beep Config", beep_config_enum), + SOC_ENUM("Beep Pitch", beep_pitch_enum), + SOC_ENUM("Beep on Time", beep_ontime_enum), + SOC_ENUM("Beep off Time", beep_offtime_enum), + SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv), + SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1), + SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), + SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), + + SOC_SINGLE("Tone Control Switch", CS42L52_BEEP_TONE_CTL, 0, 1, 1), + SOC_SINGLE_TLV("Treble Gain Volume", + CS42L52_TONE_CTL, 4, 15, 1, hl_tlv), + SOC_SINGLE_TLV("Bass Gain Volume", + CS42L52_TONE_CTL, 0, 15, 1, hl_tlv), + + /* Limiter */ + SOC_SINGLE_TLV("Limiter Max Threshold Volume", + CS42L52_LIMITER_CTL1, 5, 7, 0, limiter_tlv), + SOC_SINGLE_TLV("Limiter Cushion Threshold Volume", + CS42L52_LIMITER_CTL1, 2, 7, 0, limiter_tlv), + SOC_SINGLE_TLV("Limiter Release Rate Volume", + CS42L52_LIMITER_CTL2, 0, 63, 0, limiter_tlv), + SOC_SINGLE_TLV("Limiter Attack Rate Volume", + CS42L52_LIMITER_AT_RATE, 0, 63, 0, limiter_tlv), + + SOC_SINGLE("Limiter SR Switch", CS42L52_LIMITER_CTL1, 1, 1, 0), + SOC_SINGLE("Limiter ZC Switch", CS42L52_LIMITER_CTL1, 0, 1, 0), + SOC_SINGLE("Limiter Switch", CS42L52_LIMITER_CTL2, 7, 1, 0), + + /* ALC */ + SOC_SINGLE_TLV("ALC Attack Rate Volume", CS42L52_ALC_CTL, + 0, 63, 0, limiter_tlv), + SOC_SINGLE_TLV("ALC Release Rate Volume", CS42L52_ALC_RATE, + 0, 63, 0, limiter_tlv), + SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L52_ALC_THRESHOLD, + 5, 7, 0, limiter_tlv), + SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L52_ALC_THRESHOLD, + 2, 7, 0, limiter_tlv), + + SOC_DOUBLE_R("ALC SR Capture Switch", CS42L52_PGAA_CTL, + CS42L52_PGAB_CTL, 7, 1, 1), + SOC_DOUBLE_R("ALC ZC Capture Switch", CS42L52_PGAA_CTL, + CS42L52_PGAB_CTL, 6, 1, 1), + SOC_DOUBLE("ALC Capture Switch", CS42L52_ALC_CTL, 6, 7, 1, 0), + + /* Noise gate */ + SOC_ENUM("NG Type Switch", ng_type_enum), + SOC_SINGLE("NG Enable Switch", CS42L52_NOISE_GATE_CTL, 6, 1, 0), + SOC_SINGLE("NG Boost Switch", CS42L52_NOISE_GATE_CTL, 5, 1, 1), + SOC_ENUM("NG Threshold", ng_threshold_enum), + SOC_ENUM("NG Delay", ng_delay_enum), + + SOC_DOUBLE("HPF Switch", CS42L52_ANALOG_HPF_CTL, 5, 7, 1, 0), + + SOC_DOUBLE("Analog SR Switch", CS42L52_ANALOG_HPF_CTL, 1, 3, 1, 1), + SOC_DOUBLE("Analog ZC Switch", CS42L52_ANALOG_HPF_CTL, 0, 2, 1, 1), + SOC_SINGLE("Digital SR Switch", CS42L52_MISC_CTL, 1, 1, 0), + SOC_SINGLE("Digital ZC Switch", CS42L52_MISC_CTL, 0, 1, 0), + SOC_SINGLE("Deemphasis Switch", CS42L52_MISC_CTL, 2, 1, 0), + + SOC_SINGLE("Batt Compensation Switch", CS42L52_BATT_COMPEN, 7, 1, 0), + SOC_SINGLE("Batt VP Monitor Switch", CS42L52_BATT_COMPEN, 6, 1, 0), + SOC_SINGLE("Batt VP ref", CS42L52_BATT_COMPEN, 0, 0x0f, 0), + + SOC_SINGLE("PGA AIN1L Switch", CS42L52_ADC_PGA_A, 0, 1, 0), + SOC_SINGLE("PGA AIN1R Switch", CS42L52_ADC_PGA_B, 0, 1, 0), + SOC_SINGLE("PGA AIN2L Switch", CS42L52_ADC_PGA_A, 1, 1, 0), + SOC_SINGLE("PGA AIN2R Switch", CS42L52_ADC_PGA_B, 1, 1, 0), + + SOC_SINGLE("PGA AIN3L Switch", CS42L52_ADC_PGA_A, 2, 1, 0), + SOC_SINGLE("PGA AIN3R Switch", CS42L52_ADC_PGA_B, 2, 1, 0), + + SOC_SINGLE("PGA AIN4L Switch", CS42L52_ADC_PGA_A, 3, 1, 0), + SOC_SINGLE("PGA AIN4R Switch", CS42L52_ADC_PGA_B, 3, 1, 0), + + SOC_SINGLE("PGA MICA Switch", CS42L52_ADC_PGA_A, 4, 1, 0), + SOC_SINGLE("PGA MICB Switch", CS42L52_ADC_PGA_B, 4, 1, 0), + +}; + +static const struct snd_soc_dapm_widget cs42l52_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + SND_SOC_DAPM_INPUT("AIN3L"), + SND_SOC_DAPM_INPUT("AIN3R"), + SND_SOC_DAPM_INPUT("AIN4L"), + SND_SOC_DAPM_INPUT("AIN4R"), + SND_SOC_DAPM_INPUT("MICA"), + SND_SOC_DAPM_INPUT("MICB"), + SND_SOC_DAPM_SIGGEN("Beep"), + + SND_SOC_DAPM_AIF_OUT("AIFOUTL", NULL, 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIFOUTR", NULL, 0, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("MICA Mux", SND_SOC_NOPM, 0, 0, &mica_mux), + SND_SOC_DAPM_MUX("MICB Mux", SND_SOC_NOPM, 0, 0, &micb_mux), + + SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L52_PWRCTL1, 1, 1), + SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L52_PWRCTL1, 2, 1), + SND_SOC_DAPM_PGA("PGA Left", CS42L52_PWRCTL1, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("PGA Right", CS42L52_PWRCTL1, 4, 1, NULL, 0), + + SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adca_mux), + SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcb_mux), + + SND_SOC_DAPM_MUX("ADC Left Swap", SND_SOC_NOPM, + 0, 0, &adca_mixer), + SND_SOC_DAPM_MUX("ADC Right Swap", SND_SOC_NOPM, + 0, 0, &adcb_mixer), + + SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM, + 0, 0, &digital_output_mux), + + SND_SOC_DAPM_PGA("PGA MICA", CS42L52_PWRCTL2, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("PGA MICB", CS42L52_PWRCTL2, 2, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Mic Bias", CS42L52_PWRCTL2, 0, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Charge Pump", CS42L52_PWRCTL1, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_IN("AIFINL", NULL, 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFINR", NULL, 0, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_DAC("DAC Left", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Right", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_SWITCH("Bypass Left", CS42L52_MISC_CTL, + 6, 0, &passthrul_ctl), + SND_SOC_DAPM_SWITCH("Bypass Right", CS42L52_MISC_CTL, + 7, 0, &passthrur_ctl), + + SND_SOC_DAPM_MUX("PCM Left Swap", SND_SOC_NOPM, + 0, 0, &pcma_mixer), + SND_SOC_DAPM_MUX("PCM Right Swap", SND_SOC_NOPM, + 0, 0, &pcmb_mixer), + + SND_SOC_DAPM_SWITCH("HP Left Amp", SND_SOC_NOPM, 0, 0, &hpl_ctl), + SND_SOC_DAPM_SWITCH("HP Right Amp", SND_SOC_NOPM, 0, 0, &hpr_ctl), + + SND_SOC_DAPM_SWITCH("SPK Left Amp", SND_SOC_NOPM, 0, 0, &spkl_ctl), + SND_SOC_DAPM_SWITCH("SPK Right Amp", SND_SOC_NOPM, 0, 0, &spkr_ctl), + + SND_SOC_DAPM_OUTPUT("HPOUTA"), + SND_SOC_DAPM_OUTPUT("HPOUTB"), + SND_SOC_DAPM_OUTPUT("SPKOUTA"), + SND_SOC_DAPM_OUTPUT("SPKOUTB"), + +}; + +static const struct snd_soc_dapm_route cs42l52_audio_map[] = { + + {"Capture", NULL, "AIFOUTL"}, + {"Capture", NULL, "AIFOUTL"}, + + {"AIFOUTL", NULL, "Output Mux"}, + {"AIFOUTR", NULL, "Output Mux"}, + + {"Output Mux", "ADC", "ADC Left"}, + {"Output Mux", "ADC", "ADC Right"}, + + {"ADC Left", NULL, "Charge Pump"}, + {"ADC Right", NULL, "Charge Pump"}, + + {"Charge Pump", NULL, "ADC Left Mux"}, + {"Charge Pump", NULL, "ADC Right Mux"}, + + {"ADC Left Mux", "Input1A", "AIN1L"}, + {"ADC Right Mux", "Input1B", "AIN1R"}, + {"ADC Left Mux", "Input2A", "AIN2L"}, + {"ADC Right Mux", "Input2B", "AIN2R"}, + {"ADC Left Mux", "Input3A", "AIN3L"}, + {"ADC Right Mux", "Input3B", "AIN3R"}, + {"ADC Left Mux", "Input4A", "AIN4L"}, + {"ADC Right Mux", "Input4B", "AIN4R"}, + {"ADC Left Mux", "PGA Input Left", "PGA Left"}, + {"ADC Right Mux", "PGA Input Right" , "PGA Right"}, + + {"PGA Left", "Switch", "AIN1L"}, + {"PGA Right", "Switch", "AIN1R"}, + {"PGA Left", "Switch", "AIN2L"}, + {"PGA Right", "Switch", "AIN2R"}, + {"PGA Left", "Switch", "AIN3L"}, + {"PGA Right", "Switch", "AIN3R"}, + {"PGA Left", "Switch", "AIN4L"}, + {"PGA Right", "Switch", "AIN4R"}, + + {"PGA Left", "Switch", "PGA MICA"}, + {"PGA MICA", NULL, "MICA"}, + + {"PGA Right", "Switch", "PGA MICB"}, + {"PGA MICB", NULL, "MICB"}, + + {"HPOUTA", NULL, "HP Left Amp"}, + {"HPOUTB", NULL, "HP Right Amp"}, + {"HP Left Amp", NULL, "Bypass Left"}, + {"HP Right Amp", NULL, "Bypass Right"}, + {"Bypass Left", "Switch", "PGA Left"}, + {"Bypass Right", "Switch", "PGA Right"}, + {"HP Left Amp", "Switch", "DAC Left"}, + {"HP Right Amp", "Switch", "DAC Right"}, + + {"SPKOUTA", NULL, "SPK Left Amp"}, + {"SPKOUTB", NULL, "SPK Right Amp"}, + + {"SPK Left Amp", NULL, "Beep"}, + {"SPK Right Amp", NULL, "Beep"}, + {"SPK Left Amp", "Switch", "Playback"}, + {"SPK Right Amp", "Switch", "Playback"}, + + {"DAC Left", NULL, "Beep"}, + {"DAC Right", NULL, "Beep"}, + {"DAC Left", NULL, "Playback"}, + {"DAC Right", NULL, "Playback"}, + + {"Output Mux", "DSP", "Playback"}, + {"Output Mux", "DSP", "Playback"}, + + {"AIFINL", NULL, "Playback"}, + {"AIFINR", NULL, "Playback"}, + +}; + +struct cs42l52_clk_para { + u32 mclk; + u32 rate; + u8 speed; + u8 group; + u8 videoclk; + u8 ratio; + u8 mclkdiv2; +}; + +static const struct cs42l52_clk_para clk_map_table[] = { + /*8k*/ + {12288000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 8000, CLK_QS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 8000, CLK_QS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 0}, + + /*11.025k*/ + {11289600, 11025, CLK_QS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 11025, CLK_QS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /*16k*/ + {12288000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 16000, CLK_HS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 16000, CLK_HS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 1}, + + /*22.05k*/ + {11289600, 22050, CLK_HS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 22050, CLK_HS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /* 32k */ + {12288000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 32000, CLK_SS_MODE, CLK_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 32000, CLK_SS_MODE, CLK_32K, CLK_27M_MCLK, CLK_R_125, 0}, + + /* 44.1k */ + {11289600, 44100, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 44100, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /* 48k */ + {12288000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 1}, + {27000000, 48000, CLK_SS_MODE, CLK_NO_32K, CLK_27M_MCLK, CLK_R_125, 1}, + + /* 88.2k */ + {11289600, 88200, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {16934400, 88200, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + + /* 96k */ + {12288000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {18432000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_128, 0}, + {12000000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 0}, + {24000000, 96000, CLK_DS_MODE, CLK_NO_32K, CLK_NO_27M, CLK_R_125, 1}, +}; + +static int cs42l52_get_clk(int mclk, int rate) +{ + int i, ret = 0; + u_int mclk1, mclk2 = 0; + + for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) { + if (clk_map_table[i].rate == rate) { + mclk1 = clk_map_table[i].mclk; + if (abs(mclk - mclk1) < abs(mclk - mclk2)) { + mclk2 = mclk1; + ret = i; + } + } + } + if (ret > ARRAY_SIZE(clk_map_table)) + return -EINVAL; + return ret; +} + +static int cs42l52_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + + if ((freq >= CS42L52_MIN_CLK) && (freq <= CS42L52_MAX_CLK)) { + cs42l52->sysclk = freq; + } else { + dev_err(codec->dev, "Invalid freq paramter\n"); + return -EINVAL; + } + return 0; +} + +static int cs42l52_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface = CS42L52_IFACE_CTL1_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iface = CS42L52_IFACE_CTL1_SLAVE; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= CS42L52_IFACE_CTL1_ADC_FMT_I2S | + CS42L52_IFACE_CTL1_DAC_FMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface |= CS42L52_IFACE_CTL1_DAC_FMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= CS42L52_IFACE_CTL1_ADC_FMT_LEFT_J | + CS42L52_IFACE_CTL1_DAC_FMT_LEFT_J; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= CS42L52_IFACE_CTL1_DSP_MODE_EN; + break; + case SND_SOC_DAIFMT_DSP_B: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= CS42L52_IFACE_CTL1_INV_SCLK; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= CS42L52_IFACE_CTL1_INV_SCLK; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + default: + ret = -EINVAL; + } + cs42l52->config.format = iface; + snd_soc_write(codec, CS42L52_IFACE_CTL1, cs42l52->config.format); + + return 0; +} + +static int cs42l52_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) + snd_soc_update_bits(codec, CS42L52_PB_CTL1, + CS42L52_PB_CTL1_MUTE_MASK, + CS42L52_PB_CTL1_MUTE); + else + snd_soc_update_bits(codec, CS42L52_PB_CTL1, + CS42L52_PB_CTL1_MUTE_MASK, + CS42L52_PB_CTL1_UNMUTE); + + return 0; +} + +static int cs42l52_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + u32 clk = 0; + int index; + + index = cs42l52_get_clk(cs42l52->sysclk, params_rate(params)); + if (index >= 0) { + cs42l52->sysclk = clk_map_table[index].mclk; + + clk |= (clk_map_table[index].speed << CLK_SPEED_SHIFT) | + (clk_map_table[index].group << CLK_32K_SR_SHIFT) | + (clk_map_table[index].videoclk << CLK_27M_MCLK_SHIFT) | + (clk_map_table[index].ratio << CLK_RATIO_SHIFT) | + clk_map_table[index].mclkdiv2; + + snd_soc_write(codec, CS42L52_CLK_CTL, clk); + } else { + dev_err(codec->dev, "can't get correct mclk\n"); + return -EINVAL; + } + + return 0; +} + +static int cs42l52_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, CS42L52_PWRCTL1, + CS42L52_PWRCTL1_PDN_CODEC, 0); + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + regcache_cache_only(cs42l52->regmap, false); + regcache_sync(cs42l52->regmap); + } + snd_soc_write(codec, CS42L52_PWRCTL1, CS42L52_PWRCTL1_PDN_ALL); + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, CS42L52_PWRCTL1, CS42L52_PWRCTL1_PDN_ALL); + regcache_cache_only(cs42l52->regmap, true); + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +#define CS42L52_RATES (SNDRV_PCM_RATE_8000_96000) + +#define CS42L52_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_U18_3LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE) + +static struct snd_soc_dai_ops cs42l52_ops = { + .hw_params = cs42l52_pcm_hw_params, + .digital_mute = cs42l52_digital_mute, + .set_fmt = cs42l52_set_fmt, + .set_sysclk = cs42l52_set_sysclk, +}; + +static struct snd_soc_dai_driver cs42l52_dai = { + .name = "cs42l52", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L52_RATES, + .formats = CS42L52_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L52_RATES, + .formats = CS42L52_FORMATS, + }, + .ops = &cs42l52_ops, +}; + +static int cs42l52_suspend(struct snd_soc_codec *codec) +{ + cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cs42l52_resume(struct snd_soc_codec *codec) +{ + cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) +static int beep_rates[] = { + 261, 522, 585, 667, 706, 774, 889, 1000, + 1043, 1200, 1333, 1412, 1600, 1714, 2000, 2182 +}; + +static void cs42l52_beep_work(struct work_struct *work) +{ + struct cs42l52_private *cs42l52 = + container_of(work, struct cs42l52_private, beep_work); + struct snd_soc_codec *codec = cs42l52->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int i; + int val = 0; + int best = 0; + + if (cs42l52->beep_rate) { + for (i = 0; i < ARRAY_SIZE(beep_rates); i++) { + if (abs(cs42l52->beep_rate - beep_rates[i]) < + abs(cs42l52->beep_rate - beep_rates[best])) + best = i; + } + + dev_dbg(codec->dev, "Set beep rate %dHz for requested %dHz\n", + beep_rates[best], cs42l52->beep_rate); + + val = (best << CS42L52_BEEP_RATE_SHIFT); + + snd_soc_dapm_enable_pin(dapm, "Beep"); + } else { + dev_dbg(codec->dev, "Disabling beep\n"); + snd_soc_dapm_disable_pin(dapm, "Beep"); + } + + snd_soc_update_bits(codec, CS42L52_BEEP_FREQ, + CS42L52_BEEP_RATE_MASK, val); + + snd_soc_dapm_sync(dapm); +} + +/* For usability define a way of injecting beep events for the device - + * many systems will not have a keyboard. + */ +static int cs42l52_beep_event(struct input_dev *dev, unsigned int type, + unsigned int code, int hz) +{ + struct snd_soc_codec *codec = input_get_drvdata(dev); + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "Beep event %x %x\n", code, hz); + + switch (code) { + case SND_BELL: + if (hz) + hz = 261; + case SND_TONE: + break; + default: + return -1; + } + + /* Kick the beep from a workqueue */ + cs42l52->beep_rate = hz; + schedule_work(&cs42l52->beep_work); + return 0; +} + +static ssize_t cs42l52_beep_set(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct cs42l52_private *cs42l52 = dev_get_drvdata(dev); + long int time; + int ret; + + ret = kstrtol(buf, 10, &time); + if (ret != 0) + return ret; + + input_event(cs42l52->beep, EV_SND, SND_TONE, time); + + return count; +} + +static DEVICE_ATTR(beep, 0200, NULL, cs42l52_beep_set); + +static void cs42l52_init_beep(struct snd_soc_codec *codec) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + int ret; + + cs42l52->beep = input_allocate_device(); + if (!cs42l52->beep) { + dev_err(codec->dev, "Failed to allocate beep device\n"); + return; + } + + INIT_WORK(&cs42l52->beep_work, cs42l52_beep_work); + cs42l52->beep_rate = 0; + + cs42l52->beep->name = "CS42L52 Beep Generator"; + cs42l52->beep->phys = dev_name(codec->dev); + cs42l52->beep->id.bustype = BUS_I2C; + + cs42l52->beep->evbit[0] = BIT_MASK(EV_SND); + cs42l52->beep->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE); + cs42l52->beep->event = cs42l52_beep_event; + cs42l52->beep->dev.parent = codec->dev; + input_set_drvdata(cs42l52->beep, codec); + + ret = input_register_device(cs42l52->beep); + if (ret != 0) { + input_free_device(cs42l52->beep); + cs42l52->beep = NULL; + dev_err(codec->dev, "Failed to register beep device\n"); + } + + ret = device_create_file(codec->dev, &dev_attr_beep); + if (ret != 0) { + dev_err(codec->dev, "Failed to create keyclick file: %d\n", + ret); + } +} + +static void cs42l52_free_beep(struct snd_soc_codec *codec) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + + device_remove_file(codec->dev, &dev_attr_beep); + input_unregister_device(cs42l52->beep); + cancel_work_sync(&cs42l52->beep_work); + cs42l52->beep = NULL; + + snd_soc_update_bits(codec, CS42L52_BEEP_TONE_CTL, + CS42L52_BEEP_EN_MASK, 0); +} +#else +static void cs42l52_init_beep(struct snd_soc_codec *codec) +{ +} + +static void cs42l52_free_beep(struct snd_soc_codec *codec) +{ +} +#endif + +static int cs42l52_probe(struct snd_soc_codec *codec) +{ + struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = cs42l52->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + regcache_cache_only(cs42l52->regmap, true); + + cs42l52_init_beep(codec); + + cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + cs42l52->sysclk = CS42L52_DEFAULT_CLK; + cs42l52->config.format = CS42L52_DEFAULT_FORMAT; + + /* Set Platform MICx CFG */ + snd_soc_update_bits(codec, CS42L52_MICA_CTL, + CS42L52_MIC_CTL_TYPE_MASK, + cs42l52->pdata.mica_cfg << + CS42L52_MIC_CTL_TYPE_SHIFT); + + snd_soc_update_bits(codec, CS42L52_MICB_CTL, + CS42L52_MIC_CTL_TYPE_MASK, + cs42l52->pdata.micb_cfg << + CS42L52_MIC_CTL_TYPE_SHIFT); + + /* if Single Ended, Get Mic_Select */ + if (cs42l52->pdata.mica_cfg) + snd_soc_update_bits(codec, CS42L52_MICA_CTL, + CS42L52_MIC_CTL_MIC_SEL_MASK, + cs42l52->pdata.mica_sel << + CS42L52_MIC_CTL_MIC_SEL_SHIFT); + if (cs42l52->pdata.micb_cfg) + snd_soc_update_bits(codec, CS42L52_MICB_CTL, + CS42L52_MIC_CTL_MIC_SEL_MASK, + cs42l52->pdata.micb_sel << + CS42L52_MIC_CTL_MIC_SEL_SHIFT); + + /* Set Platform Charge Pump Freq */ + snd_soc_update_bits(codec, CS42L52_CHARGE_PUMP, + CS42L52_CHARGE_PUMP_MASK, + cs42l52->pdata.chgfreq << + CS42L52_CHARGE_PUMP_SHIFT); + + /* Set Platform Bias Level */ + snd_soc_update_bits(codec, CS42L52_IFACE_CTL2, + CS42L52_IFACE_CTL2_BIAS_LVL, + cs42l52->pdata.micbias_lvl); + + return ret; +} + +static int cs42l52_remove(struct snd_soc_codec *codec) +{ + cs42l52_free_beep(codec); + cs42l52_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs42l52 = { + .probe = cs42l52_probe, + .remove = cs42l52_remove, + .suspend = cs42l52_suspend, + .resume = cs42l52_resume, + .set_bias_level = cs42l52_set_bias_level, + + .dapm_widgets = cs42l52_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42l52_dapm_widgets), + .dapm_routes = cs42l52_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs42l52_audio_map), + + .controls = cs42l52_snd_controls, + .num_controls = ARRAY_SIZE(cs42l52_snd_controls), +}; + +/* Current and threshold powerup sequence Pg37 */ +static const struct reg_default cs42l52_threshold_patch[] = { + + { 0x00, 0x99 }, + { 0x3E, 0xBA }, + { 0x47, 0x80 }, + { 0x32, 0xBB }, + { 0x32, 0x3B }, + { 0x00, 0x00 }, + +}; + +static struct regmap_config cs42l52_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42L52_MAX_REGISTER, + .reg_defaults = cs42l52_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs42l52_reg_defaults), + .readable_reg = cs42l52_readable_register, + .volatile_reg = cs42l52_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs42l52_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs42l52_private *cs42l52; + int ret; + unsigned int devid = 0; + unsigned int reg; + + cs42l52 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l52_private), + GFP_KERNEL); + if (cs42l52 == NULL) + return -ENOMEM; + cs42l52->dev = &i2c_client->dev; + + cs42l52->regmap = devm_regmap_init_i2c(i2c_client, &cs42l52_regmap); + if (IS_ERR(cs42l52->regmap)) { + ret = PTR_ERR(cs42l52->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + i2c_set_clientdata(i2c_client, cs42l52); + + if (dev_get_platdata(&i2c_client->dev)) + memcpy(&cs42l52->pdata, dev_get_platdata(&i2c_client->dev), + sizeof(cs42l52->pdata)); + + ret = regmap_register_patch(cs42l52->regmap, cs42l52_threshold_patch, + ARRAY_SIZE(cs42l52_threshold_patch)); + if (ret != 0) + dev_warn(cs42l52->dev, "Failed to apply regmap patch: %d\n", + ret); + + ret = regmap_read(cs42l52->regmap, CS42L52_CHIP, ®); + devid = reg & CS42L52_CHIP_ID_MASK; + if (devid != CS42L52_CHIP_ID) { + ret = -ENODEV; + dev_err(&i2c_client->dev, + "CS42L52 Device ID (%X). Expected %X\n", + devid, CS42L52_CHIP_ID); + return ret; + } + + regcache_cache_only(cs42l52->regmap, true); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs42l52, &cs42l52_dai, 1); + if (ret < 0) + return ret; + return 0; +} + +static int cs42l52_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id cs42l52_id[] = { + { "cs42l52", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, cs42l52_id); + +static struct i2c_driver cs42l52_i2c_driver = { + .driver = { + .name = "cs42l52", + .owner = THIS_MODULE, + }, + .id_table = cs42l52_id, + .probe = cs42l52_i2c_probe, + .remove = __devexit_p(cs42l52_i2c_remove), +}; + +module_i2c_driver(cs42l52_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS42L52 driver"); +MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe@nucleusys.com>"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin@cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h new file mode 100644 index 000000000000..60985c059071 --- /dev/null +++ b/sound/soc/codecs/cs42l52.h @@ -0,0 +1,274 @@ +/* + * cs42l52.h -- CS42L52 ALSA SoC audio driver + * + * Copyright 2012 CirrusLogic, Inc. + * + * Author: Georgi Vlaev <joe@nucleusys.com> + * Author: Brian Austin <brian.austin@cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef __CS42L52_H__ +#define __CS42L52_H__ + +#define CS42L52_NAME "CS42L52" +#define CS42L52_DEFAULT_CLK 12000000 +#define CS42L52_MIN_CLK 11000000 +#define CS42L52_MAX_CLK 27000000 +#define CS42L52_DEFAULT_FORMAT SNDRV_PCM_FMTBIT_S16_LE +#define CS42L52_DEFAULT_MAX_CHANS 2 +#define CS42L52_SYSCLK 1 + +#define CS42L52_CHIP_SWICTH (1 << 17) +#define CS42L52_ALL_IN_ONE (1 << 16) +#define CS42L52_CHIP_ONE 0x00 +#define CS42L52_CHIP_TWO 0x01 +#define CS42L52_CHIP_THR 0x02 +#define CS42L52_CHIP_MASK 0x0f + +#define CS42L52_FIX_BITS_CTL 0x00 +#define CS42L52_CHIP 0x01 +#define CS42L52_CHIP_ID 0xE0 +#define CS42L52_CHIP_ID_MASK 0xF8 +#define CS42L52_CHIP_REV_A0 0x00 +#define CS42L52_CHIP_REV_A1 0x01 +#define CS42L52_CHIP_REV_B0 0x02 +#define CS42L52_CHIP_REV_MASK 0x03 + +#define CS42L52_PWRCTL1 0x02 +#define CS42L52_PWRCTL1_PDN_ALL 0x9F +#define CS42L52_PWRCTL1_PDN_CHRG 0x80 +#define CS42L52_PWRCTL1_PDN_PGAB 0x10 +#define CS42L52_PWRCTL1_PDN_PGAA 0x08 +#define CS42L52_PWRCTL1_PDN_ADCB 0x04 +#define CS42L52_PWRCTL1_PDN_ADCA 0x02 +#define CS42L52_PWRCTL1_PDN_CODEC 0x01 + +#define CS42L52_PWRCTL2 0x03 +#define CS42L52_PWRCTL2_OVRDB (1 << 4) +#define CS42L52_PWRCTL2_OVRDA (1 << 3) +#define CS42L52_PWRCTL2_PDN_MICB (1 << 2) +#define CS42L52_PWRCTL2_PDN_MICB_SHIFT 2 +#define CS42L52_PWRCTL2_PDN_MICA (1 << 1) +#define CS42L52_PWRCTL2_PDN_MICA_SHIFT 1 +#define CS42L52_PWRCTL2_PDN_MICBIAS (1 << 0) +#define CS42L52_PWRCTL2_PDN_MICBIAS_SHIFT 0 + +#define CS42L52_PWRCTL3 0x04 +#define CS42L52_PWRCTL3_HPB_PDN_SHIFT 6 +#define CS42L52_PWRCTL3_HPB_ON_LOW 0x00 +#define CS42L52_PWRCTL3_HPB_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_HPB_ALWAYS_ON 0x02 +#define CS42L52_PWRCTL3_HPB_ALWAYS_OFF 0x03 +#define CS42L52_PWRCTL3_HPA_PDN_SHIFT 4 +#define CS42L52_PWRCTL3_HPA_ON_LOW 0x00 +#define CS42L52_PWRCTL3_HPA_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_HPA_ALWAYS_ON 0x02 +#define CS42L52_PWRCTL3_HPA_ALWAYS_OFF 0x03 +#define CS42L52_PWRCTL3_SPKB_PDN_SHIFT 2 +#define CS42L52_PWRCTL3_SPKB_ON_LOW 0x00 +#define CS42L52_PWRCTL3_SPKB_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_SPKB_ALWAYS_ON 0x02 +#define CS42L52_PWRCTL3_PDN_SPKB (1 << 2) +#define CS42L52_PWRCTL3_PDN_SPKA (1 << 0) +#define CS42L52_PWRCTL3_SPKA_PDN_SHIFT 0 +#define CS42L52_PWRCTL3_SPKA_ON_LOW 0x00 +#define CS42L52_PWRCTL3_SPKA_ON_HIGH 0x01 +#define CS42L52_PWRCTL3_SPKA_ALWAYS_ON 0x02 + +#define CS42L52_DEFAULT_OUTPUT_STATE 0x05 +#define CS42L52_PWRCTL3_CONF_MASK 0x03 + +#define CS42L52_CLK_CTL 0x05 +#define CLK_AUTODECT_ENABLE (1 << 7) +#define CLK_SPEED_SHIFT 5 +#define CLK_DS_MODE 0x00 +#define CLK_SS_MODE 0x01 +#define CLK_HS_MODE 0x02 +#define CLK_QS_MODE 0x03 +#define CLK_32K_SR_SHIFT 4 +#define CLK_32K 0x01 +#define CLK_NO_32K 0x00 +#define CLK_27M_MCLK_SHIFT 3 +#define CLK_27M_MCLK 0x01 +#define CLK_NO_27M 0x00 +#define CLK_RATIO_SHIFT 1 +#define CLK_R_128 0x00 +#define CLK_R_125 0x01 +#define CLK_R_132 0x02 +#define CLK_R_136 0x03 + +#define CS42L52_IFACE_CTL1 0x06 +#define CS42L52_IFACE_CTL1_MASTER (1 << 7) +#define CS42L52_IFACE_CTL1_SLAVE (0 << 7) +#define CS42L52_IFACE_CTL1_INV_SCLK (1 << 6) +#define CS42L52_IFACE_CTL1_ADC_FMT_I2S (1 << 5) +#define CS42L52_IFACE_CTL1_ADC_FMT_LEFT_J (0 << 5) +#define CS42L52_IFACE_CTL1_DSP_MODE_EN (1 << 4) +#define CS42L52_IFACE_CTL1_DAC_FMT_LEFT_J (0 << 2) +#define CS42L52_IFACE_CTL1_DAC_FMT_I2S (1 << 2) +#define CS42L52_IFACE_CTL1_DAC_FMT_RIGHT_J (2 << 2) +#define CS42L52_IFACE_CTL1_WL_32BIT (0x00) +#define CS42L52_IFACE_CTL1_WL_24BIT (0x01) +#define CS42L52_IFACE_CTL1_WL_20BIT (0x02) +#define CS42L52_IFACE_CTL1_WL_16BIT (0x03) +#define CS42L52_IFACE_CTL1_WL_MASK 0xFFFF + +#define CS42L52_IFACE_CTL2 0x07 +#define CS42L52_IFACE_CTL2_SC_MC_EQ (1 << 6) +#define CS42L52_IFACE_CTL2_LOOPBACK (1 << 5) +#define CS42L52_IFACE_CTL2_S_MODE_OUTPUT_EN (0 << 4) +#define CS42L52_IFACE_CTL2_S_MODE_OUTPUT_HIZ (1 << 4) +#define CS42L52_IFACE_CTL2_HP_SW_INV (1 << 3) +#define CS42L52_IFACE_CTL2_BIAS_LVL 0x07 + +#define CS42L52_ADC_PGA_A 0x08 +#define CS42L52_ADC_PGA_B 0x09 +#define CS42L52_ADC_SEL_SHIFT 5 +#define CS42L52_ADC_SEL_AIN1 0x00 +#define CS42L52_ADC_SEL_AIN2 0x01 +#define CS42L52_ADC_SEL_AIN3 0x02 +#define CS42L52_ADC_SEL_AIN4 0x03 +#define CS42L52_ADC_SEL_PGA 0x04 + +#define CS42L52_ANALOG_HPF_CTL 0x0A +#define CS42L52_HPF_CTL_ANLGSFTB (1 << 3) +#define CS42L52_HPF_CTL_ANLGSFTA (1 << 0) + +#define CS42L52_ADC_HPF_FREQ 0x0B +#define CS42L52_ADC_MISC_CTL 0x0C +#define CS42L52_ADC_MISC_CTL_SOURCE_DSP (1 << 6) + +#define CS42L52_PB_CTL1 0x0D +#define CS42L52_PB_CTL1_HP_GAIN_SHIFT 5 +#define CS42L52_PB_CTL1_HP_GAIN_03959 0x00 +#define CS42L52_PB_CTL1_HP_GAIN_04571 0x01 +#define CS42L52_PB_CTL1_HP_GAIN_05111 0x02 +#define CS42L52_PB_CTL1_HP_GAIN_06047 0x03 +#define CS42L52_PB_CTL1_HP_GAIN_07099 0x04 +#define CS42L52_PB_CTL1_HP_GAIN_08399 0x05 +#define CS42L52_PB_CTL1_HP_GAIN_10000 0x06 +#define CS42L52_PB_CTL1_HP_GAIN_11430 0x07 +#define CS42L52_PB_CTL1_INV_PCMB (1 << 3) +#define CS42L52_PB_CTL1_INV_PCMA (1 << 2) +#define CS42L52_PB_CTL1_MSTB_MUTE (1 << 1) +#define CS42L52_PB_CTL1_MSTA_MUTE (1 << 0) +#define CS42L52_PB_CTL1_MUTE_MASK 0xFFFD +#define CS42L52_PB_CTL1_MUTE 3 +#define CS42L52_PB_CTL1_UNMUTE 0 + +#define CS42L52_MISC_CTL 0x0E +#define CS42L52_MISC_CTL_DEEMPH (1 << 2) +#define CS42L52_MISC_CTL_DIGSFT (1 << 1) +#define CS42L52_MISC_CTL_DIGZC (1 << 0) + +#define CS42L52_PB_CTL2 0x0F +#define CS42L52_PB_CTL2_HPB_MUTE (1 << 7) +#define CS42L52_PB_CTL2_HPA_MUTE (1 << 6) +#define CS42L52_PB_CTL2_SPKB_MUTE (1 << 5) +#define CS42L52_PB_CTL2_SPKA_MUTE (1 << 4) +#define CS42L52_PB_CTL2_SPK_SWAP (1 << 2) +#define CS42L52_PB_CTL2_SPK_MONO (1 << 1) +#define CS42L52_PB_CTL2_SPK_MUTE50 (1 << 0) + +#define CS42L52_MICA_CTL 0x10 +#define CS42L52_MICB_CTL 0x11 +#define CS42L52_MIC_CTL_MIC_SEL_MASK 0xBF +#define CS42L52_MIC_CTL_MIC_SEL_SHIFT 6 +#define CS42L52_MIC_CTL_TYPE_MASK 0xDF +#define CS42L52_MIC_CTL_TYPE_SHIFT 5 + + +#define CS42L52_PGAA_CTL 0x12 +#define CS42L52_PGAB_CTL 0x13 +#define CS42L52_PGAX_CTL_VOL_12DB 24 +#define CS42L52_PGAX_CTL_VOL_6DB 12 /*step size 0.5db*/ + +#define CS42L52_PASSTHRUA_VOL 0x14 +#define CS42L52_PASSTHRUB_VOL 0x15 + +#define CS42L52_ADCA_VOL 0x16 +#define CS42L52_ADCB_VOL 0x17 +#define CS42L52_ADCX_VOL_24DB 24 /*step size 1db*/ +#define CS42L52_ADCX_VOL_12DB 12 +#define CS42L52_ADCX_VOL_6DB 6 + +#define CS42L52_ADCA_MIXER_VOL 0x18 +#define CS42L52_ADCB_MIXER_VOL 0x19 +#define CS42L52_ADC_MIXER_VOL_12DB 0x18 + +#define CS42L52_PCMA_MIXER_VOL 0x1A +#define CS42L52_PCMB_MIXER_VOL 0x1B + +#define CS42L52_BEEP_FREQ 0x1C +#define CS42L52_BEEP_VOL 0x1D +#define CS42L52_BEEP_TONE_CTL 0x1E +#define CS42L52_BEEP_RATE_SHIFT 4 +#define CS42L52_BEEP_RATE_MASK 0x0F + +#define CS42L52_TONE_CTL 0x1F +#define CS42L52_BEEP_EN_MASK 0x3F + +#define CS42L52_MASTERA_VOL 0x20 +#define CS42L52_MASTERB_VOL 0x21 + +#define CS42L52_HPA_VOL 0x22 +#define CS42L52_HPB_VOL 0x23 +#define CS42L52_DEFAULT_HP_VOL 0xF0 + +#define CS42L52_SPKA_VOL 0x24 +#define CS42L52_SPKB_VOL 0x25 +#define CS42L52_DEFAULT_SPK_VOL 0xF0 + +#define CS42L52_ADC_PCM_MIXER 0x26 + +#define CS42L52_LIMITER_CTL1 0x27 +#define CS42L52_LIMITER_CTL2 0x28 +#define CS42L52_LIMITER_AT_RATE 0x29 + +#define CS42L52_ALC_CTL 0x2A +#define CS42L52_ALC_CTL_ALCB_ENABLE_SHIFT 7 +#define CS42L52_ALC_CTL_ALCA_ENABLE_SHIFT 6 +#define CS42L52_ALC_CTL_FASTEST_ATTACK 0 + +#define CS42L52_ALC_RATE 0x2B +#define CS42L52_ALC_SLOWEST_RELEASE 0x3F + +#define CS42L52_ALC_THRESHOLD 0x2C +#define CS42L52_ALC_MAX_RATE_SHIFT 5 +#define CS42L52_ALC_MIN_RATE_SHIFT 2 +#define CS42L52_ALC_RATE_0DB 0 +#define CS42L52_ALC_RATE_3DB 1 +#define CS42L52_ALC_RATE_6DB 2 + +#define CS42L52_NOISE_GATE_CTL 0x2D +#define CS42L52_NG_ENABLE_SHIFT 6 +#define CS42L52_NG_THRESHOLD_SHIFT 2 +#define CS42L52_NG_MIN_70DB 2 +#define CS42L52_NG_DELAY_SHIFT 0 +#define CS42L52_NG_DELAY_100MS 1 + +#define CS42L52_CLK_STATUS 0x2E +#define CS42L52_BATT_COMPEN 0x2F + +#define CS42L52_BATT_LEVEL 0x30 +#define CS42L52_SPK_STATUS 0x31 +#define CS42L52_SPK_STATUS_PIN_SHIFT 3 +#define CS42L52_SPK_STATUS_PIN_HIGH 1 + +#define CS42L52_TEM_CTL 0x32 +#define CS42L52_TEM_CTL_SET 0x80 +#define CS42L52_THE_FOLDBACK 0x33 +#define CS42L52_CHARGE_PUMP 0x34 +#define CS42L52_CHARGE_PUMP_MASK 0xF0 +#define CS42L52_CHARGE_PUMP_SHIFT 4 +#define CS42L52_FIX_BITS1 0x3E +#define CS42L52_FIX_BITS2 0x47 + +#define CS42L52_MAX_REGISTER 0x34 + +#endif diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 07c44b71f096..2c08c4cb465a 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -43,9 +43,6 @@ struct cs42l73_private { }; static const struct reg_default cs42l73_reg_defaults[] = { - { 1, 0x42 }, /* r01 - Device ID A&B */ - { 2, 0xA7 }, /* r02 - Device ID C&D */ - { 3, 0x30 }, /* r03 - Device ID E */ { 6, 0xF1 }, /* r06 - Power Ctl 1 */ { 7, 0xDF }, /* r07 - Power Ctl 2 */ { 8, 0x3F }, /* r08 - Power Ctl 3 */ @@ -402,37 +399,37 @@ static const struct snd_kcontrol_new ear_amp_ctl = static const struct snd_kcontrol_new cs42l73_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume", - CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7, - 0xffffffC1, 0x0C, hpaloa_tlv), + CS42L73_HPAAVOL, CS42L73_HPBAVOL, 0, + 0x41, 0x4B, hpaloa_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, - CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv), + CS42L73_LOBAVOL, 0, 0x41, 0x4B, hpaloa_tlv), SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, - CS42L73_MICBPREPGABVOL, 5, 0xffffff35, - 0x34, micpga_tlv), + CS42L73_MICBPREPGABVOL, 5, 0x34, + 0x24, micpga_tlv), SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, CS42L73_MICBPREPGABVOL, 6, 1, 1), SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL, - CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv), + CS42L73_IPBDVOL, 0, 0xA0, 0x6C, ipd_tlv), SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume", - CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5, - 0xE4, hl_tlv), + CS42L73_HLADVOL, CS42L73_HLBDVOL, + 0, 0x34, 0xE4, hl_tlv), SOC_SINGLE_TLV("ADC A Boost Volume", CS42L73_ADCIPC, 2, 0x01, 1, adc_boost_tlv), SOC_SINGLE_TLV("ADC B Boost Volume", - CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), + CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), - SOC_SINGLE_TLV("Speakerphone Digital Playback Volume", - CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv), + SOC_SINGLE_SX_TLV("Speakerphone Digital Volume", + CS42L73_SPKDVOL, 0, 0x34, 0xE4, hl_tlv), - SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume", - CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv), + SOC_SINGLE_SX_TLV("Ear Speaker Digital Volume", + CS42L73_ESLDVOL, 0, 0x34, 0xE4, hl_tlv), SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7, 1, 1), @@ -568,22 +565,22 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = { attn_tlv), SOC_SINGLE_TLV("SPK-IP Mono Volume", - CS42L73_SPKMIPMA, 0, 0x3E, 1, attn_tlv), + CS42L73_SPKMIPMA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("SPK-XSP Mono Volume", - CS42L73_SPKMXSPA, 0, 0x3E, 1, attn_tlv), + CS42L73_SPKMXSPA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("SPK-ASP Mono Volume", - CS42L73_SPKMASPA, 0, 0x3E, 1, attn_tlv), + CS42L73_SPKMASPA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("SPK-VSP Mono Volume", - CS42L73_SPKMVSPMA, 0, 0x3E, 1, attn_tlv), + CS42L73_SPKMVSPMA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("ESL-IP Mono Volume", - CS42L73_ESLMIPMA, 0, 0x3E, 1, attn_tlv), + CS42L73_ESLMIPMA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("ESL-XSP Mono Volume", - CS42L73_ESLMXSPA, 0, 0x3E, 1, attn_tlv), + CS42L73_ESLMXSPA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("ESL-ASP Mono Volume", - CS42L73_ESLMASPA, 0, 0x3E, 1, attn_tlv), + CS42L73_ESLMASPA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("ESL-VSP Mono Volume", - CS42L73_ESLMVSPMA, 0, 0x3E, 1, attn_tlv), + CS42L73_ESLMVSPMA, 0, 0x3F, 1, attn_tlv), SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum), @@ -599,17 +596,17 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS42L73_PWRCTL2, 7, 1, NULL, 0), - SND_SOC_DAPM_AIF_OUT("XSPOUTL", "XSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("XSPOUTL", NULL, 0, CS42L73_PWRCTL2, 1, 1), - SND_SOC_DAPM_AIF_OUT("XSPOUTR", "XSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("XSPOUTR", NULL, 0, CS42L73_PWRCTL2, 1, 1), - SND_SOC_DAPM_AIF_OUT("ASPOUTL", "ASP Capture", 0, + SND_SOC_DAPM_AIF_OUT("ASPOUTL", NULL, 0, CS42L73_PWRCTL2, 3, 1), - SND_SOC_DAPM_AIF_OUT("ASPOUTR", "ASP Capture", 0, + SND_SOC_DAPM_AIF_OUT("ASPOUTR", NULL, 0, CS42L73_PWRCTL2, 3, 1), - SND_SOC_DAPM_AIF_OUT("VSPOUTL", "VSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("VSPOUTL", NULL, 0, CS42L73_PWRCTL2, 4, 1), - SND_SOC_DAPM_AIF_OUT("VSPOUTR", "VSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("VSPOUTR", NULL, 0, CS42L73_PWRCTL2, 4, 1), SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -638,21 +635,21 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { SND_SOC_DAPM_MIXER("VSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("VSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_AIF_IN("XSPINL", "XSP Playback", 0, + SND_SOC_DAPM_AIF_IN("XSPINL", NULL, 0, CS42L73_PWRCTL2, 0, 1), - SND_SOC_DAPM_AIF_IN("XSPINR", "XSP Playback", 0, + SND_SOC_DAPM_AIF_IN("XSPINR", NULL, 0, CS42L73_PWRCTL2, 0, 1), - SND_SOC_DAPM_AIF_IN("XSPINM", "XSP Playback", 0, + SND_SOC_DAPM_AIF_IN("XSPINM", NULL, 0, CS42L73_PWRCTL2, 0, 1), - SND_SOC_DAPM_AIF_IN("ASPINL", "ASP Playback", 0, + SND_SOC_DAPM_AIF_IN("ASPINL", NULL, 0, CS42L73_PWRCTL2, 2, 1), - SND_SOC_DAPM_AIF_IN("ASPINR", "ASP Playback", 0, + SND_SOC_DAPM_AIF_IN("ASPINR", NULL, 0, CS42L73_PWRCTL2, 2, 1), - SND_SOC_DAPM_AIF_IN("ASPINM", "ASP Playback", 0, + SND_SOC_DAPM_AIF_IN("ASPINM", NULL, 0, CS42L73_PWRCTL2, 2, 1), - SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0, + SND_SOC_DAPM_AIF_IN("VSPIN", NULL, 0, CS42L73_PWRCTL2, 4, 1), SND_SOC_DAPM_MIXER("HL Left Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -776,6 +773,14 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = { {"HL Left Mixer", NULL, "VSPIN"}, {"HL Right Mixer", NULL, "VSPIN"}, + {"ASPINL", NULL, "ASP Playback"}, + {"ASPINM", NULL, "ASP Playback"}, + {"ASPINR", NULL, "ASP Playback"}, + {"XSPINL", NULL, "XSP Playback"}, + {"XSPINM", NULL, "XSP Playback"}, + {"XSPINR", NULL, "XSP Playback"}, + {"VSPIN", NULL, "VSP Playback"}, + /* Capture Paths */ {"MIC1", NULL, "MIC1 Bias"}, {"PGA Left Mux", "Mic 1", "MIC1"}, @@ -822,6 +827,13 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = { {"VSPOUTL", NULL, "VSPL Output Mixer"}, {"VSPOUTR", NULL, "VSPR Output Mixer"}, + + {"ASP Capture", NULL, "ASPOUTL"}, + {"ASP Capture", NULL, "ASPOUTR"}, + {"XSP Capture", NULL, "XSPOUTL"}, + {"XSP Capture", NULL, "XSPOUTR"}, + {"VSP Capture", NULL, "VSPOUTL"}, + {"VSP Capture", NULL, "VSPOUTR"}, }; struct cs42l73_mclk_div { @@ -1091,8 +1103,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); int id = dai->id; int mclk_coeff; @@ -1351,11 +1362,11 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, i2c_set_clientdata(i2c_client, cs42l73); - cs42l73->regmap = regmap_init_i2c(i2c_client, &cs42l73_regmap); + cs42l73->regmap = devm_regmap_init_i2c(i2c_client, &cs42l73_regmap); if (IS_ERR(cs42l73->regmap)) { ret = PTR_ERR(cs42l73->regmap); dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); - goto err; + return ret; } /* initialize codec */ ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_AB, ®); @@ -1373,13 +1384,13 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "CS42L73 Device ID (%X). Expected %X\n", devid, CS42L73_DEVID); - goto err_regmap; + return ret; } ret = regmap_read(cs42l73->regmap, CS42L73_REVID, ®); if (ret < 0) { dev_err(&i2c_client->dev, "Get Revision ID failed\n"); - goto err_regmap; + return ret;; } dev_info(&i2c_client->dev, @@ -1391,23 +1402,13 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, &soc_codec_dev_cs42l73, cs42l73_dai, ARRAY_SIZE(cs42l73_dai)); if (ret < 0) - goto err_regmap; + return ret; return 0; - -err_regmap: - regmap_exit(cs42l73->regmap); - -err: - return ret; } static __devexit int cs42l73_i2c_remove(struct i2c_client *client) { - struct cs42l73_private *cs42l73 = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - regmap_exit(cs42l73->regmap); - return 0; } @@ -1429,25 +1430,7 @@ static struct i2c_driver cs42l73_i2c_driver = { }; -static int __init cs42l73_modinit(void) -{ - int ret; - ret = i2c_add_driver(&cs42l73_i2c_driver); - if (ret != 0) { - pr_err("Failed to register CS42L73 I2C driver: %d\n", ret); - return ret; - } - return 0; -} - -module_init(cs42l73_modinit); - -static void __exit cs42l73_exit(void) -{ - i2c_del_driver(&cs42l73_i2c_driver); -} - -module_exit(cs42l73_exit); +module_i2c_driver(cs42l73_i2c_driver); MODULE_DESCRIPTION("ASoC CS42L73 driver"); MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe@nucleusys.com>"); diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 7843711729bc..af5db7080519 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -17,6 +17,7 @@ #include <linux/delay.h> #include <linux/i2c.h> +#include <linux/spi/spi.h> #include <linux/regmap.h> #include <linux/slab.h> #include <linux/module.h> @@ -27,6 +28,7 @@ #include <sound/tlv.h> /* DA7210 register space */ +#define DA7210_PAGE_CONTROL 0x00 #define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 @@ -146,6 +148,7 @@ #define DA7210_DAI_EN (1 << 7) /*PLL_DIV3 bit fields */ +#define DA7210_PLL_DIV_L_MASK (0xF << 0) #define DA7210_MCLK_RANGE_10_20_MHZ (1 << 4) #define DA7210_PLL_BYP (1 << 6) @@ -162,12 +165,16 @@ #define DA7210_PLL_FS_48000 (0xB << 0) #define DA7210_PLL_FS_88200 (0xE << 0) #define DA7210_PLL_FS_96000 (0xF << 0) +#define DA7210_MCLK_DET_EN (0x1 << 5) +#define DA7210_MCLK_SRM_EN (0x1 << 6) #define DA7210_PLL_EN (0x1 << 7) /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) /* CONTROL bit fields */ +#define DA7210_REG_EN (1 << 0) +#define DA7210_BIAS_EN (1 << 2) #define DA7210_NOISE_SUP_EN (1 << 3) /* IN_GAIN bit fields */ @@ -206,6 +213,47 @@ #define DA7210_OUT2_OUTMIX_L (1 << 6) #define DA7210_OUT2_EN (1 << 7) +struct pll_div { + int fref; + int fout; + u8 div1; + u8 div2; + u8 div3; + u8 mode; /* 0 = slave, 1 = master */ +}; + +/* PLL dividers table */ +static const struct pll_div da7210_pll_div[] = { + /* for MASTER mode, fs = 44.1Khz */ + { 12000000, 2822400, 0xE8, 0x6C, 0x2, 1}, /* MCLK=12Mhz */ + { 13000000, 2822400, 0xDF, 0x28, 0xC, 1}, /* MCLK=13Mhz */ + { 13500000, 2822400, 0xDB, 0x0A, 0xD, 1}, /* MCLK=13.5Mhz */ + { 14400000, 2822400, 0xD4, 0x5A, 0x2, 1}, /* MCLK=14.4Mhz */ + { 19200000, 2822400, 0xBB, 0x43, 0x9, 1}, /* MCLK=19.2Mhz */ + { 19680000, 2822400, 0xB9, 0x6D, 0xA, 1}, /* MCLK=19.68Mhz */ + { 19800000, 2822400, 0xB8, 0xFB, 0xB, 1}, /* MCLK=19.8Mhz */ + /* for MASTER mode, fs = 48Khz */ + { 12000000, 3072000, 0xF3, 0x12, 0x7, 1}, /* MCLK=12Mhz */ + { 13000000, 3072000, 0xE8, 0xFD, 0x5, 1}, /* MCLK=13Mhz */ + { 13500000, 3072000, 0xE4, 0x82, 0x3, 1}, /* MCLK=13.5Mhz */ + { 14400000, 3072000, 0xDD, 0x3A, 0x0, 1}, /* MCLK=14.4Mhz */ + { 19200000, 3072000, 0xC1, 0xEB, 0x8, 1}, /* MCLK=19.2Mhz */ + { 19680000, 3072000, 0xBF, 0xEC, 0x0, 1}, /* MCLK=19.68Mhz */ + { 19800000, 3072000, 0xBF, 0x70, 0x0, 1}, /* MCLK=19.8Mhz */ + /* for SLAVE mode with SRM */ + { 12000000, 2822400, 0xED, 0xBF, 0x5, 0}, /* MCLK=12Mhz */ + { 13000000, 2822400, 0xE4, 0x13, 0x0, 0}, /* MCLK=13Mhz */ + { 13500000, 2822400, 0xDF, 0xC6, 0x8, 0}, /* MCLK=13.5Mhz */ + { 14400000, 2822400, 0xD8, 0xCA, 0x1, 0}, /* MCLK=14.4Mhz */ + { 19200000, 2822400, 0xBE, 0x97, 0x9, 0}, /* MCLK=19.2Mhz */ + { 19680000, 2822400, 0xBC, 0xAC, 0xD, 0}, /* MCLK=19.68Mhz */ + { 19800000, 2822400, 0xBC, 0x35, 0xE, 0}, /* MCLK=19.8Mhz */ +}; + +enum clk_src { + DA7210_CLKSRC_MCLK +}; + #define DA7210_VERSION "0.0.1" /* @@ -628,9 +676,12 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Codec private data */ struct da7210_priv { struct regmap *regmap; + unsigned int mclk_rate; + int master; }; static struct reg_default da7210_reg_defaults[] = { + { 0x00, 0x00 }, { 0x01, 0x11 }, { 0x03, 0x00 }, { 0x04, 0x00 }, @@ -713,10 +764,10 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); u32 dai_cfg1; - u32 fs, bypass; + u32 fs, sysclk; /* set DAI source to Left and Right ADC */ snd_soc_write(codec, DA7210_DAI_SRC_SEL, @@ -749,43 +800,43 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 8000: fs = DA7210_PLL_FS_8000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 11025: fs = DA7210_PLL_FS_11025; - bypass = 0; + sysclk = 2822400; break; case 12000: fs = DA7210_PLL_FS_12000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 16000: fs = DA7210_PLL_FS_16000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 22050: fs = DA7210_PLL_FS_22050; - bypass = 0; + sysclk = 2822400; break; case 32000: fs = DA7210_PLL_FS_32000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 44100: fs = DA7210_PLL_FS_44100; - bypass = 0; + sysclk = 2822400; break; case 48000: fs = DA7210_PLL_FS_48000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; case 88200: fs = DA7210_PLL_FS_88200; - bypass = 0; + sysclk = 2822400; break; case 96000: fs = DA7210_PLL_FS_96000; - bypass = DA7210_PLL_BYP; + sysclk = 3072000; break; default: return -EINVAL; @@ -795,8 +846,26 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); - snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); + if (da7210->mclk_rate && (da7210->mclk_rate != sysclk)) { + /* PLL mode, disable PLL bypass */ + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, 0); + + if (!da7210->master) { + /* PLL slave mode, also enable SRM */ + snd_soc_update_bits(codec, DA7210_PLL, + (DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN), + (DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN)); + } + } else { + /* PLL bypass mode, enable PLL bypass and Auto Detection */ + snd_soc_update_bits(codec, DA7210_PLL, DA7210_MCLK_DET_EN, + DA7210_MCLK_DET_EN); + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, + DA7210_PLL_BYP); + } /* Enable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, DA7210_SC_MST_EN); @@ -810,17 +879,24 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); u32 dai_cfg1; u32 dai_cfg3; dai_cfg1 = 0x7f & snd_soc_read(codec, DA7210_DAI_CFG1); dai_cfg3 = 0xfc & snd_soc_read(codec, DA7210_DAI_CFG3); + if ((snd_soc_read(codec, DA7210_PLL) & DA7210_PLL_EN) && + (!(snd_soc_read(codec, DA7210_PLL_DIV3) & DA7210_PLL_BYP))) + return -EINVAL; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: + da7210->master = 1; dai_cfg1 |= DA7210_DAI_MODE_MASTER; break; case SND_SOC_DAIFMT_CBS_CFS: + da7210->master = 0; dai_cfg1 |= DA7210_DAI_MODE_SLAVE; break; default: @@ -872,10 +948,101 @@ static int da7210_mute(struct snd_soc_dai *dai, int mute) #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +static int da7210_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case DA7210_CLKSRC_MCLK: + switch (freq) { + case 12000000: + case 13000000: + case 13500000: + case 14400000: + case 19200000: + case 19680000: + case 19800000: + da7210->mclk_rate = freq; + return 0; + default: + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } +} + +/** + * da7210_set_dai_pll :Configure the codec PLL + * @param codec_dai : pointer to codec DAI + * @param pll_id : da7210 has only one pll, so pll_id is always zero + * @param fref : MCLK frequency, should be < 20MHz + * @param fout : FsDM value, Refer page 44 & 45 of datasheet + * @return int : Zero for success, negative error code for error + * + * Note: Supported PLL input frequencies are 12MHz, 13MHz, 13.5MHz, 14.4MHz, + * 19.2MHz, 19.6MHz and 19.8MHz + */ +static int da7210_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + + u8 pll_div1, pll_div2, pll_div3, cnt; + + /* In slave mode, there is only one set of divisors */ + if (!da7210->master) + fout = 2822400; + + /* Search pll div array for correct divisors */ + for (cnt = 0; cnt < ARRAY_SIZE(da7210_pll_div); cnt++) { + /* check fref, mode and fout */ + if ((fref == da7210_pll_div[cnt].fref) && + (da7210->master == da7210_pll_div[cnt].mode) && + (fout == da7210_pll_div[cnt].fout)) { + /* all match, pick up divisors */ + pll_div1 = da7210_pll_div[cnt].div1; + pll_div2 = da7210_pll_div[cnt].div2; + pll_div3 = da7210_pll_div[cnt].div3; + break; + } + } + if (cnt >= ARRAY_SIZE(da7210_pll_div)) + goto err; + + /* Disable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); + /* Write PLL dividers */ + snd_soc_write(codec, DA7210_PLL_DIV1, pll_div1); + snd_soc_write(codec, DA7210_PLL_DIV2, pll_div2); + snd_soc_update_bits(codec, DA7210_PLL_DIV3, + DA7210_PLL_DIV_L_MASK, pll_div3); + + /* Enable PLL */ + snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); + + /* Enable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, + DA7210_SC_MST_EN); + return 0; +err: + dev_err(codec_dai->dev, "Unsupported PLL input frequency %d\n", fref); + return -EINVAL; +} + /* DAI operations */ static const struct snd_soc_dai_ops da7210_dai_ops = { .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .set_sysclk = da7210_set_dai_sysclk, + .set_pll = da7210_set_dai_pll, .digital_mute = da7210_mute, }; @@ -915,24 +1082,11 @@ static int da7210_probe(struct snd_soc_codec *codec) return ret; } - /* FIXME - * - * This driver use fixed value here - * And below settings expects MCLK = 12.288MHz - * - * When you select different MCLK, please check... - * DA7210_PLL_DIV1 val - * DA7210_PLL_DIV2 val - * DA7210_PLL_DIV3 val - * DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx - */ + da7210->mclk_rate = 0; /* This will be set from set_sysclk() */ + da7210->master = 0; /* This will be set from set_fmt() */ - /* - * make sure that DA7210 use bypass mode before start up - */ - snd_soc_write(codec, DA7210_STARTUP1, 0); - snd_soc_write(codec, DA7210_PLL_DIV3, - DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); + /* Enable internal regulator & bias current */ + snd_soc_write(codec, DA7210_CONTROL, DA7210_REG_EN | DA7210_BIAS_EN); /* * ADC settings @@ -1007,34 +1161,13 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Enable Aux2 */ snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Set PLL Master clock range 10-20 MHz, enable PLL bypass */ + snd_soc_write(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ | + DA7210_PLL_BYP); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); - /* - * If 48kHz sound came, it use bypass mode, - * and when it is 44.1kHz, it use PLL. - * - * This time, this driver sets PLL always ON - * and controls bypass/PLL mode by switching - * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. - * see da7210_hw_params - */ - snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ - snd_soc_write(codec, DA7210_PLL_DIV2, 0x99); - snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A | - DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); - snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); - - /* As suggested by Dialog */ - /* unlock */ - regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x8B); - regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0xB4); - regmap_write(da7210->regmap, DA7210_A_PLL1, 0x01); - regmap_write(da7210->regmap, DA7210_A_CP_MODE, 0x7C); - /* re-lock */ - regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x00); - regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0x00); - /* Activate all enabled subsystem */ snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); @@ -1055,7 +1188,26 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .num_dapm_routes = ARRAY_SIZE(da7210_audio_map), }; -static struct regmap_config da7210_regmap = { +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +static struct reg_default da7210_regmap_i2c_patch[] = { + + /* System controller master disable */ + { DA7210_STARTUP1, 0x00 }, + /* Set PLL Master clock range 10-20 MHz */ + { DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ }, + + /* to unlock */ + { DA7210_A_HID_UNLOCK, 0x8B}, + { DA7210_A_TEST_UNLOCK, 0xB4}, + { DA7210_A_PLL1, 0x01}, + { DA7210_A_CP_MODE, 0x7C}, + /* to re-lock */ + { DA7210_A_HID_UNLOCK, 0x00}, + { DA7210_A_TEST_UNLOCK, 0x00}, +}; + +static const struct regmap_config da7210_regmap_config_i2c = { .reg_bits = 8, .val_bits = 8, @@ -1066,7 +1218,6 @@ static struct regmap_config da7210_regmap = { .cache_type = REGCACHE_RBTREE, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int __devinit da7210_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1080,13 +1231,18 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, da7210); - da7210->regmap = regmap_init_i2c(i2c, &da7210_regmap); + da7210->regmap = regmap_init_i2c(i2c, &da7210_regmap_config_i2c); if (IS_ERR(da7210->regmap)) { ret = PTR_ERR(da7210->regmap); dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); return ret; } + ret = regmap_register_patch(da7210->regmap, da7210_regmap_i2c_patch, + ARRAY_SIZE(da7210_regmap_i2c_patch)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da7210, &da7210_dai, 1); if (ret < 0) { @@ -1119,7 +1275,7 @@ MODULE_DEVICE_TABLE(i2c, da7210_i2c_id); /* I2C codec control layer */ static struct i2c_driver da7210_i2c_driver = { .driver = { - .name = "da7210-codec", + .name = "da7210", .owner = THIS_MODULE, }, .probe = da7210_i2c_probe, @@ -1128,12 +1284,112 @@ static struct i2c_driver da7210_i2c_driver = { }; #endif +#if defined(CONFIG_SPI_MASTER) + +static struct reg_default da7210_regmap_spi_patch[] = { + /* Dummy read to give two pulses over nCS for SPI */ + { DA7210_AUX2, 0x00 }, + { DA7210_AUX2, 0x00 }, + + /* System controller master disable */ + { DA7210_STARTUP1, 0x00 }, + /* Set PLL Master clock range 10-20 MHz */ + { DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ }, + + /* to set PAGE1 of SPI register space */ + { DA7210_PAGE_CONTROL, 0x80 }, + /* to unlock */ + { DA7210_A_HID_UNLOCK, 0x8B}, + { DA7210_A_TEST_UNLOCK, 0xB4}, + { DA7210_A_PLL1, 0x01}, + { DA7210_A_CP_MODE, 0x7C}, + /* to re-lock */ + { DA7210_A_HID_UNLOCK, 0x00}, + { DA7210_A_TEST_UNLOCK, 0x00}, + /* to set back PAGE0 of SPI register space */ + { DA7210_PAGE_CONTROL, 0x00 }, +}; + +static const struct regmap_config da7210_regmap_config_spi = { + .reg_bits = 8, + .val_bits = 8, + .read_flag_mask = 0x01, + .write_flag_mask = 0x00, + + .reg_defaults = da7210_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da7210_reg_defaults), + .volatile_reg = da7210_volatile_register, + .readable_reg = da7210_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit da7210_spi_probe(struct spi_device *spi) +{ + struct da7210_priv *da7210; + int ret; + + da7210 = devm_kzalloc(&spi->dev, sizeof(struct da7210_priv), + GFP_KERNEL); + if (!da7210) + return -ENOMEM; + + spi_set_drvdata(spi, da7210); + da7210->regmap = devm_regmap_init_spi(spi, &da7210_regmap_config_spi); + if (IS_ERR(da7210->regmap)) { + ret = PTR_ERR(da7210->regmap); + dev_err(&spi->dev, "Failed to register regmap: %d\n", ret); + return ret; + } + + ret = regmap_register_patch(da7210->regmap, da7210_regmap_spi_patch, + ARRAY_SIZE(da7210_regmap_spi_patch)); + if (ret != 0) + dev_warn(&spi->dev, "Failed to apply regmap patch: %d\n", ret); + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_da7210, &da7210_dai, 1); + if (ret < 0) + goto err_regmap; + + return ret; + +err_regmap: + regmap_exit(da7210->regmap); + + return ret; +} + +static int __devexit da7210_spi_remove(struct spi_device *spi) +{ + struct da7210_priv *da7210 = spi_get_drvdata(spi); + snd_soc_unregister_codec(&spi->dev); + regmap_exit(da7210->regmap); + return 0; +} + +static struct spi_driver da7210_spi_driver = { + .driver = { + .name = "da7210", + .owner = THIS_MODULE, + }, + .probe = da7210_spi_probe, + .remove = __devexit_p(da7210_spi_remove) +}; +#endif + static int __init da7210_modinit(void) { int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&da7210_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&da7210_spi_driver); + if (ret) { + printk(KERN_ERR "Failed to register da7210 SPI driver: %d\n", + ret); + } +#endif return ret; } module_init(da7210_modinit); @@ -1143,6 +1399,9 @@ static void __exit da7210_exit(void) #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&da7210_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&da7210_spi_driver); +#endif } module_exit(da7210_exit); diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c new file mode 100644 index 000000000000..01be2a320e21 --- /dev/null +++ b/sound/soc/codecs/da732x.c @@ -0,0 +1,1627 @@ +/* + * da732x.c --- Dialog DA732X ALSA SoC Audio Driver + * + * Copyright (C) 2012 Dialog Semiconductor GmbH + * + * Author: Michal Hajduk <Michal.Hajduk@diasemi.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/sysfs.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <asm/div64.h> + +#include "da732x.h" +#include "da732x_reg.h" + + +struct da732x_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + + unsigned int sysclk; + bool pll_en; +}; + +/* + * da732x register cache - default settings + */ +static struct reg_default da732x_reg_cache[] = { + { DA732X_REG_REF1 , 0x02 }, + { DA732X_REG_BIAS_EN , 0x80 }, + { DA732X_REG_BIAS1 , 0x00 }, + { DA732X_REG_BIAS2 , 0x00 }, + { DA732X_REG_BIAS3 , 0x00 }, + { DA732X_REG_BIAS4 , 0x00 }, + { DA732X_REG_MICBIAS2 , 0x00 }, + { DA732X_REG_MICBIAS1 , 0x00 }, + { DA732X_REG_MICDET , 0x00 }, + { DA732X_REG_MIC1_PRE , 0x01 }, + { DA732X_REG_MIC1 , 0x40 }, + { DA732X_REG_MIC2_PRE , 0x01 }, + { DA732X_REG_MIC2 , 0x40 }, + { DA732X_REG_AUX1L , 0x75 }, + { DA732X_REG_AUX1R , 0x75 }, + { DA732X_REG_MIC3_PRE , 0x01 }, + { DA732X_REG_MIC3 , 0x40 }, + { DA732X_REG_INP_PINBIAS , 0x00 }, + { DA732X_REG_INP_ZC_EN , 0x00 }, + { DA732X_REG_INP_MUX , 0x50 }, + { DA732X_REG_HP_DET , 0x00 }, + { DA732X_REG_HPL_DAC_OFFSET , 0x00 }, + { DA732X_REG_HPL_DAC_OFF_CNTL , 0x00 }, + { DA732X_REG_HPL_OUT_OFFSET , 0x00 }, + { DA732X_REG_HPL , 0x40 }, + { DA732X_REG_HPL_VOL , 0x0F }, + { DA732X_REG_HPR_DAC_OFFSET , 0x00 }, + { DA732X_REG_HPR_DAC_OFF_CNTL , 0x00 }, + { DA732X_REG_HPR_OUT_OFFSET , 0x00 }, + { DA732X_REG_HPR , 0x40 }, + { DA732X_REG_HPR_VOL , 0x0F }, + { DA732X_REG_LIN2 , 0x4F }, + { DA732X_REG_LIN3 , 0x4F }, + { DA732X_REG_LIN4 , 0x4F }, + { DA732X_REG_OUT_ZC_EN , 0x00 }, + { DA732X_REG_HP_LIN1_GNDSEL , 0x00 }, + { DA732X_REG_CP_HP1 , 0x0C }, + { DA732X_REG_CP_HP2 , 0x03 }, + { DA732X_REG_CP_CTRL1 , 0x00 }, + { DA732X_REG_CP_CTRL2 , 0x99 }, + { DA732X_REG_CP_CTRL3 , 0x25 }, + { DA732X_REG_CP_LEVEL_MASK , 0x3F }, + { DA732X_REG_CP_DET , 0x00 }, + { DA732X_REG_CP_STATUS , 0x00 }, + { DA732X_REG_CP_THRESH1 , 0x00 }, + { DA732X_REG_CP_THRESH2 , 0x00 }, + { DA732X_REG_CP_THRESH3 , 0x00 }, + { DA732X_REG_CP_THRESH4 , 0x00 }, + { DA732X_REG_CP_THRESH5 , 0x00 }, + { DA732X_REG_CP_THRESH6 , 0x00 }, + { DA732X_REG_CP_THRESH7 , 0x00 }, + { DA732X_REG_CP_THRESH8 , 0x00 }, + { DA732X_REG_PLL_DIV_LO , 0x00 }, + { DA732X_REG_PLL_DIV_MID , 0x00 }, + { DA732X_REG_PLL_DIV_HI , 0x00 }, + { DA732X_REG_PLL_CTRL , 0x02 }, + { DA732X_REG_CLK_CTRL , 0xaa }, + { DA732X_REG_CLK_DSP , 0x07 }, + { DA732X_REG_CLK_EN1 , 0x00 }, + { DA732X_REG_CLK_EN2 , 0x00 }, + { DA732X_REG_CLK_EN3 , 0x00 }, + { DA732X_REG_CLK_EN4 , 0x00 }, + { DA732X_REG_CLK_EN5 , 0x00 }, + { DA732X_REG_AIF_MCLK , 0x00 }, + { DA732X_REG_AIFA1 , 0x02 }, + { DA732X_REG_AIFA2 , 0x00 }, + { DA732X_REG_AIFA3 , 0x08 }, + { DA732X_REG_AIFB1 , 0x02 }, + { DA732X_REG_AIFB2 , 0x00 }, + { DA732X_REG_AIFB3 , 0x08 }, + { DA732X_REG_PC_CTRL , 0xC0 }, + { DA732X_REG_DATA_ROUTE , 0x00 }, + { DA732X_REG_DSP_CTRL , 0x00 }, + { DA732X_REG_CIF_CTRL2 , 0x00 }, + { DA732X_REG_HANDSHAKE , 0x00 }, + { DA732X_REG_SPARE1_OUT , 0x00 }, + { DA732X_REG_SPARE2_OUT , 0x00 }, + { DA732X_REG_SPARE1_IN , 0x00 }, + { DA732X_REG_ADC1_PD , 0x00 }, + { DA732X_REG_ADC1_HPF , 0x00 }, + { DA732X_REG_ADC1_SEL , 0x00 }, + { DA732X_REG_ADC1_EQ12 , 0x00 }, + { DA732X_REG_ADC1_EQ34 , 0x00 }, + { DA732X_REG_ADC1_EQ5 , 0x00 }, + { DA732X_REG_ADC2_PD , 0x00 }, + { DA732X_REG_ADC2_HPF , 0x00 }, + { DA732X_REG_ADC2_SEL , 0x00 }, + { DA732X_REG_ADC2_EQ12 , 0x00 }, + { DA732X_REG_ADC2_EQ34 , 0x00 }, + { DA732X_REG_ADC2_EQ5 , 0x00 }, + { DA732X_REG_DAC1_HPF , 0x00 }, + { DA732X_REG_DAC1_L_VOL , 0x00 }, + { DA732X_REG_DAC1_R_VOL , 0x00 }, + { DA732X_REG_DAC1_SEL , 0x00 }, + { DA732X_REG_DAC1_SOFTMUTE , 0x00 }, + { DA732X_REG_DAC1_EQ12 , 0x00 }, + { DA732X_REG_DAC1_EQ34 , 0x00 }, + { DA732X_REG_DAC1_EQ5 , 0x00 }, + { DA732X_REG_DAC2_HPF , 0x00 }, + { DA732X_REG_DAC2_L_VOL , 0x00 }, + { DA732X_REG_DAC2_R_VOL , 0x00 }, + { DA732X_REG_DAC2_SEL , 0x00 }, + { DA732X_REG_DAC2_SOFTMUTE , 0x00 }, + { DA732X_REG_DAC2_EQ12 , 0x00 }, + { DA732X_REG_DAC2_EQ34 , 0x00 }, + { DA732X_REG_DAC2_EQ5 , 0x00 }, + { DA732X_REG_DAC3_HPF , 0x00 }, + { DA732X_REG_DAC3_VOL , 0x00 }, + { DA732X_REG_DAC3_SEL , 0x00 }, + { DA732X_REG_DAC3_SOFTMUTE , 0x00 }, + { DA732X_REG_DAC3_EQ12 , 0x00 }, + { DA732X_REG_DAC3_EQ34 , 0x00 }, + { DA732X_REG_DAC3_EQ5 , 0x00 }, + { DA732X_REG_BIQ_BYP , 0x00 }, + { DA732X_REG_DMA_CMD , 0x00 }, + { DA732X_REG_DMA_ADDR0 , 0x00 }, + { DA732X_REG_DMA_ADDR1 , 0x00 }, + { DA732X_REG_DMA_DATA0 , 0x00 }, + { DA732X_REG_DMA_DATA1 , 0x00 }, + { DA732X_REG_DMA_DATA2 , 0x00 }, + { DA732X_REG_DMA_DATA3 , 0x00 }, + { DA732X_REG_UNLOCK , 0x00 }, +}; + +static inline int da732x_get_input_div(struct snd_soc_codec *codec, int sysclk) +{ + int val; + int ret; + + if (sysclk < DA732X_MCLK_10MHZ) { + val = DA732X_MCLK_RET_0_10MHZ; + ret = DA732X_MCLK_VAL_0_10MHZ; + } else if ((sysclk >= DA732X_MCLK_10MHZ) && + (sysclk < DA732X_MCLK_20MHZ)) { + val = DA732X_MCLK_RET_10_20MHZ; + ret = DA732X_MCLK_VAL_10_20MHZ; + } else if ((sysclk >= DA732X_MCLK_20MHZ) && + (sysclk < DA732X_MCLK_40MHZ)) { + val = DA732X_MCLK_RET_20_40MHZ; + ret = DA732X_MCLK_VAL_20_40MHZ; + } else if ((sysclk >= DA732X_MCLK_40MHZ) && + (sysclk <= DA732X_MCLK_54MHZ)) { + val = DA732X_MCLK_RET_40_54MHZ; + ret = DA732X_MCLK_VAL_40_54MHZ; + } else { + return -EINVAL; + } + + snd_soc_write(codec, DA732X_REG_PLL_CTRL, val); + + return ret; +} + +static void da732x_set_charge_pump(struct snd_soc_codec *codec, int state) +{ + switch (state) { + case DA732X_ENABLE_CP: + snd_soc_write(codec, DA732X_REG_CLK_EN2, DA732X_CP_CLK_EN); + snd_soc_write(codec, DA732X_REG_CP_HP2, DA732X_HP_CP_EN | + DA732X_HP_CP_REG | DA732X_HP_CP_PULSESKIP); + snd_soc_write(codec, DA732X_REG_CP_CTRL1, DA732X_CP_EN | + DA732X_CP_CTRL_CPVDD1); + snd_soc_write(codec, DA732X_REG_CP_CTRL2, + DA732X_CP_MANAGE_MAGNITUDE | DA732X_CP_BOOST); + snd_soc_write(codec, DA732X_REG_CP_CTRL3, DA732X_CP_1MHZ); + break; + case DA732X_DISABLE_CP: + snd_soc_write(codec, DA732X_REG_CLK_EN2, DA732X_CP_CLK_DIS); + snd_soc_write(codec, DA732X_REG_CP_HP2, DA732X_HP_CP_DIS); + snd_soc_write(codec, DA732X_REG_CP_CTRL1, DA723X_CP_DIS); + break; + default: + pr_err(KERN_ERR "Wrong charge pump state\n"); + break; + } +} + +static const DECLARE_TLV_DB_SCALE(mic_boost_tlv, DA732X_MIC_PRE_VOL_DB_MIN, + DA732X_MIC_PRE_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(mic_pga_tlv, DA732X_MIC_VOL_DB_MIN, + DA732X_MIC_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(aux_pga_tlv, DA732X_AUX_VOL_DB_MIN, + DA732X_AUX_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(hp_pga_tlv, DA732X_HP_VOL_DB_MIN, + DA732X_AUX_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(lin2_pga_tlv, DA732X_LIN2_VOL_DB_MIN, + DA732X_LIN2_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(lin3_pga_tlv, DA732X_LIN3_VOL_DB_MIN, + DA732X_LIN3_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(lin4_pga_tlv, DA732X_LIN4_VOL_DB_MIN, + DA732X_LIN4_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(adc_pga_tlv, DA732X_ADC_VOL_DB_MIN, + DA732X_ADC_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(dac_pga_tlv, DA732X_DAC_VOL_DB_MIN, + DA732X_DAC_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(eq_band_pga_tlv, DA732X_EQ_BAND_VOL_DB_MIN, + DA732X_EQ_BAND_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(eq_overall_tlv, DA732X_EQ_OVERALL_VOL_DB_MIN, + DA732X_EQ_OVERALL_VOL_DB_INC, 0); + +/* High Pass Filter */ +static const char *da732x_hpf_mode[] = { + "Disable", "Music", "Voice", +}; + +static const char *da732x_hpf_music[] = { + "1.8Hz", "3.75Hz", "7.5Hz", "15Hz", +}; + +static const char *da732x_hpf_voice[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", + "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da732x_dac1_hpf_mode_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MODE_SHIFT, + DA732X_HPF_MODE_MAX, da732x_hpf_mode) +}; + +static const struct soc_enum da732x_dac2_hpf_mode_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MODE_SHIFT, + DA732X_HPF_MODE_MAX, da732x_hpf_mode) +}; + +static const struct soc_enum da732x_dac3_hpf_mode_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MODE_SHIFT, + DA732X_HPF_MODE_MAX, da732x_hpf_mode) +}; + +static const struct soc_enum da732x_adc1_hpf_mode_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MODE_SHIFT, + DA732X_HPF_MODE_MAX, da732x_hpf_mode) +}; + +static const struct soc_enum da732x_adc2_hpf_mode_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MODE_SHIFT, + DA732X_HPF_MODE_MAX, da732x_hpf_mode) +}; + +static const struct soc_enum da732x_dac1_hp_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MUSIC_SHIFT, + DA732X_HPF_MUSIC_MAX, da732x_hpf_music) +}; + +static const struct soc_enum da732x_dac2_hp_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MUSIC_SHIFT, + DA732X_HPF_MUSIC_MAX, da732x_hpf_music) +}; + +static const struct soc_enum da732x_dac3_hp_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MUSIC_SHIFT, + DA732X_HPF_MUSIC_MAX, da732x_hpf_music) +}; + +static const struct soc_enum da732x_adc1_hp_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MUSIC_SHIFT, + DA732X_HPF_MUSIC_MAX, da732x_hpf_music) +}; + +static const struct soc_enum da732x_adc2_hp_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MUSIC_SHIFT, + DA732X_HPF_MUSIC_MAX, da732x_hpf_music) +}; + +static const struct soc_enum da732x_dac1_voice_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_VOICE_SHIFT, + DA732X_HPF_VOICE_MAX, da732x_hpf_voice) +}; + +static const struct soc_enum da732x_dac2_voice_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_VOICE_SHIFT, + DA732X_HPF_VOICE_MAX, da732x_hpf_voice) +}; + +static const struct soc_enum da732x_dac3_voice_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_VOICE_SHIFT, + DA732X_HPF_VOICE_MAX, da732x_hpf_voice) +}; + +static const struct soc_enum da732x_adc1_voice_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_VOICE_SHIFT, + DA732X_HPF_VOICE_MAX, da732x_hpf_voice) +}; + +static const struct soc_enum da732x_adc2_voice_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_VOICE_SHIFT, + DA732X_HPF_VOICE_MAX, da732x_hpf_voice) +}; + + +static int da732x_hpf_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *enum_ctrl = (struct soc_enum *)kcontrol->private_value; + unsigned int reg = enum_ctrl->reg; + unsigned int sel = ucontrol->value.integer.value[0]; + unsigned int bits; + + switch (sel) { + case DA732X_HPF_DISABLED: + bits = DA732X_HPF_DIS; + break; + case DA732X_HPF_VOICE: + bits = DA732X_HPF_VOICE_EN; + break; + case DA732X_HPF_MUSIC: + bits = DA732X_HPF_MUSIC_EN; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, reg, DA732X_HPF_MASK, bits); + + return 0; +} + +static int da732x_hpf_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *enum_ctrl = (struct soc_enum *)kcontrol->private_value; + unsigned int reg = enum_ctrl->reg; + int val; + + val = snd_soc_read(codec, reg) & DA732X_HPF_MASK; + + switch (val) { + case DA732X_HPF_VOICE_EN: + ucontrol->value.integer.value[0] = DA732X_HPF_VOICE; + break; + case DA732X_HPF_MUSIC_EN: + ucontrol->value.integer.value[0] = DA732X_HPF_MUSIC; + break; + default: + ucontrol->value.integer.value[0] = DA732X_HPF_DISABLED; + break; + } + + return 0; +} + +static const struct snd_kcontrol_new da732x_snd_controls[] = { + /* Input PGAs */ + SOC_SINGLE_RANGE_TLV("MIC1 Boost Volume", DA732X_REG_MIC1_PRE, + DA732X_MICBOOST_SHIFT, DA732X_MICBOOST_MIN, + DA732X_MICBOOST_MAX, 0, mic_boost_tlv), + SOC_SINGLE_RANGE_TLV("MIC2 Boost Volume", DA732X_REG_MIC2_PRE, + DA732X_MICBOOST_SHIFT, DA732X_MICBOOST_MIN, + DA732X_MICBOOST_MAX, 0, mic_boost_tlv), + SOC_SINGLE_RANGE_TLV("MIC3 Boost Volume", DA732X_REG_MIC3_PRE, + DA732X_MICBOOST_SHIFT, DA732X_MICBOOST_MIN, + DA732X_MICBOOST_MAX, 0, mic_boost_tlv), + + /* MICs */ + SOC_SINGLE("MIC1 Switch", DA732X_REG_MIC1, DA732X_MIC_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_RANGE_TLV("MIC1 Volume", DA732X_REG_MIC1, + DA732X_MIC_VOL_SHIFT, DA732X_MIC_VOL_VAL_MIN, + DA732X_MIC_VOL_VAL_MAX, 0, mic_pga_tlv), + SOC_SINGLE("MIC2 Switch", DA732X_REG_MIC2, DA732X_MIC_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_RANGE_TLV("MIC2 Volume", DA732X_REG_MIC2, + DA732X_MIC_VOL_SHIFT, DA732X_MIC_VOL_VAL_MIN, + DA732X_MIC_VOL_VAL_MAX, 0, mic_pga_tlv), + SOC_SINGLE("MIC3 Switch", DA732X_REG_MIC3, DA732X_MIC_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_RANGE_TLV("MIC3 Volume", DA732X_REG_MIC3, + DA732X_MIC_VOL_SHIFT, DA732X_MIC_VOL_VAL_MIN, + DA732X_MIC_VOL_VAL_MAX, 0, mic_pga_tlv), + + /* AUXs */ + SOC_SINGLE("AUX1L Switch", DA732X_REG_AUX1L, DA732X_AUX_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("AUX1L Volume", DA732X_REG_AUX1L, + DA732X_AUX_VOL_SHIFT, DA732X_AUX_VOL_VAL_MAX, + DA732X_NO_INVERT, aux_pga_tlv), + SOC_SINGLE("AUX1R Switch", DA732X_REG_AUX1R, DA732X_AUX_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("AUX1R Volume", DA732X_REG_AUX1R, + DA732X_AUX_VOL_SHIFT, DA732X_AUX_VOL_VAL_MAX, + DA732X_NO_INVERT, aux_pga_tlv), + + /* ADCs */ + SOC_DOUBLE_TLV("ADC1 Volume", DA732X_REG_ADC1_SEL, + DA732X_ADCL_VOL_SHIFT, DA732X_ADCR_VOL_SHIFT, + DA732X_ADC_VOL_VAL_MAX, DA732X_INVERT, adc_pga_tlv), + + SOC_DOUBLE_TLV("ADC2 Volume", DA732X_REG_ADC2_SEL, + DA732X_ADCL_VOL_SHIFT, DA732X_ADCR_VOL_SHIFT, + DA732X_ADC_VOL_VAL_MAX, DA732X_INVERT, adc_pga_tlv), + + /* DACs */ + SOC_DOUBLE("Digital Playback DAC12 Switch", DA732X_REG_DAC1_SEL, + DA732X_DACL_MUTE_SHIFT, DA732X_DACR_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_DOUBLE_R_TLV("Digital Playback DAC12 Volume", DA732X_REG_DAC1_L_VOL, + DA732X_REG_DAC1_R_VOL, DA732X_DAC_VOL_SHIFT, + DA732X_DAC_VOL_VAL_MAX, DA732X_INVERT, dac_pga_tlv), + SOC_SINGLE("Digital Playback DAC3 Switch", DA732X_REG_DAC2_SEL, + DA732X_DACL_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("Digital Playback DAC3 Volume", DA732X_REG_DAC2_L_VOL, + DA732X_DAC_VOL_SHIFT, DA732X_DAC_VOL_VAL_MAX, + DA732X_INVERT, dac_pga_tlv), + SOC_SINGLE("Digital Playback DAC4 Switch", DA732X_REG_DAC2_SEL, + DA732X_DACR_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("Digital Playback DAC4 Volume", DA732X_REG_DAC2_R_VOL, + DA732X_DAC_VOL_SHIFT, DA732X_DAC_VOL_VAL_MAX, + DA732X_INVERT, dac_pga_tlv), + SOC_SINGLE("Digital Playback DAC5 Switch", DA732X_REG_DAC3_SEL, + DA732X_DACL_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("Digital Playback DAC5 Volume", DA732X_REG_DAC3_VOL, + DA732X_DAC_VOL_SHIFT, DA732X_DAC_VOL_VAL_MAX, + DA732X_INVERT, dac_pga_tlv), + + /* High Pass Filters */ + SOC_ENUM_EXT("DAC1 High Pass Filter Mode", + da732x_dac1_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set), + SOC_ENUM("DAC1 High Pass Filter", da732x_dac1_hp_filter_enum), + SOC_ENUM("DAC1 Voice Filter", da732x_dac1_voice_filter_enum), + + SOC_ENUM_EXT("DAC2 High Pass Filter Mode", + da732x_dac2_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set), + SOC_ENUM("DAC2 High Pass Filter", da732x_dac2_hp_filter_enum), + SOC_ENUM("DAC2 Voice Filter", da732x_dac2_voice_filter_enum), + + SOC_ENUM_EXT("DAC3 High Pass Filter Mode", + da732x_dac3_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set), + SOC_ENUM("DAC3 High Pass Filter", da732x_dac3_hp_filter_enum), + SOC_ENUM("DAC3 Filter Mode", da732x_dac3_voice_filter_enum), + + SOC_ENUM_EXT("ADC1 High Pass Filter Mode", + da732x_adc1_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set), + SOC_ENUM("ADC1 High Pass Filter", da732x_adc1_hp_filter_enum), + SOC_ENUM("ADC1 Voice Filter", da732x_adc1_voice_filter_enum), + + SOC_ENUM_EXT("ADC2 High Pass Filter Mode", + da732x_adc2_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set), + SOC_ENUM("ADC2 High Pass Filter", da732x_adc2_hp_filter_enum), + SOC_ENUM("ADC2 Voice Filter", da732x_adc2_voice_filter_enum), + + /* Equalizers */ + SOC_SINGLE("ADC1 EQ Switch", DA732X_REG_ADC1_EQ5, + DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT), + SOC_SINGLE_TLV("ADC1 EQ Band 1 Volume", DA732X_REG_ADC1_EQ12, + DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC1 EQ Band 2 Volume", DA732X_REG_ADC1_EQ12, + DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC1 EQ Band 3 Volume", DA732X_REG_ADC1_EQ34, + DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC1 EQ Band 4 Volume", DA732X_REG_ADC1_EQ34, + DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC1 EQ Band 5 Volume", DA732X_REG_ADC1_EQ5, + DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC1 EQ Overall Volume", DA732X_REG_ADC1_EQ5, + DA732X_EQ_OVERALL_SHIFT, DA732X_EQ_OVERALL_VOL_VAL_MAX, + DA732X_INVERT, eq_overall_tlv), + + SOC_SINGLE("ADC2 EQ Switch", DA732X_REG_ADC2_EQ5, + DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT), + SOC_SINGLE_TLV("ADC2 EQ Band 1 Volume", DA732X_REG_ADC2_EQ12, + DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC2 EQ Band 2 Volume", DA732X_REG_ADC2_EQ12, + DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC2 EQ Band 3 Volume", DA732X_REG_ADC2_EQ34, + DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ACD2 EQ Band 4 Volume", DA732X_REG_ADC2_EQ34, + DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ACD2 EQ Band 5 Volume", DA732X_REG_ADC2_EQ5, + DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC2 EQ Overall Volume", DA732X_REG_ADC1_EQ5, + DA732X_EQ_OVERALL_SHIFT, DA732X_EQ_OVERALL_VOL_VAL_MAX, + DA732X_INVERT, eq_overall_tlv), + + SOC_SINGLE("DAC1 EQ Switch", DA732X_REG_DAC1_EQ5, + DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT), + SOC_SINGLE_TLV("DAC1 EQ Band 1 Volume", DA732X_REG_DAC1_EQ12, + DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC1 EQ Band 2 Volume", DA732X_REG_DAC1_EQ12, + DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC1 EQ Band 3 Volume", DA732X_REG_DAC1_EQ34, + DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC1 EQ Band 4 Volume", DA732X_REG_DAC1_EQ34, + DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC1 EQ Band 5 Volume", DA732X_REG_DAC1_EQ5, + DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + + SOC_SINGLE("DAC2 EQ Switch", DA732X_REG_DAC2_EQ5, + DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT), + SOC_SINGLE_TLV("DAC2 EQ Band 1 Volume", DA732X_REG_DAC2_EQ12, + DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC2 EQ Band 2 Volume", DA732X_REG_DAC2_EQ12, + DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC2 EQ Band 3 Volume", DA732X_REG_DAC2_EQ34, + DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC2 EQ Band 4 Volume", DA732X_REG_DAC2_EQ34, + DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC2 EQ Band 5 Volume", DA732X_REG_DAC2_EQ5, + DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + + SOC_SINGLE("DAC3 EQ Switch", DA732X_REG_DAC3_EQ5, + DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT), + SOC_SINGLE_TLV("DAC3 EQ Band 1 Volume", DA732X_REG_DAC3_EQ12, + DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC3 EQ Band 2 Volume", DA732X_REG_DAC3_EQ12, + DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC3 EQ Band 3 Volume", DA732X_REG_DAC3_EQ34, + DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC3 EQ Band 4 Volume", DA732X_REG_DAC3_EQ34, + DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC3 EQ Band 5 Volume", DA732X_REG_DAC3_EQ5, + DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + + /* Lineout 2 Reciever*/ + SOC_SINGLE("Lineout 2 Switch", DA732X_REG_LIN2, DA732X_LOUT_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("Lineout 2 Volume", DA732X_REG_LIN2, + DA732X_LOUT_VOL_SHIFT, DA732X_LOUT_VOL_VAL_MAX, + DA732X_NO_INVERT, lin2_pga_tlv), + + /* Lineout 3 SPEAKER*/ + SOC_SINGLE("Lineout 3 Switch", DA732X_REG_LIN3, DA732X_LOUT_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("Lineout 3 Volume", DA732X_REG_LIN3, + DA732X_LOUT_VOL_SHIFT, DA732X_LOUT_VOL_VAL_MAX, + DA732X_NO_INVERT, lin3_pga_tlv), + + /* Lineout 4 */ + SOC_SINGLE("Lineout 4 Switch", DA732X_REG_LIN4, DA732X_LOUT_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("Lineout 4 Volume", DA732X_REG_LIN4, + DA732X_LOUT_VOL_SHIFT, DA732X_LOUT_VOL_VAL_MAX, + DA732X_NO_INVERT, lin4_pga_tlv), + + /* Headphones */ + SOC_DOUBLE_R("Headphone Switch", DA732X_REG_HPR, DA732X_REG_HPL, + DA732X_HP_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_DOUBLE_R_TLV("Headphone Volume", DA732X_REG_HPL_VOL, + DA732X_REG_HPR_VOL, DA732X_HP_VOL_SHIFT, + DA732X_HP_VOL_VAL_MAX, DA732X_NO_INVERT, hp_pga_tlv), +}; + +static int da732x_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + switch (w->reg) { + case DA732X_REG_ADC1_PD: + snd_soc_update_bits(codec, DA732X_REG_CLK_EN3, + DA732X_ADCA_BB_CLK_EN, + DA732X_ADCA_BB_CLK_EN); + break; + case DA732X_REG_ADC2_PD: + snd_soc_update_bits(codec, DA732X_REG_CLK_EN3, + DA732X_ADCC_BB_CLK_EN, + DA732X_ADCC_BB_CLK_EN); + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, w->reg, DA732X_ADC_RST_MASK, + DA732X_ADC_SET_ACT); + snd_soc_update_bits(codec, w->reg, DA732X_ADC_PD_MASK, + DA732X_ADC_ON); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, w->reg, DA732X_ADC_PD_MASK, + DA732X_ADC_OFF); + snd_soc_update_bits(codec, w->reg, DA732X_ADC_RST_MASK, + DA732X_ADC_SET_RST); + + switch (w->reg) { + case DA732X_REG_ADC1_PD: + snd_soc_update_bits(codec, DA732X_REG_CLK_EN3, + DA732X_ADCA_BB_CLK_EN, 0); + break; + case DA732X_REG_ADC2_PD: + snd_soc_update_bits(codec, DA732X_REG_CLK_EN3, + DA732X_ADCC_BB_CLK_EN, 0); + break; + default: + return -EINVAL; + } + + break; + default: + return -EINVAL; + } + + return 0; +} + +static int da732x_out_pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, w->reg, + (1 << w->shift) | DA732X_OUT_HIZ_EN, + (1 << w->shift) | DA732X_OUT_HIZ_EN); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, w->reg, + (1 << w->shift) | DA732X_OUT_HIZ_EN, + (1 << w->shift) | DA732X_OUT_HIZ_DIS); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const char *adcl_text[] = { + "AUX1L", "MIC1" +}; + +static const char *adcr_text[] = { + "AUX1R", "MIC2", "MIC3" +}; + +static const char *enable_text[] = { + "Disabled", + "Enabled" +}; + +/* ADC1LMUX */ +static const struct soc_enum adc1l_enum = + SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1L_MUX_SEL_SHIFT, + DA732X_ADCL_MUX_MAX, adcl_text); +static const struct snd_kcontrol_new adc1l_mux = + SOC_DAPM_ENUM("ADC Route", adc1l_enum); + +/* ADC1RMUX */ +static const struct soc_enum adc1r_enum = + SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1R_MUX_SEL_SHIFT, + DA732X_ADCR_MUX_MAX, adcr_text); +static const struct snd_kcontrol_new adc1r_mux = + SOC_DAPM_ENUM("ADC Route", adc1r_enum); + +/* ADC2LMUX */ +static const struct soc_enum adc2l_enum = + SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2L_MUX_SEL_SHIFT, + DA732X_ADCL_MUX_MAX, adcl_text); +static const struct snd_kcontrol_new adc2l_mux = + SOC_DAPM_ENUM("ADC Route", adc2l_enum); + +/* ADC2RMUX */ +static const struct soc_enum adc2r_enum = + SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2R_MUX_SEL_SHIFT, + DA732X_ADCR_MUX_MAX, adcr_text); + +static const struct snd_kcontrol_new adc2r_mux = + SOC_DAPM_ENUM("ADC Route", adc2r_enum); + +static const struct soc_enum da732x_hp_left_output = + SOC_ENUM_SINGLE(DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN_SHIFT, + DA732X_DAC_EN_MAX, enable_text); + +static const struct snd_kcontrol_new hpl_mux = + SOC_DAPM_ENUM("HPL Switch", da732x_hp_left_output); + +static const struct soc_enum da732x_hp_right_output = + SOC_ENUM_SINGLE(DA732X_REG_HPR, DA732X_HP_OUT_DAC_EN_SHIFT, + DA732X_DAC_EN_MAX, enable_text); + +static const struct snd_kcontrol_new hpr_mux = + SOC_DAPM_ENUM("HPR Switch", da732x_hp_right_output); + +static const struct soc_enum da732x_speaker_output = + SOC_ENUM_SINGLE(DA732X_REG_LIN3, DA732X_LOUT_DAC_EN_SHIFT, + DA732X_DAC_EN_MAX, enable_text); + +static const struct snd_kcontrol_new spk_mux = + SOC_DAPM_ENUM("SPK Switch", da732x_speaker_output); + +static const struct soc_enum da732x_lout4_output = + SOC_ENUM_SINGLE(DA732X_REG_LIN4, DA732X_LOUT_DAC_EN_SHIFT, + DA732X_DAC_EN_MAX, enable_text); + +static const struct snd_kcontrol_new lout4_mux = + SOC_DAPM_ENUM("LOUT4 Switch", da732x_lout4_output); + +static const struct soc_enum da732x_lout2_output = + SOC_ENUM_SINGLE(DA732X_REG_LIN2, DA732X_LOUT_DAC_EN_SHIFT, + DA732X_DAC_EN_MAX, enable_text); + +static const struct snd_kcontrol_new lout2_mux = + SOC_DAPM_ENUM("LOUT2 Switch", da732x_lout2_output); + +static const struct snd_soc_dapm_widget da732x_dapm_widgets[] = { + /* Supplies */ + SND_SOC_DAPM_SUPPLY("ADC1 Supply", DA732X_REG_ADC1_PD, 0, + DA732X_NO_INVERT, da732x_adc_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("ADC2 Supply", DA732X_REG_ADC2_PD, 0, + DA732X_NO_INVERT, da732x_adc_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("DAC1 CLK", DA732X_REG_CLK_EN4, + DA732X_DACA_BB_CLK_SHIFT, DA732X_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC2 CLK", DA732X_REG_CLK_EN4, + DA732X_DACC_BB_CLK_SHIFT, DA732X_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC3 CLK", DA732X_REG_CLK_EN5, + DA732X_DACE_BB_CLK_SHIFT, DA732X_NO_INVERT, + NULL, 0), + + /* Micbias */ + SND_SOC_DAPM_SUPPLY("MICBIAS1", DA732X_REG_MICBIAS1, + DA732X_MICBIAS_EN_SHIFT, + DA732X_NO_INVERT, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", DA732X_REG_MICBIAS2, + DA732X_MICBIAS_EN_SHIFT, + DA732X_NO_INVERT, NULL, 0), + + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + SND_SOC_DAPM_INPUT("AUX1L"), + SND_SOC_DAPM_INPUT("AUX1R"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("LOUTL"), + SND_SOC_DAPM_OUTPUT("LOUTR"), + SND_SOC_DAPM_OUTPUT("ClassD"), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC1L", NULL, DA732X_REG_ADC1_SEL, + DA732X_ADCL_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_ADC("ADC1R", NULL, DA732X_REG_ADC1_SEL, + DA732X_ADCR_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_ADC("ADC2L", NULL, DA732X_REG_ADC2_SEL, + DA732X_ADCL_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_ADC("ADC2R", NULL, DA732X_REG_ADC2_SEL, + DA732X_ADCR_EN_SHIFT, DA732X_NO_INVERT), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC1L", NULL, DA732X_REG_DAC1_SEL, + DA732X_DACL_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_DAC("DAC1R", NULL, DA732X_REG_DAC1_SEL, + DA732X_DACR_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_DAC("DAC2L", NULL, DA732X_REG_DAC2_SEL, + DA732X_DACL_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_DAC("DAC2R", NULL, DA732X_REG_DAC2_SEL, + DA732X_DACR_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_DAC("DAC3", NULL, DA732X_REG_DAC3_SEL, + DA732X_DACL_EN_SHIFT, DA732X_NO_INVERT), + + /* Input Pgas */ + SND_SOC_DAPM_PGA("MIC1 PGA", DA732X_REG_MIC1, DA732X_MIC_EN_SHIFT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("MIC2 PGA", DA732X_REG_MIC2, DA732X_MIC_EN_SHIFT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("MIC3 PGA", DA732X_REG_MIC3, DA732X_MIC_EN_SHIFT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX1L PGA", DA732X_REG_AUX1L, DA732X_AUX_EN_SHIFT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX1R PGA", DA732X_REG_AUX1R, DA732X_AUX_EN_SHIFT, + 0, NULL, 0), + + SND_SOC_DAPM_PGA_E("HP Left", DA732X_REG_HPL, DA732X_HP_OUT_EN_SHIFT, + 0, NULL, 0, da732x_out_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("HP Right", DA732X_REG_HPR, DA732X_HP_OUT_EN_SHIFT, + 0, NULL, 0, da732x_out_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("LIN2", DA732X_REG_LIN2, DA732X_LIN_OUT_EN_SHIFT, + 0, NULL, 0, da732x_out_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("LIN3", DA732X_REG_LIN3, DA732X_LIN_OUT_EN_SHIFT, + 0, NULL, 0, da732x_out_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("LIN4", DA732X_REG_LIN4, DA732X_LIN_OUT_EN_SHIFT, + 0, NULL, 0, da732x_out_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* MUXs */ + SND_SOC_DAPM_MUX("ADC1 Left MUX", SND_SOC_NOPM, 0, 0, &adc1l_mux), + SND_SOC_DAPM_MUX("ADC1 Right MUX", SND_SOC_NOPM, 0, 0, &adc1r_mux), + SND_SOC_DAPM_MUX("ADC2 Left MUX", SND_SOC_NOPM, 0, 0, &adc2l_mux), + SND_SOC_DAPM_MUX("ADC2 Right MUX", SND_SOC_NOPM, 0, 0, &adc2r_mux), + + SND_SOC_DAPM_MUX("HP Left MUX", SND_SOC_NOPM, 0, 0, &hpl_mux), + SND_SOC_DAPM_MUX("HP Right MUX", SND_SOC_NOPM, 0, 0, &hpr_mux), + SND_SOC_DAPM_MUX("Speaker MUX", SND_SOC_NOPM, 0, 0, &spk_mux), + SND_SOC_DAPM_MUX("LOUT2 MUX", SND_SOC_NOPM, 0, 0, &lout2_mux), + SND_SOC_DAPM_MUX("LOUT4 MUX", SND_SOC_NOPM, 0, 0, &lout4_mux), + + /* AIF interfaces */ + SND_SOC_DAPM_AIF_OUT("AIFA Output", "AIFA Capture", 0, DA732X_REG_AIFA3, + DA732X_AIF_EN_SHIFT, 0), + SND_SOC_DAPM_AIF_IN("AIFA Input", "AIFA Playback", 0, DA732X_REG_AIFA3, + DA732X_AIF_EN_SHIFT, 0), + + SND_SOC_DAPM_AIF_OUT("AIFB Output", "AIFB Capture", 0, DA732X_REG_AIFB3, + DA732X_AIF_EN_SHIFT, 0), + SND_SOC_DAPM_AIF_IN("AIFB Input", "AIFB Playback", 0, DA732X_REG_AIFB3, + DA732X_AIF_EN_SHIFT, 0), +}; + +static const struct snd_soc_dapm_route da732x_dapm_routes[] = { + /* Inputs */ + {"AUX1L PGA", "NULL", "AUX1L"}, + {"AUX1R PGA", "NULL", "AUX1R"}, + {"MIC1 PGA", NULL, "MIC1"}, + {"MIC2 PGA", "NULL", "MIC2"}, + {"MIC3 PGA", "NULL", "MIC3"}, + + /* Capture Path */ + {"ADC1 Left MUX", "MIC1", "MIC1 PGA"}, + {"ADC1 Left MUX", "AUX1L", "AUX1L PGA"}, + + {"ADC1 Right MUX", "AUX1R", "AUX1R PGA"}, + {"ADC1 Right MUX", "MIC2", "MIC2 PGA"}, + {"ADC1 Right MUX", "MIC3", "MIC3 PGA"}, + + {"ADC2 Left MUX", "AUX1L", "AUX1L PGA"}, + {"ADC2 Left MUX", "MIC1", "MIC1 PGA"}, + + {"ADC2 Right MUX", "AUX1R", "AUX1R PGA"}, + {"ADC2 Right MUX", "MIC2", "MIC2 PGA"}, + {"ADC2 Right MUX", "MIC3", "MIC3 PGA"}, + + {"ADC1L", NULL, "ADC1 Supply"}, + {"ADC1R", NULL, "ADC1 Supply"}, + {"ADC2L", NULL, "ADC2 Supply"}, + {"ADC2R", NULL, "ADC2 Supply"}, + + {"ADC1L", NULL, "ADC1 Left MUX"}, + {"ADC1R", NULL, "ADC1 Right MUX"}, + {"ADC2L", NULL, "ADC2 Left MUX"}, + {"ADC2R", NULL, "ADC2 Right MUX"}, + + {"AIFA Output", NULL, "ADC1L"}, + {"AIFA Output", NULL, "ADC1R"}, + {"AIFB Output", NULL, "ADC2L"}, + {"AIFB Output", NULL, "ADC2R"}, + + {"HP Left MUX", "Enabled", "AIFA Input"}, + {"HP Right MUX", "Enabled", "AIFA Input"}, + {"Speaker MUX", "Enabled", "AIFB Input"}, + {"LOUT2 MUX", "Enabled", "AIFB Input"}, + {"LOUT4 MUX", "Enabled", "AIFB Input"}, + + {"DAC1L", NULL, "DAC1 CLK"}, + {"DAC1R", NULL, "DAC1 CLK"}, + {"DAC2L", NULL, "DAC2 CLK"}, + {"DAC2R", NULL, "DAC2 CLK"}, + {"DAC3", NULL, "DAC3 CLK"}, + + {"DAC1L", NULL, "HP Left MUX"}, + {"DAC1R", NULL, "HP Right MUX"}, + {"DAC2L", NULL, "Speaker MUX"}, + {"DAC2R", NULL, "LOUT4 MUX"}, + {"DAC3", NULL, "LOUT2 MUX"}, + + /* Output Pgas */ + {"HP Left", NULL, "DAC1L"}, + {"HP Right", NULL, "DAC1R"}, + {"LIN3", NULL, "DAC2L"}, + {"LIN4", NULL, "DAC2R"}, + {"LIN2", NULL, "DAC3"}, + + /* Outputs */ + {"ClassD", NULL, "LIN3"}, + {"LOUTL", NULL, "LIN2"}, + {"LOUTR", NULL, "LIN4"}, + {"HPL", NULL, "HP Left"}, + {"HPR", NULL, "HP Right"}, +}; + +static int da732x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + u32 aif = 0; + u32 reg_aif; + u32 fs; + + reg_aif = dai->driver->base; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + aif |= DA732X_AIF_WORD_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + aif |= DA732X_AIF_WORD_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + aif |= DA732X_AIF_WORD_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + aif |= DA732X_AIF_WORD_32; + break; + default: + return -EINVAL; + } + + switch (params_rate(params)) { + case 8000: + fs = DA732X_SR_8KHZ; + break; + case 11025: + fs = DA732X_SR_11_025KHZ; + break; + case 12000: + fs = DA732X_SR_12KHZ; + break; + case 16000: + fs = DA732X_SR_16KHZ; + break; + case 22050: + fs = DA732X_SR_22_05KHZ; + break; + case 24000: + fs = DA732X_SR_24KHZ; + break; + case 32000: + fs = DA732X_SR_32KHZ; + break; + case 44100: + fs = DA732X_SR_44_1KHZ; + break; + case 48000: + fs = DA732X_SR_48KHZ; + break; + case 88100: + fs = DA732X_SR_88_1KHZ; + break; + case 96000: + fs = DA732X_SR_96KHZ; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, reg_aif, DA732X_AIF_WORD_MASK, aif); + snd_soc_update_bits(codec, DA732X_REG_CLK_CTRL, DA732X_SR1_MASK, fs); + + return 0; +} + +static int da732x_set_dai_fmt(struct snd_soc_dai *dai, u32 fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u32 aif_mclk, pc_count; + u32 reg_aif1, aif1; + u32 reg_aif3, aif3; + + switch (dai->id) { + case DA732X_DAI_ID1: + reg_aif1 = DA732X_REG_AIFA1; + reg_aif3 = DA732X_REG_AIFA3; + pc_count = DA732X_PC_PULSE_AIFA | DA732X_PC_RESYNC_NOT_AUT | + DA732X_PC_SAME; + break; + case DA732X_DAI_ID2: + reg_aif1 = DA732X_REG_AIFB1; + reg_aif3 = DA732X_REG_AIFB3; + pc_count = DA732X_PC_PULSE_AIFB | DA732X_PC_RESYNC_NOT_AUT | + DA732X_PC_SAME; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + aif1 = DA732X_AIF_SLAVE; + aif_mclk = DA732X_AIFM_FRAME_64 | DA732X_AIFM_SRC_SEL_AIFA; + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif1 = DA732X_AIF_CLK_FROM_SRC; + aif_mclk = DA732X_CLK_GENERATION_AIF_A; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aif3 = DA732X_AIF_I2S_MODE; + break; + case SND_SOC_DAIFMT_RIGHT_J: + aif3 = DA732X_AIF_RIGHT_J_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + aif3 = DA732X_AIF_LEFT_J_MODE; + break; + case SND_SOC_DAIFMT_DSP_B: + aif3 = DA732X_AIF_DSP_MODE; + break; + default: + return -EINVAL; + } + + /* Clock inversion */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_B: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + aif3 |= DA732X_AIF_BCLK_INV; + break; + default: + return -EINVAL; + } + break; + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + aif3 |= DA732X_AIF_BCLK_INV | DA732X_AIF_WCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + aif3 |= DA732X_AIF_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + aif3 |= DA732X_AIF_WCLK_INV; + break; + default: + return -EINVAL; + } + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, DA732X_REG_AIF_MCLK, aif_mclk); + snd_soc_update_bits(codec, reg_aif1, DA732X_AIF1_CLK_MASK, aif1); + snd_soc_update_bits(codec, reg_aif3, DA732X_AIF_BCLK_INV | + DA732X_AIF_WCLK_INV | DA732X_AIF_MODE_MASK, aif3); + snd_soc_write(codec, DA732X_REG_PC_CTRL, pc_count); + + return 0; +} + + + +static int da732x_set_dai_pll(struct snd_soc_codec *codec, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec); + int fref, indiv; + u8 div_lo, div_mid, div_hi; + u64 frac_div; + + /* Disable PLL */ + if (freq_out == 0) { + snd_soc_update_bits(codec, DA732X_REG_PLL_CTRL, + DA732X_PLL_EN, 0); + da732x->pll_en = false; + return 0; + } + + if (da732x->pll_en) + return -EBUSY; + + if (source == DA732X_SRCCLK_MCLK) { + /* Validate Sysclk rate */ + switch (da732x->sysclk) { + case 11290000: + case 12288000: + case 22580000: + case 24576000: + case 45160000: + case 49152000: + snd_soc_write(codec, DA732X_REG_PLL_CTRL, + DA732X_PLL_BYPASS); + return 0; + default: + dev_err(codec->dev, + "Cannot use PLL Bypass, invalid SYSCLK rate\n"); + return -EINVAL; + } + } + + indiv = da732x_get_input_div(codec, da732x->sysclk); + if (indiv < 0) + return indiv; + + fref = (da732x->sysclk / indiv); + div_hi = freq_out / fref; + frac_div = (u64)(freq_out % fref) * 8192ULL; + do_div(frac_div, fref); + div_mid = (frac_div >> DA732X_1BYTE_SHIFT) & DA732X_U8_MASK; + div_lo = (frac_div) & DA732X_U8_MASK; + + snd_soc_write(codec, DA732X_REG_PLL_DIV_LO, div_lo); + snd_soc_write(codec, DA732X_REG_PLL_DIV_MID, div_mid); + snd_soc_write(codec, DA732X_REG_PLL_DIV_HI, div_hi); + + snd_soc_update_bits(codec, DA732X_REG_PLL_CTRL, DA732X_PLL_EN, + DA732X_PLL_EN); + + da732x->pll_en = true; + + return 0; +} + +static int da732x_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec); + + da732x->sysclk = freq; + + return 0; +} + +#define DA732X_RATES SNDRV_PCM_RATE_8000_96000 + +#define DA732X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops da732x_dai1_ops = { + .hw_params = da732x_hw_params, + .set_fmt = da732x_set_dai_fmt, + .set_sysclk = da732x_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops da732x_dai2_ops = { + .hw_params = da732x_hw_params, + .set_fmt = da732x_set_dai_fmt, + .set_sysclk = da732x_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver da732x_dai[] = { + { + .name = "DA732X_AIFA", + .id = DA732X_DAI_ID1, + .base = DA732X_REG_AIFA1, + .playback = { + .stream_name = "AIFA Playback", + .channels_min = 1, + .channels_max = 2, + .rates = DA732X_RATES, + .formats = DA732X_FORMATS, + }, + .capture = { + .stream_name = "AIFA Capture", + .channels_min = 1, + .channels_max = 2, + .rates = DA732X_RATES, + .formats = DA732X_FORMATS, + }, + .ops = &da732x_dai1_ops, + }, + { + .name = "DA732X_AIFB", + .id = DA732X_DAI_ID2, + .base = DA732X_REG_AIFB1, + .playback = { + .stream_name = "AIFB Playback", + .channels_min = 1, + .channels_max = 2, + .rates = DA732X_RATES, + .formats = DA732X_FORMATS, + }, + .capture = { + .stream_name = "AIFB Capture", + .channels_min = 1, + .channels_max = 2, + .rates = DA732X_RATES, + .formats = DA732X_FORMATS, + }, + .ops = &da732x_dai2_ops, + }, +}; + +static const struct regmap_config da732x_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = DA732X_MAX_REG, + .reg_defaults = da732x_reg_cache, + .num_reg_defaults = ARRAY_SIZE(da732x_reg_cache), + .cache_type = REGCACHE_RBTREE, +}; + + +static void da732x_dac_offset_adjust(struct snd_soc_codec *codec) +{ + u8 offset[DA732X_HP_DACS]; + u8 sign[DA732X_HP_DACS]; + u8 step = DA732X_DAC_OFFSET_STEP; + + /* Initialize DAC offset calibration circuits and registers */ + snd_soc_write(codec, DA732X_REG_HPL_DAC_OFFSET, + DA732X_HP_DAC_OFFSET_TRIM_VAL); + snd_soc_write(codec, DA732X_REG_HPR_DAC_OFFSET, + DA732X_HP_DAC_OFFSET_TRIM_VAL); + snd_soc_write(codec, DA732X_REG_HPL_DAC_OFF_CNTL, + DA732X_HP_DAC_OFF_CALIBRATION | + DA732X_HP_DAC_OFF_SCALE_STEPS); + snd_soc_write(codec, DA732X_REG_HPR_DAC_OFF_CNTL, + DA732X_HP_DAC_OFF_CALIBRATION | + DA732X_HP_DAC_OFF_SCALE_STEPS); + + /* Wait for voltage stabilization */ + msleep(DA732X_WAIT_FOR_STABILIZATION); + + /* Check DAC offset sign */ + sign[DA732X_HPL_DAC] = (codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & + DA732X_HP_DAC_OFF_CNTL_COMPO); + sign[DA732X_HPR_DAC] = (codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & + DA732X_HP_DAC_OFF_CNTL_COMPO); + + /* Binary search DAC offset values (both channels at once) */ + offset[DA732X_HPL_DAC] = sign[DA732X_HPL_DAC] << DA732X_HP_DAC_COMPO_SHIFT; + offset[DA732X_HPR_DAC] = sign[DA732X_HPR_DAC] << DA732X_HP_DAC_COMPO_SHIFT; + + do { + offset[DA732X_HPL_DAC] |= step; + offset[DA732X_HPR_DAC] |= step; + snd_soc_write(codec, DA732X_REG_HPL_DAC_OFFSET, + ~offset[DA732X_HPL_DAC] & DA732X_HP_DAC_OFF_MASK); + snd_soc_write(codec, DA732X_REG_HPR_DAC_OFFSET, + ~offset[DA732X_HPR_DAC] & DA732X_HP_DAC_OFF_MASK); + + msleep(DA732X_WAIT_FOR_STABILIZATION); + + if ((codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & + DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPL_DAC]) + offset[DA732X_HPL_DAC] &= ~step; + if ((codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & + DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPR_DAC]) + offset[DA732X_HPR_DAC] &= ~step; + + step >>= 1; + } while (step); + + /* Write final DAC offsets to registers */ + snd_soc_write(codec, DA732X_REG_HPL_DAC_OFFSET, + ~offset[DA732X_HPL_DAC] & DA732X_HP_DAC_OFF_MASK); + snd_soc_write(codec, DA732X_REG_HPR_DAC_OFFSET, + ~offset[DA732X_HPR_DAC] & DA732X_HP_DAC_OFF_MASK); + + /* End DAC calibration mode */ + snd_soc_write(codec, DA732X_REG_HPL_DAC_OFF_CNTL, + DA732X_HP_DAC_OFF_SCALE_STEPS); + snd_soc_write(codec, DA732X_REG_HPR_DAC_OFF_CNTL, + DA732X_HP_DAC_OFF_SCALE_STEPS); +} + +static void da732x_output_offset_adjust(struct snd_soc_codec *codec) +{ + u8 offset[DA732X_HP_AMPS]; + u8 sign[DA732X_HP_AMPS]; + u8 step = DA732X_OUTPUT_OFFSET_STEP; + + offset[DA732X_HPL_AMP] = DA732X_HP_OUT_TRIM_VAL; + offset[DA732X_HPR_AMP] = DA732X_HP_OUT_TRIM_VAL; + + /* Initialize output offset calibration circuits and registers */ + snd_soc_write(codec, DA732X_REG_HPL_OUT_OFFSET, DA732X_HP_OUT_TRIM_VAL); + snd_soc_write(codec, DA732X_REG_HPR_OUT_OFFSET, DA732X_HP_OUT_TRIM_VAL); + snd_soc_write(codec, DA732X_REG_HPL, + DA732X_HP_OUT_COMP | DA732X_HP_OUT_EN); + snd_soc_write(codec, DA732X_REG_HPR, + DA732X_HP_OUT_COMP | DA732X_HP_OUT_EN); + + /* Wait for voltage stabilization */ + msleep(DA732X_WAIT_FOR_STABILIZATION); + + /* Check output offset sign */ + sign[DA732X_HPL_AMP] = codec->hw_read(codec, DA732X_REG_HPL) & + DA732X_HP_OUT_COMPO; + sign[DA732X_HPR_AMP] = codec->hw_read(codec, DA732X_REG_HPR) & + DA732X_HP_OUT_COMPO; + + snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_OUT_COMP | + (sign[DA732X_HPL_AMP] >> DA732X_HP_OUT_COMPO_SHIFT) | + DA732X_HP_OUT_EN); + snd_soc_write(codec, DA732X_REG_HPR, DA732X_HP_OUT_COMP | + (sign[DA732X_HPR_AMP] >> DA732X_HP_OUT_COMPO_SHIFT) | + DA732X_HP_OUT_EN); + + /* Binary search output offset values (both channels at once) */ + do { + offset[DA732X_HPL_AMP] |= step; + offset[DA732X_HPR_AMP] |= step; + snd_soc_write(codec, DA732X_REG_HPL_OUT_OFFSET, + offset[DA732X_HPL_AMP]); + snd_soc_write(codec, DA732X_REG_HPR_OUT_OFFSET, + offset[DA732X_HPR_AMP]); + + msleep(DA732X_WAIT_FOR_STABILIZATION); + + if ((codec->hw_read(codec, DA732X_REG_HPL) & + DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPL_AMP]) + offset[DA732X_HPL_AMP] &= ~step; + if ((codec->hw_read(codec, DA732X_REG_HPR) & + DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPR_AMP]) + offset[DA732X_HPR_AMP] &= ~step; + + step >>= 1; + } while (step); + + /* Write final DAC offsets to registers */ + snd_soc_write(codec, DA732X_REG_HPL_OUT_OFFSET, offset[DA732X_HPL_AMP]); + snd_soc_write(codec, DA732X_REG_HPR_OUT_OFFSET, offset[DA732X_HPR_AMP]); +} + +static void da732x_hp_dc_offset_cancellation(struct snd_soc_codec *codec) +{ + /* Make sure that we have Soft Mute enabled */ + snd_soc_write(codec, DA732X_REG_DAC1_SOFTMUTE, DA732X_SOFTMUTE_EN | + DA732X_GAIN_RAMPED | DA732X_16_SAMPLES); + snd_soc_write(codec, DA732X_REG_DAC1_SEL, DA732X_DACL_EN | + DA732X_DACR_EN | DA732X_DACL_SDM | DA732X_DACR_SDM | + DA732X_DACL_MUTE | DA732X_DACR_MUTE); + snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN | + DA732X_HP_OUT_MUTE | DA732X_HP_OUT_EN); + snd_soc_write(codec, DA732X_REG_HPR, DA732X_HP_OUT_EN | + DA732X_HP_OUT_MUTE | DA732X_HP_OUT_DAC_EN); + + da732x_dac_offset_adjust(codec); + da732x_output_offset_adjust(codec); + + snd_soc_write(codec, DA732X_REG_DAC1_SEL, DA732X_DACS_DIS); + snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_DIS); + snd_soc_write(codec, DA732X_REG_HPR, DA732X_HP_DIS); +} + +static int da732x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, + DA732X_BIAS_BOOST_MASK, + DA732X_BIAS_BOOST_100PC); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + /* Init Codec */ + snd_soc_write(codec, DA732X_REG_REF1, + DA732X_VMID_FASTCHG); + snd_soc_write(codec, DA732X_REG_BIAS_EN, + DA732X_BIAS_EN); + + mdelay(DA732X_STARTUP_DELAY); + + /* Disable Fast Charge and enable DAC ref voltage */ + snd_soc_write(codec, DA732X_REG_REF1, + DA732X_REFBUFX2_EN); + + /* Enable bypass DSP routing */ + snd_soc_write(codec, DA732X_REG_DATA_ROUTE, + DA732X_BYPASS_DSP); + + /* Enable Digital subsystem */ + snd_soc_write(codec, DA732X_REG_DSP_CTRL, + DA732X_DIGITAL_EN); + + snd_soc_write(codec, DA732X_REG_SPARE1_OUT, + DA732X_HP_DRIVER_EN | + DA732X_HP_GATE_LOW | + DA732X_HP_LOOP_GAIN_CTRL); + snd_soc_write(codec, DA732X_REG_HP_LIN1_GNDSEL, + DA732X_HP_OUT_GNDSEL); + + da732x_set_charge_pump(codec, DA732X_ENABLE_CP); + + snd_soc_write(codec, DA732X_REG_CLK_EN1, + DA732X_SYS3_CLK_EN | DA732X_PC_CLK_EN); + + /* Enable Zero Crossing */ + snd_soc_write(codec, DA732X_REG_INP_ZC_EN, + DA732X_MIC1_PRE_ZC_EN | + DA732X_MIC1_ZC_EN | + DA732X_MIC2_PRE_ZC_EN | + DA732X_MIC2_ZC_EN | + DA732X_AUXL_ZC_EN | + DA732X_AUXR_ZC_EN | + DA732X_MIC3_PRE_ZC_EN | + DA732X_MIC3_ZC_EN); + snd_soc_write(codec, DA732X_REG_OUT_ZC_EN, + DA732X_HPL_ZC_EN | DA732X_HPR_ZC_EN | + DA732X_LIN2_ZC_EN | DA732X_LIN3_ZC_EN | + DA732X_LIN4_ZC_EN); + + da732x_hp_dc_offset_cancellation(codec); + + regcache_cache_only(codec->control_data, false); + regcache_sync(codec->control_data); + } else { + snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, + DA732X_BIAS_BOOST_MASK, + DA732X_BIAS_BOOST_50PC); + snd_soc_update_bits(codec, DA732X_REG_PLL_CTRL, + DA732X_PLL_EN, 0); + da732x->pll_en = false; + } + break; + case SND_SOC_BIAS_OFF: + regcache_cache_only(codec->control_data, true); + da732x_set_charge_pump(codec, DA732X_DISABLE_CP); + snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, DA732X_BIAS_EN, + DA732X_BIAS_DIS); + da732x->pll_en = false; + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +static int da732x_probe(struct snd_soc_codec *codec) +{ + struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret = 0; + + da732x->codec = codec; + + dapm->idle_bias_off = false; + + codec->control_data = da732x->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec.\n"); + goto err; + } + + da732x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); +err: + return ret; +} + +static int da732x_remove(struct snd_soc_codec *codec) +{ + + da732x_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_da732x = { + .probe = da732x_probe, + .remove = da732x_remove, + .set_bias_level = da732x_set_bias_level, + .controls = da732x_snd_controls, + .num_controls = ARRAY_SIZE(da732x_snd_controls), + .dapm_widgets = da732x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da732x_dapm_widgets), + .dapm_routes = da732x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(da732x_dapm_routes), + .set_pll = da732x_set_dai_pll, + .reg_cache_size = ARRAY_SIZE(da732x_reg_cache), +}; + +static __devinit int da732x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da732x_priv *da732x; + unsigned int reg; + int ret; + + da732x = devm_kzalloc(&i2c->dev, sizeof(struct da732x_priv), + GFP_KERNEL); + if (!da732x) + return -ENOMEM; + + i2c_set_clientdata(i2c, da732x); + + da732x->regmap = devm_regmap_init_i2c(i2c, &da732x_regmap); + if (IS_ERR(da732x->regmap)) { + ret = PTR_ERR(da732x->regmap); + dev_err(&i2c->dev, "Failed to initialize regmap\n"); + goto err; + } + + ret = regmap_read(da732x->regmap, DA732X_REG_ID, ®); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret); + goto err; + } + + dev_info(&i2c->dev, "Revision: %d.%d\n", + (reg & DA732X_ID_MAJOR_MASK), (reg & DA732X_ID_MINOR_MASK)); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da732x, + da732x_dai, ARRAY_SIZE(da732x_dai)); + if (ret != 0) + dev_err(&i2c->dev, "Failed to register codec.\n"); + +err: + return ret; +} + +static __devexit int da732x_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id da732x_i2c_id[] = { + { "da7320", 0}, + { } +}; +MODULE_DEVICE_TABLE(i2c, da732x_i2c_id); + +static struct i2c_driver da732x_i2c_driver = { + .driver = { + .name = "da7320", + .owner = THIS_MODULE, + }, + .probe = da732x_i2c_probe, + .remove = __devexit_p(da732x_i2c_remove), + .id_table = da732x_i2c_id, +}; + +module_i2c_driver(da732x_i2c_driver); + + +MODULE_DESCRIPTION("ASoC DA732X driver"); +MODULE_AUTHOR("Michal Hajduk <michal.hajduk@diasemi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h new file mode 100644 index 000000000000..c8ce5475de22 --- /dev/null +++ b/sound/soc/codecs/da732x.h @@ -0,0 +1,133 @@ +/* + * da732x.h -- Dialog DA732X ALSA SoC Audio Driver Header File + * + * Copyright (C) 2012 Dialog Semiconductor GmbH + * + * Author: Michal Hajduk <Michal.Hajduk@diasemi.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __DA732X_H_ +#define __DA732X_H + +#include <sound/soc.h> + +/* General */ +#define DA732X_U8_MASK 0xFF +#define DA732X_4BYTES 4 +#define DA732X_3BYTES 3 +#define DA732X_2BYTES 2 +#define DA732X_1BYTE 1 +#define DA732X_1BYTE_SHIFT 8 +#define DA732X_2BYTES_SHIFT 16 +#define DA732X_3BYTES_SHIFT 24 +#define DA732X_4BYTES_SHIFT 32 + +#define DA732X_DACS_DIS 0x0 +#define DA732X_HP_DIS 0x0 +#define DA732X_CLEAR_REG 0x0 + +/* Calibration */ +#define DA732X_DAC_OFFSET_STEP 0x20 +#define DA732X_OUTPUT_OFFSET_STEP 0x80 +#define DA732X_HP_OUT_TRIM_VAL 0x0 +#define DA732X_WAIT_FOR_STABILIZATION 1 +#define DA732X_HPL_DAC 0 +#define DA732X_HPR_DAC 1 +#define DA732X_HP_DACS 2 +#define DA732X_HPL_AMP 0 +#define DA732X_HPR_AMP 1 +#define DA732X_HP_AMPS 2 + +/* Clock settings */ +#define DA732X_STARTUP_DELAY 100 +#define DA732X_PLL_OUT_196608 196608000 +#define DA732X_PLL_OUT_180634 180633600 +#define DA732X_PLL_OUT_SRM 188620800 +#define DA732X_MCLK_10MHZ 10000000 +#define DA732X_MCLK_20MHZ 20000000 +#define DA732X_MCLK_40MHZ 40000000 +#define DA732X_MCLK_54MHZ 54000000 +#define DA732X_MCLK_RET_0_10MHZ 0 +#define DA732X_MCLK_VAL_0_10MHZ 1 +#define DA732X_MCLK_RET_10_20MHZ 1 +#define DA732X_MCLK_VAL_10_20MHZ 2 +#define DA732X_MCLK_RET_20_40MHZ 2 +#define DA732X_MCLK_VAL_20_40MHZ 4 +#define DA732X_MCLK_RET_40_54MHZ 3 +#define DA732X_MCLK_VAL_40_54MHZ 8 +#define DA732X_DAI_ID1 0 +#define DA732X_DAI_ID2 1 +#define DA732X_SRCCLK_PLL 0 +#define DA732X_SRCCLK_MCLK 1 + +#define DA732X_LIN_LP_VOL 0x4F +#define DA732X_LP_VOL 0x40 + +/* Kcontrols */ +#define DA732X_DAC_EN_MAX 2 +#define DA732X_ADCL_MUX_MAX 2 +#define DA732X_ADCR_MUX_MAX 3 +#define DA732X_HPF_MODE_MAX 3 +#define DA732X_HPF_MODE_SHIFT 4 +#define DA732X_HPF_MUSIC_SHIFT 0 +#define DA732X_HPF_MUSIC_MAX 4 +#define DA732X_HPF_VOICE_SHIFT 4 +#define DA732X_HPF_VOICE_MAX 8 +#define DA732X_EQ_EN_MAX 1 +#define DA732X_HPF_VOICE 1 +#define DA732X_HPF_MUSIC 2 +#define DA732X_HPF_DISABLED 0 +#define DA732X_NO_INVERT 0 +#define DA732X_INVERT 1 +#define DA732X_SWITCH_MAX 1 +#define DA732X_ENABLE_CP 1 +#define DA732X_DISABLE_CP 0 +#define DA732X_DISABLE_ALL_CLKS 0 +#define DA732X_RESET_ADCS 0 + +/* dB values */ +#define DA732X_MIC_VOL_DB_MIN 0 +#define DA732X_MIC_VOL_DB_INC 50 +#define DA732X_MIC_PRE_VOL_DB_MIN 0 +#define DA732X_MIC_PRE_VOL_DB_INC 600 +#define DA732X_AUX_VOL_DB_MIN -6000 +#define DA732X_AUX_VOL_DB_INC 150 +#define DA732X_HP_VOL_DB_MIN -2250 +#define DA732X_HP_VOL_DB_INC 150 +#define DA732X_LIN2_VOL_DB_MIN -1650 +#define DA732X_LIN2_VOL_DB_INC 150 +#define DA732X_LIN3_VOL_DB_MIN -1650 +#define DA732X_LIN3_VOL_DB_INC 150 +#define DA732X_LIN4_VOL_DB_MIN -2250 +#define DA732X_LIN4_VOL_DB_INC 150 +#define DA732X_EQ_BAND_VOL_DB_MIN -1050 +#define DA732X_EQ_BAND_VOL_DB_INC 150 +#define DA732X_DAC_VOL_DB_MIN -7725 +#define DA732X_DAC_VOL_DB_INC 75 +#define DA732X_ADC_VOL_DB_MIN 0 +#define DA732X_ADC_VOL_DB_INC -1 +#define DA732X_EQ_OVERALL_VOL_DB_MIN -1800 +#define DA732X_EQ_OVERALL_VOL_DB_INC 600 + +#define DA732X_SOC_ENUM_DOUBLE_R(xreg, xrreg, xmax, xtext) \ + {.reg = xreg, .reg2 = xrreg, .max = xmax, .texts = xtext} + +enum da732x_sysctl { + DA732X_SR_8KHZ = 0x1, + DA732X_SR_11_025KHZ = 0x2, + DA732X_SR_12KHZ = 0x3, + DA732X_SR_16KHZ = 0x5, + DA732X_SR_22_05KHZ = 0x6, + DA732X_SR_24KHZ = 0x7, + DA732X_SR_32KHZ = 0x9, + DA732X_SR_44_1KHZ = 0xA, + DA732X_SR_48KHZ = 0xB, + DA732X_SR_88_1KHZ = 0xE, + DA732X_SR_96KHZ = 0xF, +}; + +#endif /* __DA732X_H_ */ diff --git a/sound/soc/codecs/da732x_reg.h b/sound/soc/codecs/da732x_reg.h new file mode 100644 index 000000000000..bdd03ca4b2de --- /dev/null +++ b/sound/soc/codecs/da732x_reg.h @@ -0,0 +1,654 @@ +/* + * da732x_reg.h --- Dialog DA732X ALSA SoC Audio Registers Header File + * + * Copyright (C) 2012 Dialog Semiconductor GmbH + * + * Author: Michal Hajduk <Michal.Hajduk@diasemi.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __DA732X_REG_H_ +#define __DA732X_REG_H_ + +/* DA732X registers */ +#define DA732X_REG_STATUS_EXT 0x00 +#define DA732X_REG_STATUS 0x01 +#define DA732X_REG_REF1 0x02 +#define DA732X_REG_BIAS_EN 0x03 +#define DA732X_REG_BIAS1 0x04 +#define DA732X_REG_BIAS2 0x05 +#define DA732X_REG_BIAS3 0x06 +#define DA732X_REG_BIAS4 0x07 +#define DA732X_REG_MICBIAS2 0x0F +#define DA732X_REG_MICBIAS1 0x10 +#define DA732X_REG_MICDET 0x11 +#define DA732X_REG_MIC1_PRE 0x12 +#define DA732X_REG_MIC1 0x13 +#define DA732X_REG_MIC2_PRE 0x14 +#define DA732X_REG_MIC2 0x15 +#define DA732X_REG_AUX1L 0x16 +#define DA732X_REG_AUX1R 0x17 +#define DA732X_REG_MIC3_PRE 0x18 +#define DA732X_REG_MIC3 0x19 +#define DA732X_REG_INP_PINBIAS 0x1A +#define DA732X_REG_INP_ZC_EN 0x1B +#define DA732X_REG_INP_MUX 0x1D +#define DA732X_REG_HP_DET 0x20 +#define DA732X_REG_HPL_DAC_OFFSET 0x21 +#define DA732X_REG_HPL_DAC_OFF_CNTL 0x22 +#define DA732X_REG_HPL_OUT_OFFSET 0x23 +#define DA732X_REG_HPL 0x24 +#define DA732X_REG_HPL_VOL 0x25 +#define DA732X_REG_HPR_DAC_OFFSET 0x26 +#define DA732X_REG_HPR_DAC_OFF_CNTL 0x27 +#define DA732X_REG_HPR_OUT_OFFSET 0x28 +#define DA732X_REG_HPR 0x29 +#define DA732X_REG_HPR_VOL 0x2A +#define DA732X_REG_LIN2 0x2B +#define DA732X_REG_LIN3 0x2C +#define DA732X_REG_LIN4 0x2D +#define DA732X_REG_OUT_ZC_EN 0x2E +#define DA732X_REG_HP_LIN1_GNDSEL 0x37 +#define DA732X_REG_CP_HP1 0x3A +#define DA732X_REG_CP_HP2 0x3B +#define DA732X_REG_CP_CTRL1 0x40 +#define DA732X_REG_CP_CTRL2 0x41 +#define DA732X_REG_CP_CTRL3 0x42 +#define DA732X_REG_CP_LEVEL_MASK 0x43 +#define DA732X_REG_CP_DET 0x44 +#define DA732X_REG_CP_STATUS 0x45 +#define DA732X_REG_CP_THRESH1 0x46 +#define DA732X_REG_CP_THRESH2 0x47 +#define DA732X_REG_CP_THRESH3 0x48 +#define DA732X_REG_CP_THRESH4 0x49 +#define DA732X_REG_CP_THRESH5 0x4A +#define DA732X_REG_CP_THRESH6 0x4B +#define DA732X_REG_CP_THRESH7 0x4C +#define DA732X_REG_CP_THRESH8 0x4D +#define DA732X_REG_PLL_DIV_LO 0x50 +#define DA732X_REG_PLL_DIV_MID 0x51 +#define DA732X_REG_PLL_DIV_HI 0x52 +#define DA732X_REG_PLL_CTRL 0x53 +#define DA732X_REG_CLK_CTRL 0x54 +#define DA732X_REG_CLK_DSP 0x5A +#define DA732X_REG_CLK_EN1 0x5B +#define DA732X_REG_CLK_EN2 0x5C +#define DA732X_REG_CLK_EN3 0x5D +#define DA732X_REG_CLK_EN4 0x5E +#define DA732X_REG_CLK_EN5 0x5F +#define DA732X_REG_AIF_MCLK 0x60 +#define DA732X_REG_AIFA1 0x61 +#define DA732X_REG_AIFA2 0x62 +#define DA732X_REG_AIFA3 0x63 +#define DA732X_REG_AIFB1 0x64 +#define DA732X_REG_AIFB2 0x65 +#define DA732X_REG_AIFB3 0x66 +#define DA732X_REG_PC_CTRL 0x6A +#define DA732X_REG_DATA_ROUTE 0x70 +#define DA732X_REG_DSP_CTRL 0x71 +#define DA732X_REG_CIF_CTRL2 0x74 +#define DA732X_REG_HANDSHAKE 0x75 +#define DA732X_REG_MBOX0 0x76 +#define DA732X_REG_MBOX1 0x77 +#define DA732X_REG_MBOX2 0x78 +#define DA732X_REG_MBOX_STATUS 0x79 +#define DA732X_REG_SPARE1_OUT 0x7D +#define DA732X_REG_SPARE2_OUT 0x7E +#define DA732X_REG_SPARE1_IN 0x7F +#define DA732X_REG_ID 0x81 +#define DA732X_REG_ADC1_PD 0x90 +#define DA732X_REG_ADC1_HPF 0x93 +#define DA732X_REG_ADC1_SEL 0x94 +#define DA732X_REG_ADC1_EQ12 0x95 +#define DA732X_REG_ADC1_EQ34 0x96 +#define DA732X_REG_ADC1_EQ5 0x97 +#define DA732X_REG_ADC2_PD 0x98 +#define DA732X_REG_ADC2_HPF 0x9B +#define DA732X_REG_ADC2_SEL 0x9C +#define DA732X_REG_ADC2_EQ12 0x9D +#define DA732X_REG_ADC2_EQ34 0x9E +#define DA732X_REG_ADC2_EQ5 0x9F +#define DA732X_REG_DAC1_HPF 0xA0 +#define DA732X_REG_DAC1_L_VOL 0xA1 +#define DA732X_REG_DAC1_R_VOL 0xA2 +#define DA732X_REG_DAC1_SEL 0xA3 +#define DA732X_REG_DAC1_SOFTMUTE 0xA4 +#define DA732X_REG_DAC1_EQ12 0xA5 +#define DA732X_REG_DAC1_EQ34 0xA6 +#define DA732X_REG_DAC1_EQ5 0xA7 +#define DA732X_REG_DAC2_HPF 0xB0 +#define DA732X_REG_DAC2_L_VOL 0xB1 +#define DA732X_REG_DAC2_R_VOL 0xB2 +#define DA732X_REG_DAC2_SEL 0xB3 +#define DA732X_REG_DAC2_SOFTMUTE 0xB4 +#define DA732X_REG_DAC2_EQ12 0xB5 +#define DA732X_REG_DAC2_EQ34 0xB6 +#define DA732X_REG_DAC2_EQ5 0xB7 +#define DA732X_REG_DAC3_HPF 0xC0 +#define DA732X_REG_DAC3_VOL 0xC1 +#define DA732X_REG_DAC3_SEL 0xC3 +#define DA732X_REG_DAC3_SOFTMUTE 0xC4 +#define DA732X_REG_DAC3_EQ12 0xC5 +#define DA732X_REG_DAC3_EQ34 0xC6 +#define DA732X_REG_DAC3_EQ5 0xC7 +#define DA732X_REG_BIQ_BYP 0xD2 +#define DA732X_REG_DMA_CMD 0xD3 +#define DA732X_REG_DMA_ADDR0 0xD4 +#define DA732X_REG_DMA_ADDR1 0xD5 +#define DA732X_REG_DMA_DATA0 0xD6 +#define DA732X_REG_DMA_DATA1 0xD7 +#define DA732X_REG_DMA_DATA2 0xD8 +#define DA732X_REG_DMA_DATA3 0xD9 +#define DA732X_REG_DMA_STATUS 0xDA +#define DA732X_REG_BROWNOUT 0xDF +#define DA732X_REG_UNLOCK 0xE0 + +#define DA732X_MAX_REG DA732X_REG_UNLOCK +/* + * Bits + */ + +/* DA732X_REG_STATUS_EXT (addr=0x00) */ +#define DA732X_STATUS_EXT_DSP (1 << 4) +#define DA732X_STATUS_EXT_CLEAR (0 << 0) + +/* DA732X_REG_STATUS (addr=0x01) */ +#define DA732X_STATUS_PLL_LOCK (1 << 0) +#define DA732X_STATUS_PLL_MCLK_DET (1 << 1) +#define DA732X_STATUS_HPDET_OUT (1 << 2) +#define DA732X_STATUS_INP_MIXDET_1 (1 << 3) +#define DA732X_STATUS_INP_MIXDET_2 (1 << 4) +#define DA732X_STATUS_BO_STATUS (1 << 5) + +/* DA732X_REG_REF1 (addr=0x02) */ +#define DA732X_VMID_FASTCHG (1 << 1) +#define DA732X_VMID_FASTDISCHG (1 << 2) +#define DA732X_REFBUFX2_EN (1 << 6) +#define DA732X_REFBUFX2_DIS (0 << 6) + +/* DA732X_REG_BIAS_EN (addr=0x03) */ +#define DA732X_BIAS_BOOST_MASK (3 << 0) +#define DA732X_BIAS_BOOST_100PC (0 << 0) +#define DA732X_BIAS_BOOST_133PC (1 << 0) +#define DA732X_BIAS_BOOST_88PC (2 << 0) +#define DA732X_BIAS_BOOST_50PC (3 << 0) +#define DA732X_BIAS_EN (1 << 7) +#define DA732X_BIAS_DIS (0 << 7) + +/* DA732X_REG_BIAS1 (addr=0x04) */ +#define DA732X_BIAS1_HP_DAC_BIAS_MASK (3 << 0) +#define DA732X_BIAS1_HP_DAC_BIAS_100PC (0 << 0) +#define DA732X_BIAS1_HP_DAC_BIAS_150PC (1 << 0) +#define DA732X_BIAS1_HP_DAC_BIAS_50PC (2 << 0) +#define DA732X_BIAS1_HP_DAC_BIAS_75PC (3 << 0) +#define DA732X_BIAS1_HP_OUT_BIAS_MASK (7 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_100PC (0 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_125PC (1 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_150PC (2 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_175PC (3 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_200PC (4 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_250PC (5 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_300PC (6 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_350PC (7 << 4) + +/* DA732X_REG_BIAS2 (addr=0x05) */ +#define DA732X_BIAS2_LINE2_DAC_BIAS_MASK (3 << 0) +#define DA732X_BIAS2_LINE2_DAC_BIAS_100PC (0 << 0) +#define DA732X_BIAS2_LINE2_DAC_BIAS_150PC (1 << 0) +#define DA732X_BIAS2_LINE2_DAC_BIAS_50PC (2 << 0) +#define DA732X_BIAS2_LINE2_DAC_BIAS_75PC (3 << 0) +#define DA732X_BIAS2_LINE2_OUT_BIAS_MASK (7 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_100PC (0 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_125PC (1 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_150PC (2 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_175PC (3 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_200PC (4 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_250PC (5 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_300PC (6 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_350PC (7 << 4) + +/* DA732X_REG_BIAS3 (addr=0x06) */ +#define DA732X_BIAS3_LINE3_DAC_BIAS_MASK (3 << 0) +#define DA732X_BIAS3_LINE3_DAC_BIAS_100PC (0 << 0) +#define DA732X_BIAS3_LINE3_DAC_BIAS_150PC (1 << 0) +#define DA732X_BIAS3_LINE3_DAC_BIAS_50PC (2 << 0) +#define DA732X_BIAS3_LINE3_DAC_BIAS_75PC (3 << 0) +#define DA732X_BIAS3_LINE3_OUT_BIAS_MASK (7 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_100PC (0 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_125PC (1 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_150PC (2 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_175PC (3 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_200PC (4 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_250PC (5 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_300PC (6 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_350PC (7 << 4) + +/* DA732X_REG_BIAS4 (addr=0x07) */ +#define DA732X_BIAS4_LINE4_DAC_BIAS_MASK (3 << 0) +#define DA732X_BIAS4_LINE4_DAC_BIAS_100PC (0 << 0) +#define DA732X_BIAS4_LINE4_DAC_BIAS_150PC (1 << 0) +#define DA732X_BIAS4_LINE4_DAC_BIAS_50PC (2 << 0) +#define DA732X_BIAS4_LINE4_DAC_BIAS_75PC (3 << 0) +#define DA732X_BIAS4_LINE4_OUT_BIAS_MASK (7 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_100PC (0 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_125PC (1 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_150PC (2 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_175PC (3 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_200PC (4 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_250PC (5 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_300PC (6 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_350PC (7 << 4) + +/* DA732X_REG_SIF_VDD_SEL (addr=0x08) */ +#define DA732X_SIF_VDD_SEL_AIFA_VDD2 (1 << 0) +#define DA732X_SIF_VDD_SEL_AIFB_VDD2 (1 << 1) +#define DA732X_SIF_VDD_SEL_CIFA_VDD2 (1 << 4) + +/* DA732X_REG_MICBIAS2/1 (addr=0x0F/0x10) */ +#define DA732X_MICBIAS_VOLTAGE_MASK (0x0F << 0) +#define DA732X_MICBIAS_VOLTAGE_2V (0x00 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V05 (0x01 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V1 (0x02 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V15 (0x03 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V2 (0x04 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V25 (0x05 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V3 (0x06 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V35 (0x07 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V4 (0x08 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V45 (0x09 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V5 (0x0A << 0) +#define DA732X_MICBIAS_EN (1 << 7) +#define DA732X_MICBIAS_EN_SHIFT 7 +#define DA732X_MICBIAS_VOLTAGE_SHIFT 0 +#define DA732X_MICBIAS_VOLTAGE_MAX 0x0B + +/* DA732X_REG_MICDET (addr=0x11) */ +#define DA732X_MICDET_INP_MICRES (1 << 0) +#define DA732X_MICDET_INP_MICHOOK (1 << 1) +#define DA732X_MICDET_INP_DEBOUNCE_PRD_8MS (0 << 0) +#define DA732X_MICDET_INP_DEBOUNCE_PRD_16MS (1 << 0) +#define DA732X_MICDET_INP_DEBOUNCE_PRD_32MS (2 << 0) +#define DA732X_MICDET_INP_DEBOUNCE_PRD_64MS (3 << 0) +#define DA732X_MICDET_INP_MICDET_EN (1 << 7) + +/* DA732X_REG_MIC1/2/3_PRE (addr=0x11/0x14/0x18) */ +#define DA732X_MICBOOST_MASK 0x7 +#define DA732X_MICBOOST_SHIFT 0 +#define DA732X_MICBOOST_MIN 0x1 +#define DA732X_MICBOOST_MAX DA732X_MICBOOST_MASK + +/* DA732X_REG_MIC1/2/3 (addr=0x13/0x15/0x19) */ +#define DA732X_MIC_VOL_SHIFT 0 +#define DA732X_MIC_VOL_VAL_MASK 0x1F +#define DA732X_MIC_MUTE_SHIFT 6 +#define DA732X_MIC_EN_SHIFT 7 +#define DA732X_MIC_VOL_VAL_MIN 0x7 +#define DA732X_MIC_VOL_VAL_MAX DA732X_MIC_VOL_VAL_MASK + +/* DA732X_REG_AUX1L/R (addr=0x16/0x17) */ +#define DA732X_AUX_VOL_SHIFT 0 +#define DA732X_AUX_VOL_MASK 0x7 +#define DA732X_AUX_MUTE_SHIFT 6 +#define DA732X_AUX_EN_SHIFT 7 +#define DA732X_AUX_VOL_VAL_MAX DA732X_AUX_VOL_MASK + +/* DA732X_REG_INP_PINBIAS (addr=0x1A) */ +#define DA732X_INP_MICL_PINBIAS_EN (1 << 0) +#define DA732X_INP_MICR_PINBIAS_EN (1 << 1) +#define DA732X_INP_AUX1L_PINBIAS_EN (1 << 2) +#define DA732X_INP_AUX1R_PINBIAS_EN (1 << 3) +#define DA732X_INP_AUX2_PINBIAS_EN (1 << 4) + +/* DA732X_REG_INP_ZC_EN (addr=0x1B) */ +#define DA732X_MIC1_PRE_ZC_EN (1 << 0) +#define DA732X_MIC1_ZC_EN (1 << 1) +#define DA732X_MIC2_PRE_ZC_EN (1 << 2) +#define DA732X_MIC2_ZC_EN (1 << 3) +#define DA732X_AUXL_ZC_EN (1 << 4) +#define DA732X_AUXR_ZC_EN (1 << 5) +#define DA732X_MIC3_PRE_ZC_EN (1 << 6) +#define DA732X_MIC3_ZC_EN (1 << 7) + +/* DA732X_REG_INP_MUX (addr=0x1D) */ +#define DA732X_INP_ADC1L_MUX_SEL_AUX1L (0 << 0) +#define DA732X_INP_ADC1L_MUX_SEL_MIC1 (1 << 0) +#define DA732X_INP_ADC1R_MUX_SEL_MASK (3 << 2) +#define DA732X_INP_ADC1R_MUX_SEL_AUX1R (0 << 2) +#define DA732X_INP_ADC1R_MUX_SEL_MIC2 (1 << 2) +#define DA732X_INP_ADC1R_MUX_SEL_MIC3 (2 << 2) +#define DA732X_INP_ADC2L_MUX_SEL_AUX1L (0 << 4) +#define DA732X_INP_ADC2L_MUX_SEL_MICL (1 << 4) +#define DA732X_INP_ADC2R_MUX_SEL_MASK (3 << 6) +#define DA732X_INP_ADC2R_MUX_SEL_AUX1R (0 << 6) +#define DA732X_INP_ADC2R_MUX_SEL_MICR (1 << 6) +#define DA732X_INP_ADC2R_MUX_SEL_AUX2 (2 << 6) +#define DA732X_ADC1L_MUX_SEL_SHIFT 0 +#define DA732X_ADC1R_MUX_SEL_SHIFT 2 +#define DA732X_ADC2L_MUX_SEL_SHIFT 4 +#define DA732X_ADC2R_MUX_SEL_SHIFT 6 + +/* DA732X_REG_HP_DET (addr=0x20) */ +#define DA732X_HP_DET_AZ (1 << 0) +#define DA732X_HP_DET_SEL1 (1 << 1) +#define DA732X_HP_DET_IS_MASK (3 << 2) +#define DA732X_HP_DET_IS_0_5UA (0 << 2) +#define DA732X_HP_DET_IS_1UA (1 << 2) +#define DA732X_HP_DET_IS_2UA (2 << 2) +#define DA732X_HP_DET_IS_4UA (3 << 2) +#define DA732X_HP_DET_RS_MASK (3 << 4) +#define DA732X_HP_DET_RS_INFINITE (0 << 4) +#define DA732X_HP_DET_RS_100KOHM (1 << 4) +#define DA732X_HP_DET_RS_10KOHM (2 << 4) +#define DA732X_HP_DET_RS_1KOHM (3 << 4) +#define DA732X_HP_DET_EN (1 << 7) + +/* DA732X_REG_HPL_DAC_OFFSET (addr=0x21/0x26) */ +#define DA732X_HP_DAC_OFFSET_TRIM_MASK (0x3F << 0) +#define DA732X_HP_DAC_OFFSET_DAC_SIGN (1 << 6) + +/* DA732X_REG_HPL_DAC_OFF_CNTL (addr=0x22/0x27) */ +#define DA732X_HP_DAC_OFF_CNTL_CONT_MASK (7 << 0) +#define DA732X_HP_DAC_OFF_CNTL_COMPO (1 << 3) +#define DA732X_HP_DAC_OFF_CALIBRATION (1 << 0) +#define DA732X_HP_DAC_OFF_SCALE_STEPS (1 << 1) +#define DA732X_HP_DAC_OFF_MASK 0x7F +#define DA732X_HP_DAC_COMPO_SHIFT 3 + +/* DA732X_REG_HPL_OUT_OFFSET (addr=0x23/0x28) */ +#define DA732X_HP_OUT_OFFSET_MASK (0xFF << 0) +#define DA732X_HP_DAC_OFFSET_TRIM_VAL 0x7F + +/* DA732X_REG_HPL/R (addr=0x24/0x29) */ +#define DA732X_HP_OUT_SIGN (1 << 0) +#define DA732X_HP_OUT_COMP (1 << 1) +#define DA732X_HP_OUT_RESERVED (1 << 2) +#define DA732X_HP_OUT_COMPO (1 << 3) +#define DA732X_HP_OUT_DAC_EN (1 << 4) +#define DA732X_HP_OUT_HIZ_EN (1 << 5) +#define DA732X_HP_OUT_HIZ_DIS (0 << 5) +#define DA732X_HP_OUT_MUTE (1 << 6) +#define DA732X_HP_OUT_EN (1 << 7) +#define DA732X_HP_OUT_COMPO_SHIFT 3 +#define DA732X_HP_OUT_DAC_EN_SHIFT 4 +#define DA732X_HP_HIZ_SHIFT 5 +#define DA732X_HP_MUTE_SHIFT 6 +#define DA732X_HP_OUT_EN_SHIFT 7 + +#define DA732X_OUT_HIZ_EN (1 << 5) +#define DA732X_OUT_HIZ_DIS (0 << 5) + +/* DA732X_REG_HPL/R_VOL (addr=0x25/0x2A) */ +#define DA732X_HP_VOL_VAL_MASK 0xF +#define DA732X_HP_VOL_SHIFT 0 +#define DA732X_HP_VOL_VAL_MAX DA732X_HP_VOL_VAL_MASK + +/* DA732X_REG_LIN2/3/4 (addr=0x2B/0x2C/0x2D) */ +#define DA732X_LOUT_VOL_SHIFT 0 +#define DA732X_LOUT_VOL_MASK 0x0F +#define DA732X_LOUT_DAC_OFF (0 << 4) +#define DA732X_LOUT_DAC_EN (1 << 4) +#define DA732X_LOUT_HIZ_N_DIS (0 << 5) +#define DA732X_LOUT_HIZ_N_EN (1 << 5) +#define DA732X_LOUT_UNMUTED (0 << 6) +#define DA732X_LOUT_MUTED (1 << 6) +#define DA732X_LOUT_EN (0 << 7) +#define DA732X_LOUT_DIS (1 << 7) +#define DA732X_LOUT_DAC_EN_SHIFT 4 +#define DA732X_LOUT_MUTE_SHIFT 6 +#define DA732X_LIN_OUT_EN_SHIFT 7 +#define DA732X_LOUT_VOL_VAL_MAX DA732X_LOUT_VOL_MASK + +/* DA732X_REG_OUT_ZC_EN (addr=0x2E) */ +#define DA732X_HPL_ZC_EN_SHIFT 0 +#define DA732X_HPR_ZC_EN_SHIFT 1 +#define DA732X_HPL_ZC_EN (1 << 0) +#define DA732X_HPL_ZC_DIS (0 << 0) +#define DA732X_HPR_ZC_EN (1 << 1) +#define DA732X_HPR_ZC_DIS (0 << 1) +#define DA732X_LIN2_ZC_EN (1 << 2) +#define DA732X_LIN2_ZC_DIS (0 << 2) +#define DA732X_LIN3_ZC_EN (1 << 3) +#define DA732X_LIN3_ZC_DIS (0 << 3) +#define DA732X_LIN4_ZC_EN (1 << 4) +#define DA732X_LIN4_ZC_DIS (0 << 4) + +/* DA732X_REG_HP_LIN1_GNDSEL (addr=0x37) */ +#define DA732X_HP_OUT_GNDSEL (1 << 0) + +/* DA732X_REG_CP_HP2 (addr=0x3a) */ +#define DA732X_HP_CP_PULSESKIP (1 << 0) +#define DA732X_HP_CP_REG (1 << 1) +#define DA732X_HP_CP_EN (1 << 3) +#define DA732X_HP_CP_DIS (0 << 3) + +/* DA732X_REG_CP_CTRL1 (addr=0x40) */ +#define DA732X_CP_MODE_MASK (7 << 1) +#define DA732X_CP_CTRL_STANDBY (0 << 1) +#define DA732X_CP_CTRL_CPVDD6 (2 << 1) +#define DA732X_CP_CTRL_CPVDD5 (3 << 1) +#define DA732X_CP_CTRL_CPVDD4 (4 << 1) +#define DA732X_CP_CTRL_CPVDD3 (5 << 1) +#define DA732X_CP_CTRL_CPVDD2 (6 << 1) +#define DA732X_CP_CTRL_CPVDD1 (7 << 1) +#define DA723X_CP_DIS (0 << 7) +#define DA732X_CP_EN (1 << 7) + +/* DA732X_REG_CP_CTRL2 (addr=0x41) */ +#define DA732X_CP_BOOST (1 << 0) +#define DA732X_CP_MANAGE_MAGNITUDE (2 << 2) + +/* DA732X_REG_CP_CTRL3 (addr=0x42) */ +#define DA732X_CP_1MHZ (0 << 0) +#define DA732X_CP_500KHZ (1 << 0) +#define DA732X_CP_250KHZ (2 << 0) +#define DA732X_CP_125KHZ (3 << 0) +#define DA732X_CP_63KHZ (4 << 0) +#define DA732X_CP_0KHZ (5 << 0) + +/* DA732X_REG_PLL_CTRL (addr=0x53) */ +#define DA732X_PLL_INDIV_MASK (3 << 0) +#define DA732X_PLL_SRM_EN (1 << 2) +#define DA732X_PLL_EN (1 << 7) +#define DA732X_PLL_BYPASS (0 << 0) + +/* DA732X_REG_CLK_CTRL (addr=0x54) */ +#define DA732X_SR1_MASK (0xF) +#define DA732X_SR2_MASK (0xF0) + +/* DA732X_REG_CLK_DSP (addr=0x5A) */ +#define DA732X_DSP_FREQ_MASK (7 << 0) +#define DA732X_DSP_FREQ_12MHZ (0 << 0) +#define DA732X_DSP_FREQ_24MHZ (1 << 0) +#define DA732X_DSP_FREQ_36MHZ (2 << 0) +#define DA732X_DSP_FREQ_48MHZ (3 << 0) +#define DA732X_DSP_FREQ_60MHZ (4 << 0) +#define DA732X_DSP_FREQ_72MHZ (5 << 0) +#define DA732X_DSP_FREQ_84MHZ (6 << 0) +#define DA732X_DSP_FREQ_96MHZ (7 << 0) + +/* DA732X_REG_CLK_EN1 (addr=0x5B) */ +#define DA732X_DSP_CLK_EN (1 << 0) +#define DA732X_SYS3_CLK_EN (1 << 1) +#define DA732X_DSP12_CLK_EN (1 << 2) +#define DA732X_PC_CLK_EN (1 << 3) +#define DA732X_MCLK_SQR_EN (1 << 7) + +/* DA732X_REG_CLK_EN2 (addr=0x5C) */ +#define DA732X_UART_CLK_EN (1 << 1) +#define DA732X_CP_CLK_EN (1 << 2) +#define DA732X_CP_CLK_DIS (0 << 2) + +/* DA732X_REG_CLK_EN3 (addr=0x5D) */ +#define DA732X_ADCA_BB_CLK_EN (1 << 0) +#define DA732X_ADCC_BB_CLK_EN (1 << 4) + +/* DA732X_REG_CLK_EN4 (addr=0x5E) */ +#define DA732X_DACA_BB_CLK_EN (1 << 0) +#define DA732X_DACC_BB_CLK_EN (1 << 4) +#define DA732X_DACA_BB_CLK_SHIFT 0 +#define DA732X_DACC_BB_CLK_SHIFT 4 + +/* DA732X_REG_CLK_EN5 (addr=0x5F) */ +#define DA732X_DACE_BB_CLK_EN (1 << 0) +#define DA732X_DACE_BB_CLK_SHIFT 0 + +/* DA732X_REG_AIF_MCLK (addr=0x60) */ +#define DA732X_AIFM_FRAME_64 (1 << 2) +#define DA732X_AIFM_SRC_SEL_AIFA (1 << 6) +#define DA732X_CLK_GENERATION_AIF_A (1 << 4) +#define DA732X_NO_CLK_GENERATION 0x0 + +/* DA732X_REG_AIFA1 (addr=0x61) */ +#define DA732X_AIF_WORD_MASK (0x3 << 0) +#define DA732X_AIF_WORD_16 (0 << 0) +#define DA732X_AIF_WORD_20 (1 << 0) +#define DA732X_AIF_WORD_24 (2 << 0) +#define DA732X_AIF_WORD_32 (3 << 0) +#define DA732X_AIF_TDM_MONO_SHIFT (1 << 6) +#define DA732X_AIF1_CLK_MASK (1 << 7) +#define DA732X_AIF_SLAVE (0 << 7) +#define DA732X_AIF_CLK_FROM_SRC (1 << 7) + +/* DA732X_REG_AIFA3 (addr=0x63) */ +#define DA732X_AIF_MODE_SHIFT 0 +#define DA732X_AIF_MODE_MASK 0x3 +#define DA732X_AIF_I2S_MODE (0 << 0) +#define DA732X_AIF_LEFT_J_MODE (1 << 0) +#define DA732X_AIF_RIGHT_J_MODE (2 << 0) +#define DA732X_AIF_DSP_MODE (3 << 0) +#define DA732X_AIF_WCLK_INV (1 << 4) +#define DA732X_AIF_BCLK_INV (1 << 5) +#define DA732X_AIF_EN (1 << 7) +#define DA732X_AIF_EN_SHIFT 7 + +/* DA732X_REG_PC_CTRL (addr=0x6a) */ +#define DA732X_PC_PULSE_AIFA (0 << 0) +#define DA732X_PC_PULSE_AIFB (1 << 0) +#define DA732X_PC_RESYNC_AUT (1 << 6) +#define DA732X_PC_RESYNC_NOT_AUT (0 << 6) +#define DA732X_PC_SAME (1 << 7) + +/* DA732X_REG_DATA_ROUTE (addr=0x70) */ +#define DA732X_ADC1_TO_AIFA (0 << 0) +#define DA732X_DSP_TO_AIFA (1 << 0) +#define DA732X_ADC2_TO_AIFB (0 << 1) +#define DA732X_DSP_TO_AIFB (1 << 1) +#define DA732X_AIFA_TO_DAC1L (0 << 2) +#define DA732X_DSP_TO_DAC1L (1 << 2) +#define DA732X_AIFA_TO_DAC1R (0 << 3) +#define DA732X_DSP_TO_DAC1R (1 << 3) +#define DA732X_AIFB_TO_DAC2L (0 << 4) +#define DA732X_DSP_TO_DAC2L (1 << 4) +#define DA732X_AIFB_TO_DAC2R (0 << 5) +#define DA732X_DSP_TO_DAC2R (1 << 5) +#define DA732X_AIFB_TO_DAC3 (0 << 6) +#define DA732X_DSP_TO_DAC3 (1 << 6) +#define DA732X_BYPASS_DSP (0 << 0) +#define DA732X_ALL_TO_DSP (0x7F << 0) + +/* DA732X_REG_DSP_CTRL (addr=0x71) */ +#define DA732X_DIGITAL_EN (1 << 0) +#define DA732X_DIGITAL_RESET (0 << 0) +#define DA732X_DSP_CORE_EN (1 << 1) +#define DA732X_DSP_CORE_RESET (0 << 1) + +/* DA732X_REG_SPARE1_OUT (addr=0x7D)*/ +#define DA732X_HP_DRIVER_EN (1 << 0) +#define DA732X_HP_GATE_LOW (1 << 2) +#define DA732X_HP_LOOP_GAIN_CTRL (1 << 3) + +/* DA732X_REG_ID (addr=0x81)*/ +#define DA732X_ID_MINOR_MASK (0xF << 0) +#define DA732X_ID_MAJOR_MASK (0xF << 4) + +/* DA732X_REG_ADC1/2_PD (addr=0x90/0x98) */ +#define DA732X_ADC_RST_MASK (0x3 << 0) +#define DA732X_ADC_PD_MASK (0x3 << 2) +#define DA732X_ADC_SET_ACT (0x3 << 0) +#define DA732X_ADC_SET_RST (0x0 << 0) +#define DA732X_ADC_ON (0x3 << 2) +#define DA732X_ADC_OFF (0x0 << 2) + +/* DA732X_REG_ADC1/2_SEL (addr=0x94/0x9C) */ +#define DA732X_ADC_VOL_VAL_MASK 0x7 +#define DA732X_ADCL_VOL_SHIFT 0 +#define DA732X_ADCR_VOL_SHIFT 4 +#define DA732X_ADCL_EN_SHIFT 2 +#define DA732X_ADCR_EN_SHIFT 3 +#define DA732X_ADCL_EN (1 << 2) +#define DA732X_ADCR_EN (1 << 3) +#define DA732X_ADC_VOL_VAL_MAX DA732X_ADC_VOL_VAL_MASK + +/* + * DA732X_REG_ADC1/2_HPF (addr=0x93/0x9b) + * DA732x_REG_DAC1/2/3_HPG (addr=0xA5/0xB5/0xC5) + */ +#define DA732X_HPF_MUSIC_EN (1 << 3) +#define DA732X_HPF_VOICE_EN ((1 << 3) | (1 << 7)) +#define DA732X_HPF_MASK ((1 << 3) | (1 << 7)) +#define DA732X_HPF_DIS ((0 << 3) | (0 << 7)) + +/* DA732X_REG_DAC1/2/3_VOL */ +#define DA732X_DAC_VOL_VAL_MASK 0x7F +#define DA732X_DAC_VOL_SHIFT 0 +#define DA732X_DAC_VOL_VAL_MAX DA732X_DAC_VOL_VAL_MASK + +/* DA732X_REG_DAC1/2/3_SEL (addr=0xA3/0xB3/0xC3) */ +#define DA732X_DACL_EN_SHIFT 3 +#define DA732X_DACR_EN_SHIFT 7 +#define DA732X_DACL_MUTE_SHIFT 2 +#define DA732X_DACR_MUTE_SHIFT 6 +#define DA732X_DACL_EN (1 << 3) +#define DA732X_DACR_EN (1 << 7) +#define DA732X_DACL_SDM (1 << 0) +#define DA732X_DACR_SDM (1 << 4) +#define DA732X_DACL_MUTE (1 << 2) +#define DA732X_DACR_MUTE (1 << 6) + +/* DA732X_REG_DAC_SOFTMUTE (addr=0xA4/0xB4/0xC4) */ +#define DA732X_SOFTMUTE_EN (1 << 7) +#define DA732X_GAIN_RAMPED (1 << 6) +#define DA732X_16_SAMPLES (4 << 0) +#define DA732X_SOFTMUTE_MASK (1 << 7) +#define DA732X_SOFTMUTE_SHIFT 7 + +/* + * DA732x_REG_ADC1/2_EQ12 (addr=0x95/0x9D) + * DA732x_REG_ADC1/2_EQ34 (addr=0x96/0x9E) + * DA732x_REG_ADC1/2_EQ5 (addr=0x97/0x9F) + * DA732x_REG_DAC1/2/3_EQ12 (addr=0xA5/0xB5/0xC5) + * DA732x_REG_DAC1/2/3_EQ34 (addr=0xA6/0xB6/0xC6) + * DA732x_REG_DAC1/2/3_EQ5 (addr=0xA7/0xB7/0xB7) + */ +#define DA732X_EQ_VOL_VAL_MASK 0xF +#define DA732X_EQ_BAND1_SHIFT 0 +#define DA732X_EQ_BAND2_SHIFT 4 +#define DA732X_EQ_BAND3_SHIFT 0 +#define DA732X_EQ_BAND4_SHIFT 4 +#define DA732X_EQ_BAND5_SHIFT 0 +#define DA732X_EQ_OVERALL_SHIFT 4 +#define DA732X_EQ_OVERALL_VOL_VAL_MASK 0x3 +#define DA732X_EQ_DIS (0 << 7) +#define DA732X_EQ_EN (1 << 7) +#define DA732X_EQ_EN_SHIFT 7 +#define DA732X_EQ_VOL_VAL_MAX DA732X_EQ_VOL_VAL_MASK +#define DA732X_EQ_OVERALL_VOL_VAL_MAX DA732X_EQ_OVERALL_VOL_VAL_MASK + +/* DA732X_REG_DMA_CMD (addr=0xD3) */ +#define DA732X_SEL_DSP_DMA_MASK (3 << 0) +#define DA732X_SEL_DSP_DMA_DIS (0 << 0) +#define DA732X_SEL_DSP_DMA_PMEM (1 << 0) +#define DA732X_SEL_DSP_DMA_XMEM (2 << 0) +#define DA732X_SEL_DSP_DMA_YMEM (3 << 0) +#define DA732X_DSP_RW_MASK (1 << 4) +#define DA732X_DSP_DMA_WRITE (0 << 4) +#define DA732X_DSP_DMA_READ (1 << 4) + +/* DA732X_REG_DMA_STATUS (addr=0xDA) */ +#define DA732X_DSP_DMA_FREE (0 << 0) +#define DA732X_DSP_DMA_BUSY (1 << 0) + +#endif /* __DA732X_REG_H_ */ diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c new file mode 100644 index 000000000000..5d8f39e32978 --- /dev/null +++ b/sound/soc/codecs/isabelle.c @@ -0,0 +1,1176 @@ +/* + * isabelle.c - Low power high fidelity audio codec driver + * + * Copyright (c) 2012 Texas Instruments, Inc + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * + * Initially based on sound/soc/codecs/twl6040.c + * + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/regmap.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <asm/div64.h> +#include "isabelle.h" + + +/* Register default values for ISABELLE driver. */ +static struct reg_default isabelle_reg_defs[] = { + { 0, 0x00 }, + { 1, 0x00 }, + { 2, 0x00 }, + { 3, 0x00 }, + { 4, 0x00 }, + { 5, 0x00 }, + { 6, 0x00 }, + { 7, 0x00 }, + { 8, 0x00 }, + { 9, 0x00 }, + { 10, 0x00 }, + { 11, 0x00 }, + { 12, 0x00 }, + { 13, 0x00 }, + { 14, 0x00 }, + { 15, 0x00 }, + { 16, 0x00 }, + { 17, 0x00 }, + { 18, 0x00 }, + { 19, 0x00 }, + { 20, 0x00 }, + { 21, 0x02 }, + { 22, 0x02 }, + { 23, 0x02 }, + { 24, 0x02 }, + { 25, 0x0F }, + { 26, 0x8F }, + { 27, 0x0F }, + { 28, 0x8F }, + { 29, 0x00 }, + { 30, 0x00 }, + { 31, 0x00 }, + { 32, 0x00 }, + { 33, 0x00 }, + { 34, 0x00 }, + { 35, 0x00 }, + { 36, 0x00 }, + { 37, 0x00 }, + { 38, 0x00 }, + { 39, 0x00 }, + { 40, 0x00 }, + { 41, 0x00 }, + { 42, 0x00 }, + { 43, 0x00 }, + { 44, 0x00 }, + { 45, 0x00 }, + { 46, 0x00 }, + { 47, 0x00 }, + { 48, 0x00 }, + { 49, 0x00 }, + { 50, 0x00 }, + { 51, 0x00 }, + { 52, 0x00 }, + { 53, 0x00 }, + { 54, 0x00 }, + { 55, 0x00 }, + { 56, 0x00 }, + { 57, 0x00 }, + { 58, 0x00 }, + { 59, 0x00 }, + { 60, 0x00 }, + { 61, 0x00 }, + { 62, 0x00 }, + { 63, 0x00 }, + { 64, 0x00 }, + { 65, 0x00 }, + { 66, 0x00 }, + { 67, 0x00 }, + { 68, 0x00 }, + { 69, 0x90 }, + { 70, 0x90 }, + { 71, 0x90 }, + { 72, 0x00 }, + { 73, 0x00 }, + { 74, 0x00 }, + { 75, 0x00 }, + { 76, 0x00 }, + { 77, 0x00 }, + { 78, 0x00 }, + { 79, 0x00 }, + { 80, 0x00 }, + { 81, 0x00 }, + { 82, 0x00 }, + { 83, 0x00 }, + { 84, 0x00 }, + { 85, 0x07 }, + { 86, 0x00 }, + { 87, 0x00 }, + { 88, 0x00 }, + { 89, 0x07 }, + { 90, 0x80 }, + { 91, 0x07 }, + { 92, 0x07 }, + { 93, 0x00 }, + { 94, 0x00 }, + { 95, 0x00 }, + { 96, 0x00 }, + { 97, 0x00 }, + { 98, 0x00 }, + { 99, 0x00 }, +}; + +static const char *isabelle_rx1_texts[] = {"VRX1", "ARX1"}; +static const char *isabelle_rx2_texts[] = {"VRX2", "ARX2"}; + +static const struct soc_enum isabelle_rx1_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3, 1, isabelle_rx1_texts), + SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5, 1, isabelle_rx1_texts), +}; + +static const struct soc_enum isabelle_rx2_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2, 1, isabelle_rx2_texts), + SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4, 1, isabelle_rx2_texts), +}; + +/* Headset DAC playback switches */ +static const struct snd_kcontrol_new rx1_mux_controls = + SOC_DAPM_ENUM("Route", isabelle_rx1_enum); + +static const struct snd_kcontrol_new rx2_mux_controls = + SOC_DAPM_ENUM("Route", isabelle_rx2_enum); + +/* TX input selection */ +static const char *isabelle_atx_texts[] = {"AMIC1", "DMIC"}; +static const char *isabelle_vtx_texts[] = {"AMIC2", "DMIC"}; + +static const struct soc_enum isabelle_atx_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7, 1, isabelle_atx_texts), + SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_atx_texts), +}; + +static const struct soc_enum isabelle_vtx_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6, 1, isabelle_vtx_texts), + SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_vtx_texts), +}; + +static const struct snd_kcontrol_new atx_mux_controls = + SOC_DAPM_ENUM("Route", isabelle_atx_enum); + +static const struct snd_kcontrol_new vtx_mux_controls = + SOC_DAPM_ENUM("Route", isabelle_vtx_enum); + +/* Left analog microphone selection */ +static const char *isabelle_amic1_texts[] = { + "Main Mic", "Headset Mic", "Aux/FM Left"}; + +/* Left analog microphone selection */ +static const char *isabelle_amic2_texts[] = {"Sub Mic", "Aux/FM Right"}; + +static const struct soc_enum isabelle_amic1_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 5, + ARRAY_SIZE(isabelle_amic1_texts), + isabelle_amic1_texts), +}; + +static const struct soc_enum isabelle_amic2_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 4, + ARRAY_SIZE(isabelle_amic2_texts), + isabelle_amic2_texts), +}; + +static const struct snd_kcontrol_new amic1_control = + SOC_DAPM_ENUM("Route", isabelle_amic1_enum); + +static const struct snd_kcontrol_new amic2_control = + SOC_DAPM_ENUM("Route", isabelle_amic2_enum); + +static const char *isabelle_st_audio_texts[] = {"ATX1", "ATX2"}; + +static const char *isabelle_st_voice_texts[] = {"VTX1", "VTX2"}; + +static const struct soc_enum isabelle_st_audio_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7, 1, + isabelle_st_audio_texts), + SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7, 1, + isabelle_st_audio_texts), +}; + +static const struct soc_enum isabelle_st_voice_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7, 1, + isabelle_st_voice_texts), + SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7, 1, + isabelle_st_voice_texts), +}; + +static const struct snd_kcontrol_new st_audio_control = + SOC_DAPM_ENUM("Route", isabelle_st_audio_enum); + +static const struct snd_kcontrol_new st_voice_control = + SOC_DAPM_ENUM("Route", isabelle_st_voice_enum); + +/* Mixer controls */ +static const struct snd_kcontrol_new isabelle_hs_left_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC1L Playback Switch", ISABELLE_HSDRV_CFG1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_HSDRV_CFG1_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_hs_right_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC1R Playback Switch", ISABELLE_HSDRV_CFG1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_HSDRV_CFG1_REG, 4, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_hf_left_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC2L Playback Switch", ISABELLE_HFLPGA_CFG_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_HFLPGA_CFG_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_hf_right_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC2R Playback Switch", ISABELLE_HFRPGA_CFG_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_HFRPGA_CFG_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_ep_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC2L Playback Switch", ISABELLE_EARDRV_CFG1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_EARDRV_CFG1_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_aux_left_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC3L Playback Switch", ISABELLE_LINEAMP_CFG_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_LINEAMP_CFG_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_aux_right_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC3R Playback Switch", ISABELLE_LINEAMP_CFG_REG, 5, 1, 0), +SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_LINEAMP_CFG_REG, 4, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga1_left_mixer_controls[] = { +SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 7, 1, 0), +SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 6, 1, 0), +SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 5, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga1_right_mixer_controls[] = { +SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 3, 1, 0), +SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 2, 1, 0), +SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 1, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga2_left_mixer_controls[] = { +SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 7, 1, 0), +SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 6, 1, 0), +SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 5, 1, 0), +SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 4, 1, 0), +SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 3, 1, 0), +SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 2, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga2_right_mixer_controls[] = { +SOC_DAPM_SINGLE("USNC Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 7, 1, 0), +SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 3, 1, 0), +SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 2, 1, 0), +SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 1, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga3_left_mixer_controls[] = { +SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 7, 1, 0), +SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 6, 1, 0), +SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 5, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga3_right_mixer_controls[] = { +SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 3, 1, 0), +SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 2, 1, 0), +SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 1, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx1_mixer_controls[] = { +SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DL1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx2_mixer_controls[] = { +SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 5, 1, 0), +SOC_DAPM_SINGLE("DL2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 4, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx3_mixer_controls[] = { +SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 3, 1, 0), +SOC_DAPM_SINGLE("DL3 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 2, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx4_mixer_controls[] = { +SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DL4 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 0, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx5_mixer_controls[] = { +SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DL5 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx6_mixer_controls[] = { +SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("DL6 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 4, 1, 0), +}; + +static const struct snd_kcontrol_new ep_path_enable_control = + SOC_DAPM_SINGLE("Switch", ISABELLE_EARDRV_CFG2_REG, 0, 1, 0); + +/* TLV Declarations */ +static const DECLARE_TLV_DB_SCALE(mic_amp_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(afm_amp_tlv, -3300, 300, 0); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -1200, 200, 0); +static const DECLARE_TLV_DB_SCALE(hf_tlv, -5000, 200, 0); + +/* from -63 to 0 dB in 1 dB steps */ +static const DECLARE_TLV_DB_SCALE(dpga_tlv, -6300, 100, 1); + +/* from -63 to 9 dB in 1 dB steps */ +static const DECLARE_TLV_DB_SCALE(rx_tlv, -6300, 100, 1); + +static const DECLARE_TLV_DB_SCALE(st_tlv, -2700, 300, 1); +static const DECLARE_TLV_DB_SCALE(tx_tlv, -600, 100, 0); + +static const struct snd_kcontrol_new isabelle_snd_controls[] = { + SOC_DOUBLE_TLV("Headset Playback Volume", ISABELLE_HSDRV_GAIN_REG, + 4, 0, 0xF, 0, dac_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + ISABELLE_HFLPGA_CFG_REG, ISABELLE_HFRPGA_CFG_REG, + 0, 0x1F, 0, hf_tlv), + SOC_DOUBLE_TLV("Aux Playback Volume", ISABELLE_LINEAMP_GAIN_REG, + 4, 0, 0xF, 0, dac_tlv), + SOC_SINGLE_TLV("Earpiece Playback Volume", ISABELLE_EARDRV_CFG1_REG, + 0, 0xF, 0, dac_tlv), + + SOC_DOUBLE_TLV("Aux FM Volume", ISABELLE_APGA_GAIN_REG, 4, 0, 0xF, 0, + afm_amp_tlv), + SOC_SINGLE_TLV("Mic1 Capture Volume", ISABELLE_MIC1_GAIN_REG, 3, 0x1F, + 0, mic_amp_tlv), + SOC_SINGLE_TLV("Mic2 Capture Volume", ISABELLE_MIC2_GAIN_REG, 3, 0x1F, + 0, mic_amp_tlv), + + SOC_DOUBLE_R_TLV("DPGA1 Volume", ISABELLE_DPGA1L_GAIN_REG, + ISABELLE_DPGA1R_GAIN_REG, 0, 0x3F, 0, dpga_tlv), + SOC_DOUBLE_R_TLV("DPGA2 Volume", ISABELLE_DPGA2L_GAIN_REG, + ISABELLE_DPGA2R_GAIN_REG, 0, 0x3F, 0, dpga_tlv), + SOC_DOUBLE_R_TLV("DPGA3 Volume", ISABELLE_DPGA3L_GAIN_REG, + ISABELLE_DPGA3R_GAIN_REG, 0, 0x3F, 0, dpga_tlv), + + SOC_SINGLE_TLV("Sidetone Audio TX1 Volume", + ISABELLE_ATX_STPGA1_CFG_REG, 0, 0xF, 0, st_tlv), + SOC_SINGLE_TLV("Sidetone Audio TX2 Volume", + ISABELLE_ATX_STPGA2_CFG_REG, 0, 0xF, 0, st_tlv), + SOC_SINGLE_TLV("Sidetone Voice TX1 Volume", + ISABELLE_VTX_STPGA1_CFG_REG, 0, 0xF, 0, st_tlv), + SOC_SINGLE_TLV("Sidetone Voice TX2 Volume", + ISABELLE_VTX2_STPGA2_CFG_REG, 0, 0xF, 0, st_tlv), + + SOC_SINGLE_TLV("Audio TX1 Volume", ISABELLE_ATX1_DPGA_REG, 4, 0xF, 0, + tx_tlv), + SOC_SINGLE_TLV("Audio TX2 Volume", ISABELLE_ATX2_DPGA_REG, 4, 0xF, 0, + tx_tlv), + SOC_SINGLE_TLV("Voice TX1 Volume", ISABELLE_VTX1_DPGA_REG, 4, 0xF, 0, + tx_tlv), + SOC_SINGLE_TLV("Voice TX2 Volume", ISABELLE_VTX2_DPGA_REG, 4, 0xF, 0, + tx_tlv), + + SOC_SINGLE_TLV("RX1 DPGA Volume", ISABELLE_RX1_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX2 DPGA Volume", ISABELLE_RX2_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX3 DPGA Volume", ISABELLE_RX3_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX4 DPGA Volume", ISABELLE_RX4_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX5 DPGA Volume", ISABELLE_RX5_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX6 DPGA Volume", ISABELLE_RX6_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + + SOC_SINGLE("Headset Noise Gate", ISABELLE_HS_NG_CFG1_REG, 7, 1, 0), + SOC_SINGLE("Handsfree Noise Gate", ISABELLE_HF_NG_CFG1_REG, 7, 1, 0), + + SOC_SINGLE("ATX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 7, 1, 0), + SOC_SINGLE("ATX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 6, 1, 0), + SOC_SINGLE("ARX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 5, 1, 0), + SOC_SINGLE("ARX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 4, 1, 0), + SOC_SINGLE("ARX3 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 3, 1, 0), + SOC_SINGLE("ARX4 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 2, 1, 0), + SOC_SINGLE("ARX5 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 1, 1, 0), + SOC_SINGLE("ARX6 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 0, 1, 0), + SOC_SINGLE("VRX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 3, 1, 0), + SOC_SINGLE("VRX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 2, 1, 0), + + SOC_SINGLE("ATX1 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG, + 7, 1, 0), + SOC_SINGLE("ATX2 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG, + 6, 1, 0), + SOC_SINGLE("VTX1 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG, + 5, 1, 0), + SOC_SINGLE("VTX2 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG, + 4, 1, 0), + SOC_SINGLE("RX1 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 5, 1, 0), + SOC_SINGLE("RX2 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 4, 1, 0), + SOC_SINGLE("RX3 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 3, 1, 0), + SOC_SINGLE("RX4 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 2, 1, 0), + SOC_SINGLE("RX5 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 1, 1, 0), + SOC_SINGLE("RX6 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 0, 1, 0), + + SOC_SINGLE("ULATX12 Capture Switch", ISABELLE_ULATX12_INTF_CFG_REG, + 7, 1, 0), + + SOC_SINGLE("DL12 Playback Switch", ISABELLE_DL12_INTF_CFG_REG, + 7, 1, 0), + SOC_SINGLE("DL34 Playback Switch", ISABELLE_DL34_INTF_CFG_REG, + 7, 1, 0), + SOC_SINGLE("DL56 Playback Switch", ISABELLE_DL56_INTF_CFG_REG, + 7, 1, 0), + + /* DMIC Switch */ + SOC_SINGLE("DMIC Switch", ISABELLE_DMIC_CFG_REG, 0, 1, 0), +}; + +static const struct snd_soc_dapm_widget isabelle_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("MAINMIC"), + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("SUBMIC"), + SND_SOC_DAPM_INPUT("LINEIN1"), + SND_SOC_DAPM_INPUT("LINEIN2"), + SND_SOC_DAPM_INPUT("DMICDAT"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HSOL"), + SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("HFL"), + SND_SOC_DAPM_OUTPUT("HFR"), + SND_SOC_DAPM_OUTPUT("EP"), + SND_SOC_DAPM_OUTPUT("LINEOUT1"), + SND_SOC_DAPM_OUTPUT("LINEOUT2"), + + SND_SOC_DAPM_PGA("DL1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL4", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL5", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL6", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Analog input muxes for the capture amplifiers */ + SND_SOC_DAPM_MUX("Analog Left Capture Route", + SND_SOC_NOPM, 0, 0, &amic1_control), + SND_SOC_DAPM_MUX("Analog Right Capture Route", + SND_SOC_NOPM, 0, 0, &amic2_control), + + SND_SOC_DAPM_MUX("Sidetone Audio Playback", SND_SOC_NOPM, 0, 0, + &st_audio_control), + SND_SOC_DAPM_MUX("Sidetone Voice Playback", SND_SOC_NOPM, 0, 0, + &st_voice_control), + + /* AIF */ + SND_SOC_DAPM_AIF_IN("INTF1_SDI", NULL, 0, ISABELLE_INTF_EN_REG, 7, 0), + SND_SOC_DAPM_AIF_IN("INTF2_SDI", NULL, 0, ISABELLE_INTF_EN_REG, 6, 0), + + SND_SOC_DAPM_AIF_OUT("INTF1_SDO", NULL, 0, ISABELLE_INTF_EN_REG, 5, 0), + SND_SOC_DAPM_AIF_OUT("INTF2_SDO", NULL, 0, ISABELLE_INTF_EN_REG, 4, 0), + + SND_SOC_DAPM_OUT_DRV("ULATX1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("ULATX2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("ULVTX1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("ULVTX2", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Analog Capture PGAs */ + SND_SOC_DAPM_PGA("MicAmp1", ISABELLE_AMIC_CFG_REG, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("MicAmp2", ISABELLE_AMIC_CFG_REG, 4, 0, NULL, 0), + + /* Auxiliary FM PGAs */ + SND_SOC_DAPM_PGA("APGA1", ISABELLE_APGA_CFG_REG, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("APGA2", ISABELLE_APGA_CFG_REG, 6, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC1", "Left Front Capture", + ISABELLE_AMIC_CFG_REG, 7, 0), + SND_SOC_DAPM_ADC("ADC2", "Right Front Capture", + ISABELLE_AMIC_CFG_REG, 6, 0), + + /* Microphone Bias */ + SND_SOC_DAPM_SUPPLY("Headset Mic Bias", ISABELLE_ABIAS_CFG_REG, + 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Main Mic Bias", ISABELLE_ABIAS_CFG_REG, + 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Digital Mic1 Bias", + ISABELLE_DBIAS_CFG_REG, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Digital Mic2 Bias", + ISABELLE_DBIAS_CFG_REG, 2, 0, NULL, 0), + + /* Mixers */ + SND_SOC_DAPM_MIXER("Headset Left Mixer", SND_SOC_NOPM, 0, 0, + isabelle_hs_left_mixer_controls, + ARRAY_SIZE(isabelle_hs_left_mixer_controls)), + SND_SOC_DAPM_MIXER("Headset Right Mixer", SND_SOC_NOPM, 0, 0, + isabelle_hs_right_mixer_controls, + ARRAY_SIZE(isabelle_hs_right_mixer_controls)), + SND_SOC_DAPM_MIXER("Handsfree Left Mixer", SND_SOC_NOPM, 0, 0, + isabelle_hf_left_mixer_controls, + ARRAY_SIZE(isabelle_hf_left_mixer_controls)), + SND_SOC_DAPM_MIXER("Handsfree Right Mixer", SND_SOC_NOPM, 0, 0, + isabelle_hf_right_mixer_controls, + ARRAY_SIZE(isabelle_hf_right_mixer_controls)), + SND_SOC_DAPM_MIXER("LINEOUT1 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_aux_left_mixer_controls, + ARRAY_SIZE(isabelle_aux_left_mixer_controls)), + SND_SOC_DAPM_MIXER("LINEOUT2 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_aux_right_mixer_controls, + ARRAY_SIZE(isabelle_aux_right_mixer_controls)), + SND_SOC_DAPM_MIXER("Earphone Mixer", SND_SOC_NOPM, 0, 0, + isabelle_ep_mixer_controls, + ARRAY_SIZE(isabelle_ep_mixer_controls)), + + SND_SOC_DAPM_MIXER("DPGA1L Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga1_left_mixer_controls, + ARRAY_SIZE(isabelle_dpga1_left_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA1R Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga1_right_mixer_controls, + ARRAY_SIZE(isabelle_dpga1_right_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA2L Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga2_left_mixer_controls, + ARRAY_SIZE(isabelle_dpga2_left_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA2R Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga2_right_mixer_controls, + ARRAY_SIZE(isabelle_dpga2_right_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA3L Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga3_left_mixer_controls, + ARRAY_SIZE(isabelle_dpga3_left_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA3R Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga3_right_mixer_controls, + ARRAY_SIZE(isabelle_dpga3_right_mixer_controls)), + + SND_SOC_DAPM_MIXER("RX1 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx1_mixer_controls, + ARRAY_SIZE(isabelle_rx1_mixer_controls)), + SND_SOC_DAPM_MIXER("RX2 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx2_mixer_controls, + ARRAY_SIZE(isabelle_rx2_mixer_controls)), + SND_SOC_DAPM_MIXER("RX3 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx3_mixer_controls, + ARRAY_SIZE(isabelle_rx3_mixer_controls)), + SND_SOC_DAPM_MIXER("RX4 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx4_mixer_controls, + ARRAY_SIZE(isabelle_rx4_mixer_controls)), + SND_SOC_DAPM_MIXER("RX5 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx5_mixer_controls, + ARRAY_SIZE(isabelle_rx5_mixer_controls)), + SND_SOC_DAPM_MIXER("RX6 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx6_mixer_controls, + ARRAY_SIZE(isabelle_rx6_mixer_controls)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC1L", "Headset Playback", ISABELLE_DAC_CFG_REG, + 5, 0), + SND_SOC_DAPM_DAC("DAC1R", "Headset Playback", ISABELLE_DAC_CFG_REG, + 4, 0), + SND_SOC_DAPM_DAC("DAC2L", "Handsfree Playback", ISABELLE_DAC_CFG_REG, + 3, 0), + SND_SOC_DAPM_DAC("DAC2R", "Handsfree Playback", ISABELLE_DAC_CFG_REG, + 2, 0), + SND_SOC_DAPM_DAC("DAC3L", "Lineout Playback", ISABELLE_DAC_CFG_REG, + 1, 0), + SND_SOC_DAPM_DAC("DAC3R", "Lineout Playback", ISABELLE_DAC_CFG_REG, + 0, 0), + + /* Analog Playback PGAs */ + SND_SOC_DAPM_PGA("Sidetone Audio PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Sidetone Voice PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HF Left PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HF Right PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA1L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA1R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA2L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA2R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA3L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA3R", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Analog Playback Mux */ + SND_SOC_DAPM_MUX("RX1 Playback", ISABELLE_ALU_RX_EN_REG, 5, 0, + &rx1_mux_controls), + SND_SOC_DAPM_MUX("RX2 Playback", ISABELLE_ALU_RX_EN_REG, 4, 0, + &rx2_mux_controls), + + /* TX Select */ + SND_SOC_DAPM_MUX("ATX Select", ISABELLE_TX_INPUT_CFG_REG, + 7, 0, &atx_mux_controls), + SND_SOC_DAPM_MUX("VTX Select", ISABELLE_TX_INPUT_CFG_REG, + 6, 0, &vtx_mux_controls), + + SND_SOC_DAPM_SWITCH("Earphone Playback", SND_SOC_NOPM, 0, 0, + &ep_path_enable_control), + + /* Output Drivers */ + SND_SOC_DAPM_OUT_DRV("HS Left Driver", ISABELLE_HSDRV_CFG2_REG, + 1, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HS Right Driver", ISABELLE_HSDRV_CFG2_REG, + 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("LINEOUT1 Left Driver", ISABELLE_LINEAMP_CFG_REG, + 1, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("LINEOUT2 Right Driver", ISABELLE_LINEAMP_CFG_REG, + 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Earphone Driver", ISABELLE_EARDRV_CFG2_REG, + 1, 0, NULL, 0), + + SND_SOC_DAPM_OUT_DRV("HF Left Driver", ISABELLE_HFDRV_CFG_REG, + 1, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HF Right Driver", ISABELLE_HFDRV_CFG_REG, + 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route isabelle_intercon[] = { + /* Interface mapping */ + { "DL1", "DL12 Playback Switch", "INTF1_SDI" }, + { "DL2", "DL12 Playback Switch", "INTF1_SDI" }, + { "DL3", "DL34 Playback Switch", "INTF1_SDI" }, + { "DL4", "DL34 Playback Switch", "INTF1_SDI" }, + { "DL5", "DL56 Playback Switch", "INTF1_SDI" }, + { "DL6", "DL56 Playback Switch", "INTF1_SDI" }, + + { "DL1", "DL12 Playback Switch", "INTF2_SDI" }, + { "DL2", "DL12 Playback Switch", "INTF2_SDI" }, + { "DL3", "DL34 Playback Switch", "INTF2_SDI" }, + { "DL4", "DL34 Playback Switch", "INTF2_SDI" }, + { "DL5", "DL56 Playback Switch", "INTF2_SDI" }, + { "DL6", "DL56 Playback Switch", "INTF2_SDI" }, + + /* Input side mapping */ + { "Sidetone Audio PGA", NULL, "Sidetone Audio Playback" }, + { "Sidetone Voice PGA", NULL, "Sidetone Voice Playback" }, + + { "RX1 Mixer", "ST1 Playback Switch", "Sidetone Audio PGA" }, + + { "RX1 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" }, + { "RX1 Mixer", "DL1 Playback Switch", "DL1" }, + + { "RX2 Mixer", "ST2 Playback Switch", "Sidetone Audio PGA" }, + + { "RX2 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" }, + { "RX2 Mixer", "DL2 Playback Switch", "DL2" }, + + { "RX3 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" }, + { "RX3 Mixer", "DL3 Playback Switch", "DL3" }, + + { "RX4 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" }, + { "RX4 Mixer", "DL4 Playback Switch", "DL4" }, + + { "RX5 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" }, + { "RX5 Mixer", "DL5 Playback Switch", "DL5" }, + + { "RX6 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" }, + { "RX6 Mixer", "DL6 Playback Switch", "DL6" }, + + /* Capture path */ + { "Analog Left Capture Route", "Headset Mic", "HSMIC" }, + { "Analog Left Capture Route", "Main Mic", "MAINMIC" }, + { "Analog Left Capture Route", "Aux/FM Left", "LINEIN1" }, + + { "Analog Right Capture Route", "Sub Mic", "SUBMIC" }, + { "Analog Right Capture Route", "Aux/FM Right", "LINEIN2" }, + + { "MicAmp1", NULL, "Analog Left Capture Route" }, + { "MicAmp2", NULL, "Analog Right Capture Route" }, + + { "ADC1", NULL, "MicAmp1" }, + { "ADC2", NULL, "MicAmp2" }, + + { "ATX Select", "AMIC1", "ADC1" }, + { "ATX Select", "DMIC", "DMICDAT" }, + { "ATX Select", "AMIC2", "ADC2" }, + + { "VTX Select", "AMIC1", "ADC1" }, + { "VTX Select", "DMIC", "DMICDAT" }, + { "VTX Select", "AMIC2", "ADC2" }, + + { "ULATX1", "ATX1 Filter Enable Switch", "ATX Select" }, + { "ULATX1", "ATX1 Filter Bypass Switch", "ATX Select" }, + { "ULATX2", "ATX2 Filter Enable Switch", "ATX Select" }, + { "ULATX2", "ATX2 Filter Bypass Switch", "ATX Select" }, + + { "ULVTX1", "VTX1 Filter Enable Switch", "VTX Select" }, + { "ULVTX1", "VTX1 Filter Bypass Switch", "VTX Select" }, + { "ULVTX2", "VTX2 Filter Enable Switch", "VTX Select" }, + { "ULVTX2", "VTX2 Filter Bypass Switch", "VTX Select" }, + + { "INTF1_SDO", "ULATX12 Capture Switch", "ULATX1" }, + { "INTF1_SDO", "ULATX12 Capture Switch", "ULATX2" }, + { "INTF2_SDO", "ULATX12 Capture Switch", "ULATX1" }, + { "INTF2_SDO", "ULATX12 Capture Switch", "ULATX2" }, + + { "INTF1_SDO", NULL, "ULVTX1" }, + { "INTF1_SDO", NULL, "ULVTX2" }, + { "INTF2_SDO", NULL, "ULVTX1" }, + { "INTF2_SDO", NULL, "ULVTX2" }, + + /* AFM Path */ + { "APGA1", NULL, "LINEIN1" }, + { "APGA2", NULL, "LINEIN2" }, + + { "RX1 Playback", "VRX1 Filter Bypass Switch", "RX1 Mixer" }, + { "RX1 Playback", "ARX1 Filter Bypass Switch", "RX1 Mixer" }, + { "RX1 Playback", "RX1 Filter Enable Switch", "RX1 Mixer" }, + + { "RX2 Playback", "VRX2 Filter Bypass Switch", "RX2 Mixer" }, + { "RX2 Playback", "ARX2 Filter Bypass Switch", "RX2 Mixer" }, + { "RX2 Playback", "RX2 Filter Enable Switch", "RX2 Mixer" }, + + { "RX3 Playback", "ARX3 Filter Bypass Switch", "RX3 Mixer" }, + { "RX3 Playback", "RX3 Filter Enable Switch", "RX3 Mixer" }, + + { "RX4 Playback", "ARX4 Filter Bypass Switch", "RX4 Mixer" }, + { "RX4 Playback", "RX4 Filter Enable Switch", "RX4 Mixer" }, + + { "RX5 Playback", "ARX5 Filter Bypass Switch", "RX5 Mixer" }, + { "RX5 Playback", "RX5 Filter Enable Switch", "RX5 Mixer" }, + + { "RX6 Playback", "ARX6 Filter Bypass Switch", "RX6 Mixer" }, + { "RX6 Playback", "RX6 Filter Enable Switch", "RX6 Mixer" }, + + { "DPGA1L Mixer", "RX1 Playback Switch", "RX1 Playback" }, + { "DPGA1L Mixer", "RX3 Playback Switch", "RX3 Playback" }, + { "DPGA1L Mixer", "RX5 Playback Switch", "RX5 Playback" }, + + { "DPGA1R Mixer", "RX2 Playback Switch", "RX2 Playback" }, + { "DPGA1R Mixer", "RX4 Playback Switch", "RX4 Playback" }, + { "DPGA1R Mixer", "RX6 Playback Switch", "RX6 Playback" }, + + { "DPGA1L", NULL, "DPGA1L Mixer" }, + { "DPGA1R", NULL, "DPGA1R Mixer" }, + + { "DAC1L", NULL, "DPGA1L" }, + { "DAC1R", NULL, "DPGA1R" }, + + { "DPGA2L Mixer", "RX1 Playback Switch", "RX1 Playback" }, + { "DPGA2L Mixer", "RX2 Playback Switch", "RX2 Playback" }, + { "DPGA2L Mixer", "RX3 Playback Switch", "RX3 Playback" }, + { "DPGA2L Mixer", "RX4 Playback Switch", "RX4 Playback" }, + { "DPGA2L Mixer", "RX5 Playback Switch", "RX5 Playback" }, + { "DPGA2L Mixer", "RX6 Playback Switch", "RX6 Playback" }, + + { "DPGA2R Mixer", "RX2 Playback Switch", "RX2 Playback" }, + { "DPGA2R Mixer", "RX4 Playback Switch", "RX4 Playback" }, + { "DPGA2R Mixer", "RX6 Playback Switch", "RX6 Playback" }, + + { "DPGA2L", NULL, "DPGA2L Mixer" }, + { "DPGA2R", NULL, "DPGA2R Mixer" }, + + { "DAC2L", NULL, "DPGA2L" }, + { "DAC2R", NULL, "DPGA2R" }, + + { "DPGA3L Mixer", "RX1 Playback Switch", "RX1 Playback" }, + { "DPGA3L Mixer", "RX3 Playback Switch", "RX3 Playback" }, + { "DPGA3L Mixer", "RX5 Playback Switch", "RX5 Playback" }, + + { "DPGA3R Mixer", "RX2 Playback Switch", "RX2 Playback" }, + { "DPGA3R Mixer", "RX4 Playback Switch", "RX4 Playback" }, + { "DPGA3R Mixer", "RX6 Playback Switch", "RX6 Playback" }, + + { "DPGA3L", NULL, "DPGA3L Mixer" }, + { "DPGA3R", NULL, "DPGA3R Mixer" }, + + { "DAC3L", NULL, "DPGA3L" }, + { "DAC3R", NULL, "DPGA3R" }, + + { "Headset Left Mixer", "DAC1L Playback Switch", "DAC1L" }, + { "Headset Left Mixer", "APGA1 Playback Switch", "APGA1" }, + + { "Headset Right Mixer", "DAC1R Playback Switch", "DAC1R" }, + { "Headset Right Mixer", "APGA2 Playback Switch", "APGA2" }, + + { "HS Left Driver", NULL, "Headset Left Mixer" }, + { "HS Right Driver", NULL, "Headset Right Mixer" }, + + { "HSOL", NULL, "HS Left Driver" }, + { "HSOR", NULL, "HS Right Driver" }, + + /* Earphone playback path */ + { "Earphone Mixer", "DAC2L Playback Switch", "DAC2L" }, + { "Earphone Mixer", "APGA1 Playback Switch", "APGA1" }, + + { "Earphone Playback", "Switch", "Earphone Mixer" }, + { "Earphone Driver", NULL, "Earphone Playback" }, + { "EP", NULL, "Earphone Driver" }, + + { "Handsfree Left Mixer", "DAC2L Playback Switch", "DAC2L" }, + { "Handsfree Left Mixer", "APGA1 Playback Switch", "APGA1" }, + + { "Handsfree Right Mixer", "DAC2R Playback Switch", "DAC2R" }, + { "Handsfree Right Mixer", "APGA2 Playback Switch", "APGA2" }, + + { "HF Left PGA", NULL, "Handsfree Left Mixer" }, + { "HF Right PGA", NULL, "Handsfree Right Mixer" }, + + { "HF Left Driver", NULL, "HF Left PGA" }, + { "HF Right Driver", NULL, "HF Right PGA" }, + + { "HFL", NULL, "HF Left Driver" }, + { "HFR", NULL, "HF Right Driver" }, + + { "LINEOUT1 Mixer", "DAC3L Playback Switch", "DAC3L" }, + { "LINEOUT1 Mixer", "APGA1 Playback Switch", "APGA1" }, + + { "LINEOUT2 Mixer", "DAC3R Playback Switch", "DAC3R" }, + { "LINEOUT2 Mixer", "APGA2 Playback Switch", "APGA2" }, + + { "LINEOUT1 Driver", NULL, "LINEOUT1 Mixer" }, + { "LINEOUT2 Driver", NULL, "LINEOUT2 Mixer" }, + + { "LINEOUT1", NULL, "LINEOUT1 Driver" }, + { "LINEOUT2", NULL, "LINEOUT2 Driver" }, +}; + +static int isabelle_hs_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, ISABELLE_DAC1_SOFTRAMP_REG, + BIT(4), (mute ? BIT(4) : 0)); + + return 0; +} + +static int isabelle_hf_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, ISABELLE_DAC2_SOFTRAMP_REG, + BIT(4), (mute ? BIT(4) : 0)); + + return 0; +} + +static int isabelle_line_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, ISABELLE_DAC3_SOFTRAMP_REG, + BIT(4), (mute ? BIT(4) : 0)); + + return 0; +} + +static int isabelle_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, ISABELLE_PWR_EN_REG, + ISABELLE_CHIP_EN, BIT(0)); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, ISABELLE_PWR_EN_REG, + ISABELLE_CHIP_EN, 0); + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +static int isabelle_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + u16 aif = 0; + unsigned int fs_val = 0; + + switch (params_rate(params)) { + case 8000: + fs_val = ISABELLE_FS_RATE_8; + break; + case 11025: + fs_val = ISABELLE_FS_RATE_11; + break; + case 12000: + fs_val = ISABELLE_FS_RATE_12; + break; + case 16000: + fs_val = ISABELLE_FS_RATE_16; + break; + case 22050: + fs_val = ISABELLE_FS_RATE_22; + break; + case 24000: + fs_val = ISABELLE_FS_RATE_24; + break; + case 32000: + fs_val = ISABELLE_FS_RATE_32; + break; + case 44100: + fs_val = ISABELLE_FS_RATE_44; + break; + case 48000: + fs_val = ISABELLE_FS_RATE_48; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ISABELLE_FS_RATE_CFG_REG, + ISABELLE_FS_RATE_MASK, fs_val); + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S20_3LE: + aif |= ISABELLE_AIF_LENGTH_20; + break; + case SNDRV_PCM_FORMAT_S32_LE: + aif |= ISABELLE_AIF_LENGTH_32; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ISABELLE_INTF_CFG_REG, + ISABELLE_AIF_LENGTH_MASK, aif); + + return 0; +} + +static int isabelle_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int aif_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + aif_val &= ~ISABELLE_AIF_MS; + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif_val |= ISABELLE_AIF_MS; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aif_val |= ISABELLE_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + aif_val |= ISABELLE_LEFT_J_MODE; + break; + case SND_SOC_DAIFMT_PDM: + aif_val |= ISABELLE_PDM_MODE; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ISABELLE_INTF_CFG_REG, + (ISABELLE_AIF_MS | ISABELLE_AIF_FMT_MASK), aif_val); + + return 0; +} + +/* Rates supported by Isabelle driver */ +#define ISABELLE_RATES SNDRV_PCM_RATE_8000_48000 + +/* Formates supported by Isabelle driver. */ +#define ISABELLE_FORMATS (SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops isabelle_hs_dai_ops = { + .hw_params = isabelle_hw_params, + .set_fmt = isabelle_set_dai_fmt, + .digital_mute = isabelle_hs_mute, +}; + +static struct snd_soc_dai_ops isabelle_hf_dai_ops = { + .hw_params = isabelle_hw_params, + .set_fmt = isabelle_set_dai_fmt, + .digital_mute = isabelle_hf_mute, +}; + +static struct snd_soc_dai_ops isabelle_line_dai_ops = { + .hw_params = isabelle_hw_params, + .set_fmt = isabelle_set_dai_fmt, + .digital_mute = isabelle_line_mute, +}; + +static struct snd_soc_dai_ops isabelle_ul_dai_ops = { + .hw_params = isabelle_hw_params, + .set_fmt = isabelle_set_dai_fmt, +}; + +/* ISABELLE dai structure */ +static struct snd_soc_dai_driver isabelle_dai[] = { + { + .name = "isabelle-dl1", + .playback = { + .stream_name = "Headset Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ISABELLE_RATES, + .formats = ISABELLE_FORMATS, + }, + .ops = &isabelle_hs_dai_ops, + }, + { + .name = "isabelle-dl2", + .playback = { + .stream_name = "Handsfree Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ISABELLE_RATES, + .formats = ISABELLE_FORMATS, + }, + .ops = &isabelle_hf_dai_ops, + }, + { + .name = "isabelle-lineout", + .playback = { + .stream_name = "Lineout Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ISABELLE_RATES, + .formats = ISABELLE_FORMATS, + }, + .ops = &isabelle_line_dai_ops, + }, + { + .name = "isabelle-ul", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ISABELLE_RATES, + .formats = ISABELLE_FORMATS, + }, + .ops = &isabelle_ul_dai_ops, + }, +}; + +static int isabelle_probe(struct snd_soc_codec *codec) +{ + int ret = 0; + + codec->control_data = dev_get_regmap(codec->dev, NULL); + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_isabelle = { + .probe = isabelle_probe, + .set_bias_level = isabelle_set_bias_level, + .controls = isabelle_snd_controls, + .num_controls = ARRAY_SIZE(isabelle_snd_controls), + .dapm_widgets = isabelle_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(isabelle_dapm_widgets), + .dapm_routes = isabelle_intercon, + .num_dapm_routes = ARRAY_SIZE(isabelle_intercon), + .idle_bias_off = true, +}; + +static const struct regmap_config isabelle_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = ISABELLE_MAX_REGISTER, + .reg_defaults = isabelle_reg_defs, + .num_reg_defaults = ARRAY_SIZE(isabelle_reg_defs), + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit isabelle_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *isabelle_regmap; + int ret = 0; + + isabelle_regmap = devm_regmap_init_i2c(i2c, &isabelle_regmap_config); + if (IS_ERR(isabelle_regmap)) { + ret = PTR_ERR(isabelle_regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + i2c_set_clientdata(i2c, isabelle_regmap); + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_isabelle, isabelle_dai, + ARRAY_SIZE(isabelle_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + return ret; +} + +static int __devexit isabelle_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id isabelle_i2c_id[] = { + { "isabelle", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, isabelle_i2c_id); + +static struct i2c_driver isabelle_i2c_driver = { + .driver = { + .name = "isabelle", + .owner = THIS_MODULE, + }, + .probe = isabelle_i2c_probe, + .remove = __devexit_p(isabelle_i2c_remove), + .id_table = isabelle_i2c_id, +}; + +module_i2c_driver(isabelle_i2c_driver); + +MODULE_DESCRIPTION("ASoC ISABELLE driver"); +MODULE_AUTHOR("Vishwas A Deshpande <vishwas.a.deshpande@ti.com>"); +MODULE_AUTHOR("M R Swami Reddy <MR.Swami.Reddy@ti.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/isabelle.h b/sound/soc/codecs/isabelle.h new file mode 100644 index 000000000000..96d839a8c956 --- /dev/null +++ b/sound/soc/codecs/isabelle.h @@ -0,0 +1,143 @@ +/* + * isabelle.h - Low power high fidelity audio codec driver header file + * + * Copyright (c) 2012 Texas Instruments, Inc + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + */ + +#ifndef _ISABELLE_H +#define _ISABELLE_H + +#include <linux/bitops.h> + +/* ISABELLE REGISTERS */ + +#define ISABELLE_PWR_CFG_REG 0x01 +#define ISABELLE_PWR_EN_REG 0x02 +#define ISABELLE_PS_EN1_REG 0x03 +#define ISABELLE_INT1_STATUS_REG 0x04 +#define ISABELLE_INT1_MASK_REG 0x05 +#define ISABELLE_INT2_STATUS_REG 0x06 +#define ISABELLE_INT2_MASK_REG 0x07 +#define ISABELLE_HKCTL1_REG 0x08 +#define ISABELLE_HKCTL2_REG 0x09 +#define ISABELLE_HKCTL3_REG 0x0A +#define ISABELLE_ACCDET_STATUS_REG 0x0B +#define ISABELLE_BUTTON_ID_REG 0x0C +#define ISABELLE_PLL_CFG_REG 0x10 +#define ISABELLE_PLL_EN_REG 0x11 +#define ISABELLE_FS_RATE_CFG_REG 0x12 +#define ISABELLE_INTF_CFG_REG 0x13 +#define ISABELLE_INTF_EN_REG 0x14 +#define ISABELLE_ULATX12_INTF_CFG_REG 0x15 +#define ISABELLE_DL12_INTF_CFG_REG 0x16 +#define ISABELLE_DL34_INTF_CFG_REG 0x17 +#define ISABELLE_DL56_INTF_CFG_REG 0x18 +#define ISABELLE_ATX_STPGA1_CFG_REG 0x19 +#define ISABELLE_ATX_STPGA2_CFG_REG 0x1A +#define ISABELLE_VTX_STPGA1_CFG_REG 0x1B +#define ISABELLE_VTX2_STPGA2_CFG_REG 0x1C +#define ISABELLE_ATX1_DPGA_REG 0x1D +#define ISABELLE_ATX2_DPGA_REG 0x1E +#define ISABELLE_VTX1_DPGA_REG 0x1F +#define ISABELLE_VTX2_DPGA_REG 0x20 +#define ISABELLE_TX_INPUT_CFG_REG 0x21 +#define ISABELLE_RX_INPUT_CFG_REG 0x22 +#define ISABELLE_RX_INPUT_CFG2_REG 0x23 +#define ISABELLE_VOICE_HPF_CFG_REG 0x24 +#define ISABELLE_AUDIO_HPF_CFG_REG 0x25 +#define ISABELLE_RX1_DPGA_REG 0x26 +#define ISABELLE_RX2_DPGA_REG 0x27 +#define ISABELLE_RX3_DPGA_REG 0x28 +#define ISABELLE_RX4_DPGA_REG 0x29 +#define ISABELLE_RX5_DPGA_REG 0x2A +#define ISABELLE_RX6_DPGA_REG 0x2B +#define ISABELLE_ALU_TX_EN_REG 0x2C +#define ISABELLE_ALU_RX_EN_REG 0x2D +#define ISABELLE_IIR_RESYNC_REG 0x2E +#define ISABELLE_ABIAS_CFG_REG 0x30 +#define ISABELLE_DBIAS_CFG_REG 0x31 +#define ISABELLE_MIC1_GAIN_REG 0x32 +#define ISABELLE_MIC2_GAIN_REG 0x33 +#define ISABELLE_AMIC_CFG_REG 0x34 +#define ISABELLE_DMIC_CFG_REG 0x35 +#define ISABELLE_APGA_GAIN_REG 0x36 +#define ISABELLE_APGA_CFG_REG 0x37 +#define ISABELLE_TX_GAIN_DLY_REG 0x38 +#define ISABELLE_RX_GAIN_DLY_REG 0x39 +#define ISABELLE_RX_PWR_CTRL_REG 0x3A +#define ISABELLE_DPGA1LR_IN_SEL_REG 0x3B +#define ISABELLE_DPGA1L_GAIN_REG 0x3C +#define ISABELLE_DPGA1R_GAIN_REG 0x3D +#define ISABELLE_DPGA2L_IN_SEL_REG 0x3E +#define ISABELLE_DPGA2R_IN_SEL_REG 0x3F +#define ISABELLE_DPGA2L_GAIN_REG 0x40 +#define ISABELLE_DPGA2R_GAIN_REG 0x41 +#define ISABELLE_DPGA3LR_IN_SEL_REG 0x42 +#define ISABELLE_DPGA3L_GAIN_REG 0x43 +#define ISABELLE_DPGA3R_GAIN_REG 0x44 +#define ISABELLE_DAC1_SOFTRAMP_REG 0x45 +#define ISABELLE_DAC2_SOFTRAMP_REG 0x46 +#define ISABELLE_DAC3_SOFTRAMP_REG 0x47 +#define ISABELLE_DAC_CFG_REG 0x48 +#define ISABELLE_EARDRV_CFG1_REG 0x49 +#define ISABELLE_EARDRV_CFG2_REG 0x4A +#define ISABELLE_HSDRV_GAIN_REG 0x4B +#define ISABELLE_HSDRV_CFG1_REG 0x4C +#define ISABELLE_HSDRV_CFG2_REG 0x4D +#define ISABELLE_HS_NG_CFG1_REG 0x4E +#define ISABELLE_HS_NG_CFG2_REG 0x4F +#define ISABELLE_LINEAMP_GAIN_REG 0x50 +#define ISABELLE_LINEAMP_CFG_REG 0x51 +#define ISABELLE_HFL_VOL_CTRL_REG 0x52 +#define ISABELLE_HFL_SFTVOL_CTRL_REG 0x53 +#define ISABELLE_HFL_LIM_CTRL_1_REG 0x54 +#define ISABELLE_HFL_LIM_CTRL_2_REG 0x55 +#define ISABELLE_HFR_VOL_CTRL_REG 0x56 +#define ISABELLE_HFR_SFTVOL_CTRL_REG 0x57 +#define ISABELLE_HFR_LIM_CTRL_1_REG 0x58 +#define ISABELLE_HFR_LIM_CTRL_2_REG 0x59 +#define ISABELLE_HF_MODE_REG 0x5A +#define ISABELLE_HFLPGA_CFG_REG 0x5B +#define ISABELLE_HFRPGA_CFG_REG 0x5C +#define ISABELLE_HFDRV_CFG_REG 0x5D +#define ISABELLE_PDMOUT_CFG1_REG 0x5E +#define ISABELLE_PDMOUT_CFG2_REG 0x5F +#define ISABELLE_PDMOUT_L_WM_REG 0x60 +#define ISABELLE_PDMOUT_R_WM_REG 0x61 +#define ISABELLE_HF_NG_CFG1_REG 0x62 +#define ISABELLE_HF_NG_CFG2_REG 0x63 + +/* ISABELLE_PWR_EN_REG (0x02h) */ +#define ISABELLE_CHIP_EN BIT(0) + +/* ISABELLE DAI FORMATS */ +#define ISABELLE_AIF_FMT_MASK 0x70 +#define ISABELLE_I2S_MODE 0x0 +#define ISABELLE_LEFT_J_MODE 0x1 +#define ISABELLE_PDM_MODE 0x2 + +#define ISABELLE_AIF_LENGTH_MASK 0x30 +#define ISABELLE_AIF_LENGTH_20 0x00 +#define ISABELLE_AIF_LENGTH_32 0x10 + +#define ISABELLE_AIF_MS 0x80 + +#define ISABELLE_FS_RATE_MASK 0xF +#define ISABELLE_FS_RATE_8 0x0 +#define ISABELLE_FS_RATE_11 0x1 +#define ISABELLE_FS_RATE_12 0x2 +#define ISABELLE_FS_RATE_16 0x4 +#define ISABELLE_FS_RATE_22 0x5 +#define ISABELLE_FS_RATE_24 0x6 +#define ISABELLE_FS_RATE_32 0x8 +#define ISABELLE_FS_RATE_44 0x9 +#define ISABELLE_FS_RATE_48 0xA + +#define ISABELLE_MAX_REGISTER 0xFF + +#endif diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 4624e752a188..85d9cabe6d55 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -164,8 +164,7 @@ static int jz4740_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { uint32_t val; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; switch (params_rate(params)) { case 8000: diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c new file mode 100644 index 000000000000..99b0a9dcff34 --- /dev/null +++ b/sound/soc/codecs/lm49453.c @@ -0,0 +1,1549 @@ +/* + * lm49453.c - LM49453 ALSA Soc Audio driver + * + * Copyright (c) 2012 Texas Instruments, Inc + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * Initially based on sound/soc/codecs/wm8350.c + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <asm/div64.h> +#include "lm49453.h" + +static struct reg_default lm49453_reg_defs[] = { + { 0, 0x00 }, + { 1, 0x00 }, + { 2, 0x00 }, + { 3, 0x00 }, + { 4, 0x00 }, + { 5, 0x00 }, + { 6, 0x00 }, + { 7, 0x00 }, + { 8, 0x00 }, + { 9, 0x00 }, + { 10, 0x00 }, + { 11, 0x00 }, + { 12, 0x00 }, + { 13, 0x00 }, + { 14, 0x00 }, + { 15, 0x00 }, + { 16, 0x00 }, + { 17, 0x00 }, + { 18, 0x00 }, + { 19, 0x00 }, + { 20, 0x00 }, + { 21, 0x00 }, + { 22, 0x00 }, + { 23, 0x00 }, + { 32, 0x00 }, + { 33, 0x00 }, + { 35, 0x00 }, + { 36, 0x00 }, + { 37, 0x00 }, + { 46, 0x00 }, + { 48, 0x00 }, + { 49, 0x00 }, + { 51, 0x00 }, + { 56, 0x00 }, + { 58, 0x00 }, + { 59, 0x00 }, + { 60, 0x00 }, + { 61, 0x00 }, + { 62, 0x00 }, + { 63, 0x00 }, + { 64, 0x00 }, + { 65, 0x00 }, + { 66, 0x00 }, + { 67, 0x00 }, + { 68, 0x00 }, + { 69, 0x00 }, + { 70, 0x00 }, + { 71, 0x00 }, + { 72, 0x00 }, + { 73, 0x00 }, + { 74, 0x00 }, + { 75, 0x00 }, + { 76, 0x00 }, + { 77, 0x00 }, + { 78, 0x00 }, + { 79, 0x00 }, + { 80, 0x00 }, + { 81, 0x00 }, + { 82, 0x00 }, + { 83, 0x00 }, + { 85, 0x00 }, + { 85, 0x00 }, + { 86, 0x00 }, + { 87, 0x00 }, + { 88, 0x00 }, + { 89, 0x00 }, + { 90, 0x00 }, + { 91, 0x00 }, + { 92, 0x00 }, + { 93, 0x00 }, + { 94, 0x00 }, + { 95, 0x00 }, + { 96, 0x01 }, + { 97, 0x00 }, + { 98, 0x00 }, + { 99, 0x00 }, + { 100, 0x00 }, + { 101, 0x00 }, + { 102, 0x00 }, + { 103, 0x01 }, + { 105, 0x01 }, + { 106, 0x00 }, + { 107, 0x01 }, + { 107, 0x00 }, + { 108, 0x00 }, + { 109, 0x00 }, + { 110, 0x00 }, + { 111, 0x02 }, + { 112, 0x02 }, + { 113, 0x00 }, + { 121, 0x80 }, + { 122, 0xBB }, + { 123, 0x80 }, + { 124, 0xBB }, + { 128, 0x00 }, + { 130, 0x00 }, + { 131, 0x00 }, + { 132, 0x00 }, + { 133, 0x0A }, + { 134, 0x0A }, + { 135, 0x0A }, + { 136, 0x0F }, + { 137, 0x00 }, + { 138, 0x73 }, + { 139, 0x33 }, + { 140, 0x73 }, + { 141, 0x33 }, + { 142, 0x73 }, + { 143, 0x33 }, + { 144, 0x73 }, + { 145, 0x33 }, + { 146, 0x73 }, + { 147, 0x33 }, + { 148, 0x73 }, + { 149, 0x33 }, + { 150, 0x73 }, + { 151, 0x33 }, + { 152, 0x00 }, + { 153, 0x00 }, + { 154, 0x00 }, + { 155, 0x00 }, + { 176, 0x00 }, + { 177, 0x00 }, + { 178, 0x00 }, + { 179, 0x00 }, + { 180, 0x00 }, + { 181, 0x00 }, + { 182, 0x00 }, + { 183, 0x00 }, + { 184, 0x00 }, + { 185, 0x00 }, + { 186, 0x00 }, + { 189, 0x00 }, + { 188, 0x00 }, + { 194, 0x00 }, + { 195, 0x00 }, + { 196, 0x00 }, + { 197, 0x00 }, + { 200, 0x00 }, + { 201, 0x00 }, + { 202, 0x00 }, + { 203, 0x00 }, + { 204, 0x00 }, + { 205, 0x00 }, + { 208, 0x00 }, + { 209, 0x00 }, + { 210, 0x00 }, + { 211, 0x00 }, + { 213, 0x00 }, + { 214, 0x00 }, + { 215, 0x00 }, + { 216, 0x00 }, + { 217, 0x00 }, + { 218, 0x00 }, + { 219, 0x00 }, + { 221, 0x00 }, + { 222, 0x00 }, + { 224, 0x00 }, + { 225, 0x00 }, + { 226, 0x00 }, + { 227, 0x00 }, + { 228, 0x00 }, + { 229, 0x00 }, + { 230, 0x13 }, + { 231, 0x00 }, + { 232, 0x80 }, + { 233, 0x0C }, + { 234, 0xDD }, + { 235, 0x00 }, + { 236, 0x04 }, + { 237, 0x00 }, + { 238, 0x00 }, + { 239, 0x00 }, + { 240, 0x00 }, + { 241, 0x00 }, + { 242, 0x00 }, + { 243, 0x00 }, + { 244, 0x00 }, + { 245, 0x00 }, + { 248, 0x00 }, + { 249, 0x00 }, + { 254, 0x00 }, + { 255, 0x00 }, +}; + +/* codec private data */ +struct lm49453_priv { + struct regmap *regmap; + int fs_rate; +}; + +/* capture path controls */ + +static const char *lm49453_mic2mode_text[] = {"Single Ended", "Differential"}; + +static const SOC_ENUM_SINGLE_DECL(lm49453_mic2mode_enum, LM49453_P0_MICR_REG, 5, + lm49453_mic2mode_text); + +static const char *lm49453_dmic_cfg_text[] = {"DMICDAT1", "DMICDAT2"}; + +static const SOC_ENUM_SINGLE_DECL(lm49453_dmic12_cfg_enum, + LM49453_P0_DIGITAL_MIC1_CONFIG_REG, + 7, lm49453_dmic_cfg_text); + +static const SOC_ENUM_SINGLE_DECL(lm49453_dmic34_cfg_enum, + LM49453_P0_DIGITAL_MIC2_CONFIG_REG, + 7, lm49453_dmic_cfg_text); + +/* MUX Controls */ +static const char *lm49453_adcl_mux_text[] = { "MIC1", "Aux_L" }; + +static const char *lm49453_adcr_mux_text[] = { "MIC2", "Aux_R" }; + +static const struct soc_enum lm49453_adcl_enum = + SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 0, + ARRAY_SIZE(lm49453_adcl_mux_text), + lm49453_adcl_mux_text); + +static const struct soc_enum lm49453_adcr_enum = + SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 1, + ARRAY_SIZE(lm49453_adcr_mux_text), + lm49453_adcr_mux_text); + +static const struct snd_kcontrol_new lm49453_adcl_mux_control = + SOC_DAPM_ENUM("ADC Left Mux", lm49453_adcl_enum); + +static const struct snd_kcontrol_new lm49453_adcr_mux_control = + SOC_DAPM_ENUM("ADC Right Mux", lm49453_adcr_enum); + +static const struct snd_kcontrol_new lm49453_headset_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHPL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHPL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHPL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHPL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHPL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHPL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHPL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHPL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHPL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHPL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHPL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHPL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHPL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHPL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHPL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHPL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 0, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_headset_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHPR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHPR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHPR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHPR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHPR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHPR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHPR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHPR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHPR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHPR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHPR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHPR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHPR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHPR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHPR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHPR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 1, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_speaker_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLSL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLSL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLSL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLSL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLSL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLSL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLSL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLSL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLSL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLSL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLSL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLSL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLSL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLSL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLSL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLSL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 2, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_speaker_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLSR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLSR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLSR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLSR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLSR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLSR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLSR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLSR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLSR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLSR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLSR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLSR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLSR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLSR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLSR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLSR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 3, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_haptic_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHAL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHAL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHAL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHAL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHAL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHAL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHAL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHAL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHAL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHAL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHAL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHAL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHAL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHAL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHAL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHAL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 4, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_haptic_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHAR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHAR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHAR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHAR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHAR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHAR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHAR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHAR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHAR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHAR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHAR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHAR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHAR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHAR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHAR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHAR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 5, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_lineout_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLOL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLOL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLOL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLOL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLOL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLOL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLOL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLOL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLOL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLOL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLOL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLOL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLOL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLOL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLOL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLOL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 6, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_lineout_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLOR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLOR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLOR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLOR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLOR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLOR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLOR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLOR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLOR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLOR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLOR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLOR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLOR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLOR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLOR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLOR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 7, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx1_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_PORT1_TX1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_PORT1_TX1_REG, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx2_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_PORT1_TX2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT1_TX2_REG, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx3_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX3_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX3_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX3_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX3_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX3_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX3_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_PORT1_TX3_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx4_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX4_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX4_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX4_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX4_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX4_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX4_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_PORT1_TX4_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx5_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX5_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX5_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX5_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX5_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX5_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX5_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_PORT1_TX5_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx6_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX6_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX6_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX6_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX6_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX6_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX6_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_PORT1_TX6_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx7_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX7_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX7_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX7_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX7_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX7_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX7_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_PORT1_TX7_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx8_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX8_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX8_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX8_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX8_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX8_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX8_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_PORT1_TX8_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port2_tx1_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT2_TX1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT2_TX1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT2_TX1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT2_TX1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT2_TX1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT2_TX1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_PORT2_TX1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_PORT2_TX1_REG, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port2_tx2_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT2_TX2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT2_TX2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT2_TX2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT2_TX2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT2_TX2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT2_TX2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_PORT2_TX2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0), +}; + +/* TLV Declarations */ +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0); + +static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = { +/* Sidetone supports mono only */ +SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, + 0, 0x3F, 0, digital_tlv), +}; + +static const struct snd_kcontrol_new lm49453_snd_controls[] = { + /* mic1 and mic2 supports mono only */ + SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6, + 0, digital_tlv), + SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6, + 0, digital_tlv), + + SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, + LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG, + LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv), + + SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum), + SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum), + SOC_DAPM_ENUM("DMIC34 SRC", lm49453_dmic34_cfg_enum), + + /* Capture path filter enable */ + SOC_SINGLE("DMIC1 HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG, + 0, 1, 0), + SOC_SINGLE("DMIC2 HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG, + 1, 1, 0), + SOC_SINGLE("ADC HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG, + 2, 1, 0), + + SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG, + LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG, + LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG, + LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG, + LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv), + + SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG, + 0, 6, 0, digital_tlv), + + SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 0, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_2_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 2, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_3_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 4, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_4_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 6, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_5_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 0, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_6_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 2, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_7_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 4, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_8_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 6, 3, 0, port_tlv), + + SOC_SINGLE_TLV("PORT2_1_RX_LVL Volume", LM49453_P0_PORT2_RX_LVL_REG, + 0, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT2_2_RX_LVL Volume", LM49453_P0_PORT2_RX_LVL_REG, + 2, 3, 0, port_tlv), + + SOC_SINGLE("Port1 Playback Switch", LM49453_P0_AUDIO_PORT1_BASIC_REG, + 1, 1, 0), + SOC_SINGLE("Port2 Playback Switch", LM49453_P0_AUDIO_PORT2_BASIC_REG, + 1, 1, 0), + SOC_SINGLE("Port1 Capture Switch", LM49453_P0_AUDIO_PORT1_BASIC_REG, + 2, 1, 0), + SOC_SINGLE("Port2 Capture Switch", LM49453_P0_AUDIO_PORT2_BASIC_REG, + 2, 1, 0) + +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget lm49453_dapm_widgets[] = { + + /* All end points HP,EP, LS, Lineout and Haptic */ + SND_SOC_DAPM_OUTPUT("HPOUTL"), + SND_SOC_DAPM_OUTPUT("HPOUTR"), + SND_SOC_DAPM_OUTPUT("EPOUT"), + SND_SOC_DAPM_OUTPUT("LSOUTL"), + SND_SOC_DAPM_OUTPUT("LSOUTR"), + SND_SOC_DAPM_OUTPUT("LOOUTR"), + SND_SOC_DAPM_OUTPUT("LOOUTL"), + SND_SOC_DAPM_OUTPUT("HAOUTL"), + SND_SOC_DAPM_OUTPUT("HAOUTR"), + + SND_SOC_DAPM_INPUT("AMIC1"), + SND_SOC_DAPM_INPUT("AMIC2"), + SND_SOC_DAPM_INPUT("DMIC1DAT"), + SND_SOC_DAPM_INPUT("DMIC2DAT"), + SND_SOC_DAPM_INPUT("AUXL"), + SND_SOC_DAPM_INPUT("AUXR"), + + SND_SOC_DAPM_PGA("PORT1_1_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_2_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_3_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_4_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_5_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_6_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_7_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_8_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT2_1_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT2_2_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("AMIC1Bias", LM49453_P0_MICL_REG, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AMIC2Bias", LM49453_P0_MICR_REG, 6, 0, NULL, 0), + + /* playback path driver enables */ + SND_SOC_DAPM_OUT_DRV("Headset Switch", + LM49453_P0_PMC_SETUP_REG, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Earpiece Switch", + LM49453_P0_EP_REG, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker Left Switch", + LM49453_P0_DIS_PKVL_FB_REG, 0, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker Right Switch", + LM49453_P0_DIS_PKVL_FB_REG, 1, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Haptic Left Switch", + LM49453_P0_DIS_PKVL_FB_REG, 2, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Haptic Right Switch", + LM49453_P0_DIS_PKVL_FB_REG, 3, 1, NULL, 0), + + /* DAC */ + SND_SOC_DAPM_DAC("HPL DAC", "Headset", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HPR DAC", "Headset", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LSL DAC", "Speaker", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LSR DAC", "Speaker", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HAL DAC", "Haptic", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HAR DAC", "Haptic", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LOL DAC", "Lineout", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LOR DAC", "Lineout", SND_SOC_NOPM, 0, 0), + + + SND_SOC_DAPM_PGA("AUXL Input", + LM49453_P0_ANALOG_MIXER_ADC_REG, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUXR Input", + LM49453_P0_ANALOG_MIXER_ADC_REG, 3, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Sidetone", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("DMIC1 Left", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("DMIC1 Right", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("DMIC2 Left", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("DMIC2 Right", "Capture", SND_SOC_NOPM, 1, 0), + + SND_SOC_DAPM_ADC("ADC Left", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("ADC Right", "Capture", SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("ADCL Mux", SND_SOC_NOPM, 0, 0, + &lm49453_adcl_mux_control), + SND_SOC_DAPM_MUX("ADCR Mux", SND_SOC_NOPM, 0, 0, + &lm49453_adcr_mux_control), + + SND_SOC_DAPM_MUX("Mic1 Input", + SND_SOC_NOPM, 0, 0, &lm49453_adcl_mux_control), + + SND_SOC_DAPM_MUX("Mic2 Input", + SND_SOC_NOPM, 0, 0, &lm49453_adcr_mux_control), + + /* AIF */ + SND_SOC_DAPM_AIF_IN("PORT1_SDI", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 2, 0), + SND_SOC_DAPM_AIF_IN("PORT2_SDI", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 6, 0), + + SND_SOC_DAPM_AIF_OUT("PORT1_SDO", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 3, 0), + SND_SOC_DAPM_AIF_OUT("PORT2_SDO", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 7, 0), + + /* Port1 TX controls */ + SND_SOC_DAPM_OUT_DRV("P1_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_3_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_4_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_5_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_6_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_7_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_8_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Port2 TX controls */ + SND_SOC_DAPM_OUT_DRV("P2_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P2_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Sidetone Mixer */ + SND_SOC_DAPM_MIXER("Sidetone Mixer", SND_SOC_NOPM, 0, 0, + lm49453_sidetone_mixer_controls, + ARRAY_SIZE(lm49453_sidetone_mixer_controls)), + + /* DAC MIXERS */ + SND_SOC_DAPM_MIXER("HPL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_headset_left_mixer, + ARRAY_SIZE(lm49453_headset_left_mixer)), + SND_SOC_DAPM_MIXER("HPR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_headset_right_mixer, + ARRAY_SIZE(lm49453_headset_right_mixer)), + SND_SOC_DAPM_MIXER("LOL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_lineout_left_mixer, + ARRAY_SIZE(lm49453_lineout_left_mixer)), + SND_SOC_DAPM_MIXER("LOR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_lineout_right_mixer, + ARRAY_SIZE(lm49453_lineout_right_mixer)), + SND_SOC_DAPM_MIXER("LSL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_speaker_left_mixer, + ARRAY_SIZE(lm49453_speaker_left_mixer)), + SND_SOC_DAPM_MIXER("LSR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_speaker_right_mixer, + ARRAY_SIZE(lm49453_speaker_right_mixer)), + SND_SOC_DAPM_MIXER("HAL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_haptic_left_mixer, + ARRAY_SIZE(lm49453_haptic_left_mixer)), + SND_SOC_DAPM_MIXER("HAR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_haptic_right_mixer, + ARRAY_SIZE(lm49453_haptic_right_mixer)), + + /* Capture Mixer */ + SND_SOC_DAPM_MIXER("Port1_1 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx1_mixer, + ARRAY_SIZE(lm49453_port1_tx1_mixer)), + SND_SOC_DAPM_MIXER("Port1_2 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx2_mixer, + ARRAY_SIZE(lm49453_port1_tx2_mixer)), + SND_SOC_DAPM_MIXER("Port1_3 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx3_mixer, + ARRAY_SIZE(lm49453_port1_tx3_mixer)), + SND_SOC_DAPM_MIXER("Port1_4 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx4_mixer, + ARRAY_SIZE(lm49453_port1_tx4_mixer)), + SND_SOC_DAPM_MIXER("Port1_5 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx5_mixer, + ARRAY_SIZE(lm49453_port1_tx5_mixer)), + SND_SOC_DAPM_MIXER("Port1_6 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx6_mixer, + ARRAY_SIZE(lm49453_port1_tx6_mixer)), + SND_SOC_DAPM_MIXER("Port1_7 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx7_mixer, + ARRAY_SIZE(lm49453_port1_tx7_mixer)), + SND_SOC_DAPM_MIXER("Port1_8 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx8_mixer, + ARRAY_SIZE(lm49453_port1_tx8_mixer)), + + SND_SOC_DAPM_MIXER("Port2_1 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port2_tx1_mixer, + ARRAY_SIZE(lm49453_port2_tx1_mixer)), + SND_SOC_DAPM_MIXER("Port2_2 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port2_tx2_mixer, + ARRAY_SIZE(lm49453_port2_tx2_mixer)), +}; + +static const struct snd_soc_dapm_route lm49453_audio_map[] = { + /* Port SDI mapping */ + { "PORT1_1_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_2_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_3_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_4_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_5_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_6_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_7_RX", "Port1 Playback Switch", "PORT1_SDI" }, + { "PORT1_8_RX", "Port1 Playback Switch", "PORT1_SDI" }, + + { "PORT2_1_RX", "Port2 Playback Switch", "PORT2_SDI" }, + { "PORT2_2_RX", "Port2 Playback Switch", "PORT2_SDI" }, + + /* HP mapping */ + { "HPL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HPL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HPL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HPL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HPL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HPL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HPL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HPL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + { "HPL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HPL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HPL Mixer", "ADCL Switch", "ADC Left" }, + { "HPL Mixer", "ADCR Switch", "ADC Right" }, + { "HPL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HPL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HPL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HPL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "HPL Mixer", "Sidetone Switch", "Sidetone" }, + + { "HPL DAC", NULL, "HPL Mixer" }, + + { "HPR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HPR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HPR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HPR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HPR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HPR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HPR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HPR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "HPR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HPR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HPR Mixer", "ADCL Switch", "ADC Left" }, + { "HPR Mixer", "ADCR Switch", "ADC Right" }, + { "HPR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HPR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HPR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HPR Mixer", "DMIC2L Switch", "DMIC2 Right" }, + { "HPR Mixer", "Sidetone Switch", "Sidetone" }, + + { "HPR DAC", NULL, "HPR Mixer" }, + + { "HPOUTL", "Headset Switch", "HPL DAC"}, + { "HPOUTR", "Headset Switch", "HPR DAC"}, + + /* EP map */ + { "EPOUT", "Earpiece Switch", "HPL DAC" }, + + /* Speaker map */ + { "LSL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LSL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LSL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LSL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LSL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LSL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LSL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LSL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LSL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LSL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LSL Mixer", "ADCL Switch", "ADC Left" }, + { "LSL Mixer", "ADCR Switch", "ADC Right" }, + { "LSL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LSL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LSL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LSL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LSL Mixer", "Sidetone Switch", "Sidetone" }, + + { "LSL DAC", NULL, "LSL Mixer" }, + + { "LSR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LSR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LSR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LSR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LSR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LSR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LSR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LSR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LSR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LSR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LSR Mixer", "ADCL Switch", "ADC Left" }, + { "LSR Mixer", "ADCR Switch", "ADC Right" }, + { "LSR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LSR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LSR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LSR Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LSR Mixer", "Sidetone Switch", "Sidetone" }, + + { "LSR DAC", NULL, "LSR Mixer" }, + + { "LSOUTL", "Speaker Left Switch", "LSL DAC"}, + { "LSOUTR", "Speaker Left Switch", "LSR DAC"}, + + /* Haptic map */ + { "HAL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HAL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HAL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HAL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HAL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HAL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HAL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HAL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "HAL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HAL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HAL Mixer", "ADCL Switch", "ADC Left" }, + { "HAL Mixer", "ADCR Switch", "ADC Right" }, + { "HAL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HAL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HAL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HAL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "HAL Mixer", "Sidetone Switch", "Sidetone" }, + + { "HAL DAC", NULL, "HAL Mixer" }, + + { "HAR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HAR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HAR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HAR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HAR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HAR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HAR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HAR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "HAR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HAR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HAR Mixer", "ADCL Switch", "ADC Left" }, + { "HAR Mixer", "ADCR Switch", "ADC Right" }, + { "HAR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HAR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HAR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HAR Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "HAR Mixer", "Sideton Switch", "Sidetone" }, + + { "HAR DAC", NULL, "HAR Mixer" }, + + { "HAOUTL", "Haptic Left Switch", "HAL DAC" }, + { "HAOUTR", "Haptic Right Switch", "HAR DAC" }, + + /* Lineout map */ + { "LOL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LOL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LOL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LOL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LOL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LOL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LOL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LOL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LOL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LOL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LOL Mixer", "ADCL Switch", "ADC Left" }, + { "LOL Mixer", "ADCR Switch", "ADC Right" }, + { "LOL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LOL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LOL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LOL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LOL Mixer", "Sidetone Switch", "Sidetone" }, + + { "LOL DAC", NULL, "LOL Mixer" }, + + { "LOR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LOR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LOR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LOR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LOR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LOR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LOR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LOR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LOR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LOR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LOR Mixer", "ADCL Switch", "ADC Left" }, + { "LOR Mixer", "ADCR Switch", "ADC Right" }, + { "LOR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LOR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LOR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LOR Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LOR Mixer", "Sidetone Switch", "Sidetone" }, + + { "LOR DAC", NULL, "LOR Mixer" }, + + { "LOOUTL", NULL, "LOL DAC" }, + { "LOOUTR", NULL, "LOR DAC" }, + + /* TX map */ + /* Port1 mappings */ + { "Port1_1 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_1 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_1 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_1 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_1 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_1 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_2 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_2 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_2 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_2 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_2 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_2 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_3 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_3 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_3 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_3 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_3 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_3 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_4 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_4 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_4 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_4 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_4 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_4 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_5 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_5 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_5 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_5 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_5 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_5 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_6 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_6 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_6 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_6 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_6 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_6 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_7 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_7 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_7 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_7 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_7 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_7 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_8 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_8 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_8 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_8 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_8 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_8 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port2_1 Mixer", "ADCL Switch", "ADC Left" }, + { "Port2_1 Mixer", "ADCR Switch", "ADC Right" }, + { "Port2_1 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port2_1 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port2_1 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port2_1 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port2_2 Mixer", "ADCL Switch", "ADC Left" }, + { "Port2_2 Mixer", "ADCR Switch", "ADC Right" }, + { "Port2_2 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port2_2 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port2_2 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port2_2 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "P1_1_TX", NULL, "Port1_1 Mixer" }, + { "P1_2_TX", NULL, "Port1_2 Mixer" }, + { "P1_3_TX", NULL, "Port1_3 Mixer" }, + { "P1_4_TX", NULL, "Port1_4 Mixer" }, + { "P1_5_TX", NULL, "Port1_5 Mixer" }, + { "P1_6_TX", NULL, "Port1_6 Mixer" }, + { "P1_7_TX", NULL, "Port1_7 Mixer" }, + { "P1_8_TX", NULL, "Port1_8 Mixer" }, + + { "P2_1_TX", NULL, "Port2_1 Mixer" }, + { "P2_2_TX", NULL, "Port2_2 Mixer" }, + + { "PORT1_SDO", "Port1 Capture Switch", "P1_1_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_2_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_3_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_4_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_5_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_6_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_7_TX"}, + { "PORT1_SDO", "Port1 Capture Switch", "P1_8_TX"}, + + { "PORT2_SDO", "Port2 Capture Switch", "P2_1_TX"}, + { "PORT2_SDO", "Port2 Capture Switch", "P2_2_TX"}, + + { "Mic1 Input", NULL, "AMIC1" }, + { "Mic2 Input", NULL, "AMIC2" }, + + { "AUXL Input", NULL, "AUXL" }, + { "AUXR Input", NULL, "AUXR" }, + + /* AUX connections */ + { "ADCL Mux", "Aux_L", "AUXL Input" }, + { "ADCL Mux", "MIC1", "Mic1 Input" }, + + { "ADCR Mux", "Aux_R", "AUXR Input" }, + { "ADCR Mux", "MIC2", "Mic2 Input" }, + + /* ADC connection */ + { "ADC Left", NULL, "ADCL Mux"}, + { "ADC Right", NULL, "ADCR Mux"}, + + { "DMIC1 Left", NULL, "DMIC1DAT"}, + { "DMIC1 Right", NULL, "DMIC1DAT"}, + { "DMIC2 Left", NULL, "DMIC2DAT"}, + { "DMIC2 Right", NULL, "DMIC2DAT"}, + + /* Sidetone map */ + { "Sidetone Mixer", NULL, "ADC Left" }, + { "Sidetone Mixer", NULL, "ADC Right" }, + { "Sidetone Mixer", NULL, "DMIC1 Left" }, + { "Sidetone Mixer", NULL, "DMIC1 Right" }, + { "Sidetone Mixer", NULL, "DMIC2 Left" }, + { "Sidetone Mixer", NULL, "DMIC2 Right" }, + + { "Sidetone", "Sidetone Switch", "Sidetone Mixer" }, +}; + +static int lm49453_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + u16 clk_div = 0; + + lm49453->fs_rate = params_rate(params); + + /* Setting DAC clock dividers based on substream sample rate. */ + switch (lm49453->fs_rate) { + case 8000: + case 16000: + case 32000: + case 24000: + case 48000: + clk_div = 256; + break; + case 11025: + case 22050: + case 44100: + clk_div = 216; + break; + case 96000: + clk_div = 127; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, LM49453_P0_ADC_CLK_DIV_REG, clk_div); + snd_soc_write(codec, LM49453_P0_DAC_HP_CLK_DIV_REG, clk_div); + + return 0; +} + +static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + u16 aif_val; + int mode = 0; + int clk_phase = 0; + int clk_shift = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + aif_val = 0; + break; + case SND_SOC_DAIFMT_CBS_CFM: + aif_val = LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + break; + case SND_SOC_DAIFMT_CBM_CFS: + aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS; + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS | + LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + break; + default: + return -EINVAL; + } + + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_DSP_A: + mode = 1; + clk_phase = (1 << 5); + clk_shift = 1; + break; + case SND_SOC_DAIFMT_DSP_B: + mode = 1; + clk_phase = (1 << 5); + clk_shift = 0; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG, + LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5), + (aif_val | mode | clk_phase)); + + snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift); + + return 0; +} + +static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + u16 pll_clk = 0; + + switch (freq) { + case 12288000: + case 26000000: + case 19200000: + /* pll clk slection */ + pll_clk = 0; + break; + case 48000: + case 32576: + /* fll clk slection */ + pll_clk = BIT(4); + return 0; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, BIT(4), pll_clk); + + return 0; +} + +static int lm49453_hp_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(1)|BIT(0), + (mute ? (BIT(1)|BIT(0)) : 0)); + return 0; +} + +static int lm49453_lo_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(3)|BIT(2), + (mute ? (BIT(3)|BIT(2)) : 0)); + return 0; +} + +static int lm49453_ls_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(5)|BIT(4), + (mute ? (BIT(5)|BIT(4)) : 0)); + return 0; +} + +static int lm49453_ep_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(4), + (mute ? BIT(4) : 0)); + return 0; +} + +static int lm49453_ha_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(7)|BIT(6), + (mute ? (BIT(7)|BIT(6)) : 0)); + return 0; +} + +static int lm49453_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + regcache_sync(lm49453->regmap); + + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, + LM49453_PMC_SETUP_CHIP_EN, LM49453_CHIP_EN); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, + LM49453_PMC_SETUP_CHIP_EN, 0); + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +/* Formates supported by LM49453 driver. */ +#define LM49453_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops lm49453_headset_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_hp_mute, +}; + +static struct snd_soc_dai_ops lm49453_speaker_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_ls_mute, +}; + +static struct snd_soc_dai_ops lm49453_haptic_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_ha_mute, +}; + +static struct snd_soc_dai_ops lm49453_ep_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_ep_mute, +}; + +static struct snd_soc_dai_ops lm49453_lineout_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_lo_mute, +}; + +/* LM49453 dai structure. */ +static struct snd_soc_dai_driver lm49453_dai[] = { + { + .name = "LM49453 Headset", + .playback = { + .stream_name = "Headset", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 5, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_headset_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "LM49453 Speaker", + .playback = { + .stream_name = "Speaker", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_speaker_dai_ops, + }, + { + .name = "LM49453 Haptic", + .playback = { + .stream_name = "Haptic", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_haptic_dai_ops, + }, + { + .name = "LM49453 Earpiece", + .playback = { + .stream_name = "Earpiece", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_ep_dai_ops, + }, + { + .name = "LM49453 line out", + .playback = { + .stream_name = "Lineout", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_lineout_dai_ops, + }, +}; + +static int lm49453_suspend(struct snd_soc_codec *codec) +{ + lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int lm49453_resume(struct snd_soc_codec *codec) +{ + lm49453_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int lm49453_probe(struct snd_soc_codec *codec) +{ + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + codec->control_data = lm49453->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + return 0; +} + +/* power down chip */ +static int lm49453_remove(struct snd_soc_codec *codec) +{ + lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_lm49453 = { + .probe = lm49453_probe, + .remove = lm49453_remove, + .suspend = lm49453_suspend, + .resume = lm49453_resume, + .set_bias_level = lm49453_set_bias_level, + .controls = lm49453_snd_controls, + .num_controls = ARRAY_SIZE(lm49453_snd_controls), + .dapm_widgets = lm49453_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(lm49453_dapm_widgets), + .dapm_routes = lm49453_audio_map, + .num_dapm_routes = ARRAY_SIZE(lm49453_audio_map), + .idle_bias_off = true, +}; + +static const struct regmap_config lm49453_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = LM49453_MAX_REGISTER, + .reg_defaults = lm49453_reg_defs, + .num_reg_defaults = ARRAY_SIZE(lm49453_reg_defs), + .cache_type = REGCACHE_RBTREE, +}; + +static __devinit int lm49453_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct lm49453_priv *lm49453; + int ret = 0; + + lm49453 = devm_kzalloc(&i2c->dev, sizeof(struct lm49453_priv), + GFP_KERNEL); + + if (lm49453 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, lm49453); + + lm49453->regmap = regmap_init_i2c(i2c, &lm49453_regmap_config); + if (IS_ERR(lm49453->regmap)) { + ret = PTR_ERR(lm49453->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_lm49453, + lm49453_dai, ARRAY_SIZE(lm49453_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + regmap_exit(lm49453->regmap); + return ret; + } + + return ret; +} + +static int __devexit lm49453_i2c_remove(struct i2c_client *client) +{ + struct lm49453_priv *lm49453 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(lm49453->regmap); + return 0; +} + +static const struct i2c_device_id lm49453_i2c_id[] = { + { "lm49453", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, lm49453_i2c_id); + +static struct i2c_driver lm49453_i2c_driver = { + .driver = { + .name = "lm49453", + .owner = THIS_MODULE, + }, + .probe = lm49453_i2c_probe, + .remove = __devexit_p(lm49453_i2c_remove), + .id_table = lm49453_i2c_id, +}; + +module_i2c_driver(lm49453_i2c_driver); + +MODULE_DESCRIPTION("ASoC LM49453 driver"); +MODULE_AUTHOR("M R Swami Reddy <MR.Swami.Reddy@ti.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/lm49453.h b/sound/soc/codecs/lm49453.h new file mode 100644 index 000000000000..a63cfa5c0883 --- /dev/null +++ b/sound/soc/codecs/lm49453.h @@ -0,0 +1,380 @@ +/* + * lm49453.h - LM49453 ALSA Soc Audio drive + * + * Copyright (c) 2012 Texas Instruments, Inc + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + */ + +#ifndef _LM49453_H +#define _LM49453_H + +#include <linux/bitops.h> + +/* LM49453_P0 register space for page0 */ +#define LM49453_P0_PMC_SETUP_REG 0x00 +#define LM49453_P0_PLL_CLK_SEL1_REG 0x01 +#define LM49453_P0_PLL_CLK_SEL2_REG 0x02 +#define LM49453_P0_PMC_CLK_DIV_REG 0x03 +#define LM49453_P0_HSDET_CLK_DIV_REG 0x04 +#define LM49453_P0_DMIC_CLK_DIV_REG 0x05 +#define LM49453_P0_ADC_CLK_DIV_REG 0x06 +#define LM49453_P0_DAC_OT_CLK_DIV_REG 0x07 +#define LM49453_P0_PLL_HF_M_REG 0x08 +#define LM49453_P0_PLL_LF_M_REG 0x09 +#define LM49453_P0_PLL_NL_REG 0x0A +#define LM49453_P0_PLL_N_MODL_REG 0x0B +#define LM49453_P0_PLL_N_MODH_REG 0x0C +#define LM49453_P0_PLL_P1_REG 0x0D +#define LM49453_P0_PLL_P2_REG 0x0E +#define LM49453_P0_FLL_REF_FREQL_REG 0x0F +#define LM49453_P0_FLL_REF_FREQH_REG 0x10 +#define LM49453_P0_VCO_TARGETLL_REG 0x11 +#define LM49453_P0_VCO_TARGETLH_REG 0x12 +#define LM49453_P0_VCO_TARGETHL_REG 0x13 +#define LM49453_P0_VCO_TARGETHH_REG 0x14 +#define LM49453_P0_PLL_CONFIG_REG 0x15 +#define LM49453_P0_DAC_CLK_SEL_REG 0x16 +#define LM49453_P0_DAC_HP_CLK_DIV_REG 0x17 + +/* Analog Mixer Input Stages */ +#define LM49453_P0_MICL_REG 0x20 +#define LM49453_P0_MICR_REG 0x21 +#define LM49453_P0_EP_REG 0x24 +#define LM49453_P0_DIS_PKVL_FB_REG 0x25 + +/* Analog Mixer Output Stages */ +#define LM49453_P0_ANALOG_MIXER_ADC_REG 0x2E + +/*ADC or DAC */ +#define LM49453_P0_ADC_DSP_REG 0x30 +#define LM49453_P0_DAC_DSP_REG 0x31 + +/* EFFECTS ENABLES */ +#define LM49453_P0_ADC_FX_ENABLES_REG 0x33 + +/* GPIO */ +#define LM49453_P0_GPIO1_REG 0x38 +#define LM49453_P0_GPIO2_REG 0x39 +#define LM49453_P0_GPIO3_REG 0x3A +#define LM49453_P0_HAP_CTL_REG 0x3B +#define LM49453_P0_HAP_FREQ_PROG_LEFTL_REG 0x3C +#define LM49453_P0_HAP_FREQ_PROG_LEFTH_REG 0x3D +#define LM49453_P0_HAP_FREQ_PROG_RIGHTL_REG 0x3E +#define LM49453_P0_HAP_FREQ_PROG_RIGHTH_REG 0x3F + +/* DIGITAL MIXER */ +#define LM49453_P0_DMIX_CLK_SEL_REG 0x40 +#define LM49453_P0_PORT1_RX_LVL1_REG 0x41 +#define LM49453_P0_PORT1_RX_LVL2_REG 0x42 +#define LM49453_P0_PORT2_RX_LVL_REG 0x43 +#define LM49453_P0_PORT1_TX1_REG 0x44 +#define LM49453_P0_PORT1_TX2_REG 0x45 +#define LM49453_P0_PORT1_TX3_REG 0x46 +#define LM49453_P0_PORT1_TX4_REG 0x47 +#define LM49453_P0_PORT1_TX5_REG 0x48 +#define LM49453_P0_PORT1_TX6_REG 0x49 +#define LM49453_P0_PORT1_TX7_REG 0x4A +#define LM49453_P0_PORT1_TX8_REG 0x4B +#define LM49453_P0_PORT2_TX1_REG 0x4C +#define LM49453_P0_PORT2_TX2_REG 0x4D +#define LM49453_P0_STN_SEL_REG 0x4F +#define LM49453_P0_DACHPL1_REG 0x50 +#define LM49453_P0_DACHPL2_REG 0x51 +#define LM49453_P0_DACHPR1_REG 0x52 +#define LM49453_P0_DACHPR2_REG 0x53 +#define LM49453_P0_DACLOL1_REG 0x54 +#define LM49453_P0_DACLOL2_REG 0x55 +#define LM49453_P0_DACLOR1_REG 0x56 +#define LM49453_P0_DACLOR2_REG 0x57 +#define LM49453_P0_DACLSL1_REG 0x58 +#define LM49453_P0_DACLSL2_REG 0x59 +#define LM49453_P0_DACLSR1_REG 0x5A +#define LM49453_P0_DACLSR2_REG 0x5B +#define LM49453_P0_DACHAL1_REG 0x5C +#define LM49453_P0_DACHAL2_REG 0x5D +#define LM49453_P0_DACHAR1_REG 0x5E +#define LM49453_P0_DACHAR2_REG 0x5F + +/* AUDIO PORT 1 (TDM) */ +#define LM49453_P0_AUDIO_PORT1_BASIC_REG 0x60 +#define LM49453_P0_AUDIO_PORT1_CLK_GEN1_REG 0x61 +#define LM49453_P0_AUDIO_PORT1_CLK_GEN2_REG 0x62 +#define LM49453_P0_AUDIO_PORT1_CLK_GEN3_REG 0x63 +#define LM49453_P0_AUDIO_PORT1_SYNC_RATE_REG 0x64 +#define LM49453_P0_AUDIO_PORT1_SYNC_SDO_SETUP_REG 0x65 +#define LM49453_P0_AUDIO_PORT1_DATA_WIDTH_REG 0x66 +#define LM49453_P0_AUDIO_PORT1_RX_MSB_REG 0x67 +#define LM49453_P0_AUDIO_PORT1_TX_MSB_REG 0x68 +#define LM49453_P0_AUDIO_PORT1_TDM_CHANNELS_REG 0x69 + +/* AUDIO PORT 2 */ +#define LM49453_P0_AUDIO_PORT2_BASIC_REG 0x6A +#define LM49453_P0_AUDIO_PORT2_CLK_GEN1_REG 0x6B +#define LM49453_P0_AUDIO_PORT2_CLK_GEN2_REG 0x6C +#define LM49453_P0_AUDIO_PORT2_SYNC_GEN_REG 0x6D +#define LM49453_P0_AUDIO_PORT2_DATA_WIDTH_REG 0x6E +#define LM49453_P0_AUDIO_PORT2_RX_MODE_REG 0x6F +#define LM49453_P0_AUDIO_PORT2_TX_MODE_REG 0x70 + +/* SAMPLE RATE */ +#define LM49453_P0_PORT1_SR_LSB_REG 0x79 +#define LM49453_P0_PORT1_SR_MSB_REG 0x7A +#define LM49453_P0_PORT2_SR_LSB_REG 0x7B +#define LM49453_P0_PORT2_SR_MSB_REG 0x7C + +/* EFFECTS - HPFs */ +#define LM49453_P0_HPF_REG 0x80 + +/* EFFECTS ADC ALC */ +#define LM49453_P0_ADC_ALC1_REG 0x82 +#define LM49453_P0_ADC_ALC2_REG 0x83 +#define LM49453_P0_ADC_ALC3_REG 0x84 +#define LM49453_P0_ADC_ALC4_REG 0x85 +#define LM49453_P0_ADC_ALC5_REG 0x86 +#define LM49453_P0_ADC_ALC6_REG 0x87 +#define LM49453_P0_ADC_ALC7_REG 0x88 +#define LM49453_P0_ADC_ALC8_REG 0x89 +#define LM49453_P0_DMIC1_LEVELL_REG 0x8A +#define LM49453_P0_DMIC1_LEVELR_REG 0x8B +#define LM49453_P0_DMIC2_LEVELL_REG 0x8C +#define LM49453_P0_DMIC2_LEVELR_REG 0x8D +#define LM49453_P0_ADC_LEVELL_REG 0x8E +#define LM49453_P0_ADC_LEVELR_REG 0x8F +#define LM49453_P0_DAC_HP_LEVELL_REG 0x90 +#define LM49453_P0_DAC_HP_LEVELR_REG 0x91 +#define LM49453_P0_DAC_LO_LEVELL_REG 0x92 +#define LM49453_P0_DAC_LO_LEVELR_REG 0x93 +#define LM49453_P0_DAC_LS_LEVELL_REG 0x94 +#define LM49453_P0_DAC_LS_LEVELR_REG 0x95 +#define LM49453_P0_DAC_HA_LEVELL_REG 0x96 +#define LM49453_P0_DAC_HA_LEVELR_REG 0x97 +#define LM49453_P0_SOFT_MUTE_REG 0x98 +#define LM49453_P0_DMIC_MUTE_CFG_REG 0x99 +#define LM49453_P0_ADC_MUTE_CFG_REG 0x9A +#define LM49453_P0_DAC_MUTE_CFG_REG 0x9B + +/*DIGITAL MIC1 */ +#define LM49453_P0_DIGITAL_MIC1_CONFIG_REG 0xB0 +#define LM49453_P0_DIGITAL_MIC1_DATA_DELAYL_REG 0xB1 +#define LM49453_P0_DIGITAL_MIC1_DATA_DELAYR_REG 0xB2 + +/*DIGITAL MIC2 */ +#define LM49453_P0_DIGITAL_MIC2_CONFIG_REG 0xB3 +#define LM49453_P0_DIGITAL_MIC2_DATA_DELAYL_REG 0xB4 +#define LM49453_P0_DIGITAL_MIC2_DATA_DELAYR_REG 0xB5 + +/* ADC DECIMATOR */ +#define LM49453_P0_ADC_DECIMATOR_REG 0xB6 + +/* DAC CONFIGURE */ +#define LM49453_P0_DAC_CONFIG_REG 0xB7 + +/* SIDETONE */ +#define LM49453_P0_STN_VOL_ADCL_REG 0xB8 +#define LM49453_P0_STN_VOL_ADCR_REG 0xB9 +#define LM49453_P0_STN_VOL_DMIC1L_REG 0xBA +#define LM49453_P0_STN_VOL_DMIC1R_REG 0xBB +#define LM49453_P0_STN_VOL_DMIC2L_REG 0xBC +#define LM49453_P0_STN_VOL_DMIC2R_REG 0xBD + +/* ADC/DAC CLIPPING MONITORS (Read Only/Write to Clear) */ +#define LM49453_P0_ADC_DEC_CLIP_REG 0xC2 +#define LM49453_P0_ADC_HPF_CLIP_REG 0xC3 +#define LM49453_P0_ADC_LVL_CLIP_REG 0xC4 +#define LM49453_P0_DAC_LVL_CLIP_REG 0xC5 + +/* ADC ALC EFFECT MONITORS (Read Only) */ +#define LM49453_P0_ADC_LVLMONL_REG 0xC8 +#define LM49453_P0_ADC_LVLMONR_REG 0xC9 +#define LM49453_P0_ADC_ALCMONL_REG 0xCA +#define LM49453_P0_ADC_ALCMONR_REG 0xCB +#define LM49453_P0_ADC_MUTED_REG 0xCC +#define LM49453_P0_DAC_MUTED_REG 0xCD + +/* HEADSET DETECT */ +#define LM49453_P0_HSD_PPB_LONG_CNT_LIMITL_REG 0xD0 +#define LM49453_P0_HSD_PPB_LONG_CNT_LIMITR_REG 0xD1 +#define LM49453_P0_HSD_PIN3_4_EX_LOOP_CNT_LIMITL_REG 0xD2 +#define LM49453_P0_HSD_PIN3_4_EX_LOOP_CNT_LIMITH_REG 0xD3 +#define LM49453_P0_HSD_TIMEOUT1_REG 0xD4 +#define LM49453_P0_HSD_TIMEOUT2_REG 0xD5 +#define LM49453_P0_HSD_TIMEOUT3_REG 0xD6 +#define LM49453_P0_HSD_PIN3_4_CFG_REG 0xD7 +#define LM49453_P0_HSD_IRQ1_REG 0xD8 +#define LM49453_P0_HSD_IRQ2_REG 0xD9 +#define LM49453_P0_HSD_IRQ3_REG 0xDA +#define LM49453_P0_HSD_IRQ4_REG 0xDB +#define LM49453_P0_HSD_IRQ_MASK1_REG 0xDC +#define LM49453_P0_HSD_IRQ_MASK2_REG 0xDD +#define LM49453_P0_HSD_IRQ_MASK3_REG 0xDE +#define LM49453_P0_HSD_R_HPLL_REG 0xE0 +#define LM49453_P0_HSD_R_HPLH_REG 0xE1 +#define LM49453_P0_HSD_R_HPLU_REG 0xE2 +#define LM49453_P0_HSD_R_HPRL_REG 0xE3 +#define LM49453_P0_HSD_R_HPRH_REG 0xE4 +#define LM49453_P0_HSD_R_HPRU_REG 0xE5 +#define LM49453_P0_HSD_VEL_L_FINALL_REG 0xE6 +#define LM49453_P0_HSD_VEL_L_FINALH_REG 0xE7 +#define LM49453_P0_HSD_VEL_L_FINALU_REG 0xE8 +#define LM49453_P0_HSD_RO_FINALL_REG 0xE9 +#define LM49453_P0_HSD_RO_FINALH_REG 0xEA +#define LM49453_P0_HSD_RO_FINALU_REG 0xEB +#define LM49453_P0_HSD_VMIC_BIAS_FINALL_REG 0xEC +#define LM49453_P0_HSD_VMIC_BIAS_FINALH_REG 0xED +#define LM49453_P0_HSD_VMIC_BIAS_FINALU_REG 0xEE +#define LM49453_P0_HSD_PIN_CONFIG_REG 0xEF +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS1_REG 0xF1 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS2_REG 0xF2 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS3_REG 0xF3 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATEL_REG 0xF4 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATEH_REG 0xF5 + +/* I/O PULLDOWN CONFIG */ +#define LM49453_P0_PULL_CONFIG1_REG 0xF8 +#define LM49453_P0_PULL_CONFIG2_REG 0xF9 +#define LM49453_P0_PULL_CONFIG3_REG 0xFA + +/* RESET */ +#define LM49453_P0_RESET_REG 0xFE + +/* PAGE */ +#define LM49453_PAGE_REG 0xFF + +#define LM49453_MAX_REGISTER (0xFF+1) + +/* LM49453_P0_PMC_SETUP_REG (0x00h) */ +#define LM49453_PMC_SETUP_CHIP_EN (BIT(1)|BIT(0)) +#define LM49453_PMC_SETUP_PLL_EN BIT(2) +#define LM49453_PMC_SETUP_PLL_P2_EN BIT(3) +#define LM49453_PMC_SETUP_PLL_FLL BIT(4) +#define LM49453_PMC_SETUP_MCLK_OVER BIT(5) +#define LM49453_PMC_SETUP_RTC_CLK_OVER BIT(6) +#define LM49453_PMC_SETUP_CHIP_ACTIVE BIT(7) + +/* Chip Enable bits */ +#define LM49453_CHIP_EN_SHUTDOWN 0x00 +#define LM49453_CHIP_EN 0x01 +#define LM49453_CHIP_EN_HSD_DETECT 0x02 +#define LM49453_CHIP_EN_INVALID_HSD 0x03 + +/* LM49453_P0_PLL_CLK_SEL1_REG (0x01h) */ +#define LM49453_CLK_SEL1_MCLK_SEL 0x11 +#define LM49453_CLK_SEL1_RTC_SEL 0x11 +#define LM49453_CLK_SEL1_PORT1_SEL 0x10 +#define LM49453_CLK_SEL1_PORT2_SEL 0x11 + +/* LM49453_P0_PLL_CLK_SEL2_REG (0x02h) */ +#define LM49453_CLK_SEL2_ADC_CLK_SEL 0x38 + +/* LM49453_P0_FLL_REF_FREQL_REG (0x0F) */ +#define LM49453_FLL_REF_FREQ_VAL 0x8ca0001 + +/* LM49453_P0_VCO_TARGETLL_REG (0x11) */ +#define LM49453_VCO_TARGET_VAL 0x8ca0001 + +/* LM49453_P0_ADC_DSP_REG (0x30h) */ +#define LM49453_ADC_DSP_ADC_MUTEL BIT(0) +#define LM49453_ADC_DSP_ADC_MUTER BIT(1) +#define LM49453_ADC_DSP_DMIC1_MUTEL BIT(2) +#define LM49453_ADC_DSP_DMIC1_MUTER BIT(3) +#define LM49453_ADC_DSP_DMIC2_MUTEL BIT(4) +#define LM49453_ADC_DSP_DMIC2_MUTER BIT(5) +#define LM49453_ADC_DSP_MUTE_ALL 0x3F + +/* LM49453_P0_DAC_DSP_REG (0x31h) */ +#define LM49453_DAC_DSP_MUTE_ALL 0xFF + +/* LM49453_P0_AUDIO_PORT1_BASIC_REG (0x60h) */ +#define LM49453_AUDIO_PORT1_BASIC_FMT_MASK (BIT(4)|BIT(3)) +#define LM49453_AUDIO_PORT1_BASIC_CLK_MS BIT(3) +#define LM49453_AUDIO_PORT1_BASIC_SYNC_MS BIT(4) + +/* LM49453_P0_RESET_REG (0xFEh) */ +#define LM49453_RESET_REG_RST BIT(0) + +/* Page select register bits (0xFF) */ +#define LM49453_PAGE0_SELECT 0x0 +#define LM49453_PAGE1_SELECT 0x1 + +/* LM49453_P0_HSD_PIN3_4_CFG_REG (Jack Pin config - 0xD7) */ +#define LM49453_JACK_DISABLE 0x00 +#define LM49453_JACK_CONFIG1 0x01 +#define LM49453_JACK_CONFIG2 0x02 +#define LM49453_JACK_CONFIG3 0x03 +#define LM49453_JACK_CONFIG4 0x04 +#define LM49453_JACK_CONFIG5 0x05 + +/* Page 1 REGISTERS */ + +/* SIDETONE */ +#define LM49453_P1_SIDETONE_SA0L_REG 0x80 +#define LM49453_P1_SIDETONE_SA0H_REG 0x81 +#define LM49453_P1_SIDETONE_SAB0U_REG 0x82 +#define LM49453_P1_SIDETONE_SB0L_REG 0x83 +#define LM49453_P1_SIDETONE_SB0H_REG 0x84 +#define LM49453_P1_SIDETONE_SH0L_REG 0x85 +#define LM49453_P1_SIDETONE_SH0H_REG 0x86 +#define LM49453_P1_SIDETONE_SH0U_REG 0x87 +#define LM49453_P1_SIDETONE_SA1L_REG 0x88 +#define LM49453_P1_SIDETONE_SA1H_REG 0x89 +#define LM49453_P1_SIDETONE_SAB1U_REG 0x8A +#define LM49453_P1_SIDETONE_SB1L_REG 0x8B +#define LM49453_P1_SIDETONE_SB1H_REG 0x8C +#define LM49453_P1_SIDETONE_SH1L_REG 0x8D +#define LM49453_P1_SIDETONE_SH1H_REG 0x8E +#define LM49453_P1_SIDETONE_SH1U_REG 0x8F +#define LM49453_P1_SIDETONE_SA2L_REG 0x90 +#define LM49453_P1_SIDETONE_SA2H_REG 0x91 +#define LM49453_P1_SIDETONE_SAB2U_REG 0x92 +#define LM49453_P1_SIDETONE_SB2L_REG 0x93 +#define LM49453_P1_SIDETONE_SB2H_REG 0x94 +#define LM49453_P1_SIDETONE_SH2L_REG 0x95 +#define LM49453_P1_SIDETONE_SH2H_REG 0x96 +#define LM49453_P1_SIDETONE_SH2U_REG 0x97 +#define LM49453_P1_SIDETONE_SA3L_REG 0x98 +#define LM49453_P1_SIDETONE_SA3H_REG 0x99 +#define LM49453_P1_SIDETONE_SAB3U_REG 0x9A +#define LM49453_P1_SIDETONE_SB3L_REG 0x9B +#define LM49453_P1_SIDETONE_SB3H_REG 0x9C +#define LM49453_P1_SIDETONE_SH3L_REG 0x9D +#define LM49453_P1_SIDETONE_SH3H_REG 0x9E +#define LM49453_P1_SIDETONE_SH3U_REG 0x9F +#define LM49453_P1_SIDETONE_SA4L_REG 0xA0 +#define LM49453_P1_SIDETONE_SA4H_REG 0xA1 +#define LM49453_P1_SIDETONE_SAB4U_REG 0xA2 +#define LM49453_P1_SIDETONE_SB4L_REG 0xA3 +#define LM49453_P1_SIDETONE_SB4H_REG 0xA4 +#define LM49453_P1_SIDETONE_SH4L_REG 0xA5 +#define LM49453_P1_SIDETONE_SH4H_REG 0xA6 +#define LM49453_P1_SIDETONE_SH4U_REG 0xA7 +#define LM49453_P1_SIDETONE_SA5L_REG 0xA8 +#define LM49453_P1_SIDETONE_SA5H_REG 0xA9 +#define LM49453_P1_SIDETONE_SAB5U_REG 0xAA +#define LM49453_P1_SIDETONE_SB5L_REG 0xAB +#define LM49453_P1_SIDETONE_SB5H_REG 0xAC +#define LM49453_P1_SIDETONE_SH5L_REG 0xAD +#define LM49453_P1_SIDETONE_SH5H_REG 0xAE +#define LM49453_P1_SIDETONE_SH5U_REG 0xAF + +/* CHARGE PUMP CONFIG */ +#define LM49453_P1_CP_CONFIG1_REG 0xB0 +#define LM49453_P1_CP_CONFIG2_REG 0xB1 +#define LM49453_P1_CP_CONFIG3_REG 0xB2 +#define LM49453_P1_CP_CONFIG4_REG 0xB3 +#define LM49453_P1_CP_LA_VTH1L_REG 0xB4 +#define LM49453_P1_CP_LA_VTH1M_REG 0xB5 +#define LM49453_P1_CP_LA_VTH2L_REG 0xB6 +#define LM49453_P1_CP_LA_VTH2M_REG 0xB7 +#define LM49453_P1_CP_LA_VTH3L_REG 0xB8 +#define LM49453_P1_CP_LA_VTH3H_REG 0xB9 +#define LM49453_P1_CP_CLK_DIV_REG 0xBA + +/* DAC */ +#define LM49453_P1_DAC_CHOP_REG 0xC0 + +#define LM49453_CLK_SRC_MCLK 1 +#endif diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 0bb511a0388d..7cd508e16a5c 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -24,6 +24,7 @@ #include <linux/slab.h> #include <asm/div64.h> #include <sound/max98095.h> +#include <sound/jack.h> #include "max98095.h" enum max98095_type { @@ -51,6 +52,8 @@ struct max98095_priv { u8 lin_state; unsigned int mic1pre; unsigned int mic2pre; + struct snd_soc_jack *headphone_jack; + struct snd_soc_jack *mic_jack; }; static const u8 max98095_reg_def[M98095_REG_CNT] = { @@ -2173,9 +2176,126 @@ static void max98095_handle_pdata(struct snd_soc_codec *codec) max98095_handle_bq_pdata(codec); } +static irqreturn_t max98095_report_jack(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + int hp_report = 0; + int mic_report = 0; + + /* Read the Jack Status Register */ + value = snd_soc_read(codec, M98095_007_JACK_AUTO_STS); + + /* If ddone is not set, then detection isn't finished yet */ + if ((value & M98095_DDONE) == 0) + return IRQ_NONE; + + /* if hp, check its bit, and if set, clear it */ + if ((value & M98095_HP_IN || value & M98095_LO_IN) && + max98095->headphone_jack) + hp_report |= SND_JACK_HEADPHONE; + + /* if mic, check its bit, and if set, clear it */ + if ((value & M98095_MIC_IN) && max98095->mic_jack) + mic_report |= SND_JACK_MICROPHONE; + + if (max98095->headphone_jack == max98095->mic_jack) { + snd_soc_jack_report(max98095->headphone_jack, + hp_report | mic_report, + SND_JACK_HEADSET); + } else { + if (max98095->headphone_jack) + snd_soc_jack_report(max98095->headphone_jack, + hp_report, SND_JACK_HEADPHONE); + if (max98095->mic_jack) + snd_soc_jack_report(max98095->mic_jack, + mic_report, SND_JACK_MICROPHONE); + } + + return IRQ_HANDLED; +} + +static int max98095_jack_detect_enable(struct snd_soc_codec *codec) +{ + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + int detect_enable = M98095_JDEN; + unsigned int slew = M98095_DEFAULT_SLEW_DELAY; + + if (max98095->pdata->jack_detect_pin5en) + detect_enable |= M98095_PIN5EN; + + if (max98095->pdata->jack_detect_delay) + slew = max98095->pdata->jack_detect_delay; + + ret = snd_soc_write(codec, M98095_08E_JACK_DC_SLEW, slew); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + /* configure auto detection to be enabled */ + ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, detect_enable); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + return ret; +} + +static int max98095_jack_detect_disable(struct snd_soc_codec *codec) +{ + int ret = 0; + + /* configure auto detection to be disabled */ + ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, 0x0); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + return ret; +} + +int max98095_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack) +{ + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = to_i2c_client(codec->dev); + int ret = 0; + + max98095->headphone_jack = hp_jack; + max98095->mic_jack = mic_jack; + + /* only progress if we have at least 1 jack pointer */ + if (!hp_jack && !mic_jack) + return -EINVAL; + + max98095_jack_detect_enable(codec); + + /* enable interrupts for headphone jack detection */ + ret = snd_soc_update_bits(codec, M98095_013_JACK_INT_EN, + M98095_IDDONE, M98095_IDDONE); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg jack irqs %d\n", ret); + return ret; + } + + max98095_report_jack(client->irq, codec); + return 0; +} +EXPORT_SYMBOL_GPL(max98095_jack_detect); + #ifdef CONFIG_PM static int max98095_suspend(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + + if (max98095->headphone_jack || max98095->mic_jack) + max98095_jack_detect_disable(codec); + max98095_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -2183,8 +2303,16 @@ static int max98095_suspend(struct snd_soc_codec *codec) static int max98095_resume(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = to_i2c_client(codec->dev); + max98095_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (max98095->headphone_jack || max98095->mic_jack) { + max98095_jack_detect_enable(codec); + max98095_report_jack(client->irq, codec); + } + return 0; } #else @@ -2227,6 +2355,7 @@ static int max98095_probe(struct snd_soc_codec *codec) { struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); struct max98095_cdata *cdata; + struct i2c_client *client; int ret = 0; ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); @@ -2238,6 +2367,8 @@ static int max98095_probe(struct snd_soc_codec *codec) /* reset the codec, the DSP core, and disable all interrupts */ max98095_reset(codec); + client = to_i2c_client(codec->dev); + /* initialize private data */ max98095->sysclk = (unsigned)-1; @@ -2266,11 +2397,23 @@ static int max98095_probe(struct snd_soc_codec *codec) max98095->mic1pre = 0; max98095->mic2pre = 0; + if (client->irq) { + /* register an audio interrupt */ + ret = request_threaded_irq(client->irq, NULL, + max98095_report_jack, + IRQF_TRIGGER_FALLING | IRQF_TRIGGER_RISING, + "max98095", codec); + if (ret) { + dev_err(codec->dev, "Failed to request IRQ: %d\n", ret); + goto err_access; + } + } + ret = snd_soc_read(codec, M98095_0FF_REV_ID); if (ret < 0) { dev_err(codec->dev, "Failure reading hardware revision: %d\n", ret); - goto err_access; + goto err_irq; } dev_info(codec->dev, "Hardware revision: %c\n", ret - 0x40 + 'A'); @@ -2306,14 +2449,28 @@ static int max98095_probe(struct snd_soc_codec *codec) max98095_add_widgets(codec); + return 0; + +err_irq: + if (client->irq) + free_irq(client->irq, codec); err_access: return ret; } static int max98095_remove(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = to_i2c_client(codec->dev); + max98095_set_bias_level(codec, SND_SOC_BIAS_OFF); + if (max98095->headphone_jack || max98095->mic_jack) + max98095_jack_detect_disable(codec); + + if (client->irq) + free_irq(client->irq, codec); + return 0; } diff --git a/sound/soc/codecs/max98095.h b/sound/soc/codecs/max98095.h index 891584a0eb03..2ebbe4e894bf 100644 --- a/sound/soc/codecs/max98095.h +++ b/sound/soc/codecs/max98095.h @@ -175,11 +175,23 @@ /* MAX98095 Registers Bit Fields */ +/* M98095_007_JACK_AUTO_STS */ + #define M98095_MIC_IN (1<<3) + #define M98095_LO_IN (1<<5) + #define M98095_HP_IN (1<<6) + #define M98095_DDONE (1<<7) + /* M98095_00F_HOST_CFG */ #define M98095_SEG (1<<0) #define M98095_XTEN (1<<1) #define M98095_MDLLEN (1<<2) +/* M98095_013_JACK_INT_EN */ + #define M98095_IMIC_IN (1<<3) + #define M98095_ILO_IN (1<<5) + #define M98095_IHP_IN (1<<6) + #define M98095_IDDONE (1<<7) + /* M98095_027_DAI1_CLKMODE, M98095_031_DAI2_CLKMODE, M98095_03B_DAI3_CLKMODE */ #define M98095_CLKMODE_MASK 0xFF @@ -255,6 +267,10 @@ #define M98095_EQ2EN (1<<1) #define M98095_EQ1EN (1<<0) +/* M98095_089_JACK_DET_AUTO */ + #define M98095_PIN5EN (1<<2) + #define M98095_JDEN (1<<7) + /* M98095_090_PWR_EN_IN */ #define M98095_INEN (1<<7) #define M98095_MB2EN (1<<3) @@ -296,4 +312,10 @@ #define M98095_174_DAI1_BQ_BASE 0x74 #define M98095_17E_DAI2_BQ_BASE 0x7E +/* Default Delay used in Slew Rate Calculation for Jack detection */ +#define M98095_DEFAULT_SLEW_DELAY 0x18 + +extern int max98095_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack); + #endif diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c new file mode 100644 index 000000000000..6276e352125f --- /dev/null +++ b/sound/soc/codecs/mc13783.c @@ -0,0 +1,786 @@ +/* + * Copyright 2008 Juergen Beisert, kernel@pengutronix.de + * Copyright 2009 Sascha Hauer, s.hauer@pengutronix.de + * Copyright 2012 Philippe Retornaz, philippe.retornaz@epfl.ch + * + * Initial development of this code was funded by + * Phytec Messtechnik GmbH, http://www.phytec.de + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, + * MA 02110-1301, USA. + */ +#include <linux/module.h> +#include <linux/device.h> +#include <linux/mfd/mc13xxx.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/soc-dapm.h> + +#include "mc13783.h" + +#define MC13783_AUDIO_RX0 36 +#define MC13783_AUDIO_RX1 37 +#define MC13783_AUDIO_TX 38 +#define MC13783_SSI_NETWORK 39 +#define MC13783_AUDIO_CODEC 40 +#define MC13783_AUDIO_DAC 41 + +#define AUDIO_RX0_ALSPEN (1 << 5) +#define AUDIO_RX0_ALSPSEL (1 << 7) +#define AUDIO_RX0_ADDCDC (1 << 21) +#define AUDIO_RX0_ADDSTDC (1 << 22) +#define AUDIO_RX0_ADDRXIN (1 << 23) + +#define AUDIO_RX1_PGARXEN (1 << 0); +#define AUDIO_RX1_PGASTEN (1 << 5) +#define AUDIO_RX1_ARXINEN (1 << 10) + +#define AUDIO_TX_AMC1REN (1 << 5) +#define AUDIO_TX_AMC1LEN (1 << 7) +#define AUDIO_TX_AMC2EN (1 << 9) +#define AUDIO_TX_ATXINEN (1 << 11) +#define AUDIO_TX_RXINREC (1 << 13) + +#define SSI_NETWORK_CDCTXRXSLOT(x) (((x) & 0x3) << 2) +#define SSI_NETWORK_CDCTXSECSLOT(x) (((x) & 0x3) << 4) +#define SSI_NETWORK_CDCRXSECSLOT(x) (((x) & 0x3) << 6) +#define SSI_NETWORK_CDCRXSECGAIN(x) (((x) & 0x3) << 8) +#define SSI_NETWORK_CDCSUMGAIN(x) (1 << 10) +#define SSI_NETWORK_CDCFSDLY(x) (1 << 11) +#define SSI_NETWORK_DAC_SLOTS_8 (1 << 12) +#define SSI_NETWORK_DAC_SLOTS_4 (2 << 12) +#define SSI_NETWORK_DAC_SLOTS_2 (3 << 12) +#define SSI_NETWORK_DAC_SLOT_MASK (3 << 12) +#define SSI_NETWORK_DAC_RXSLOT_0_1 (0 << 14) +#define SSI_NETWORK_DAC_RXSLOT_2_3 (1 << 14) +#define SSI_NETWORK_DAC_RXSLOT_4_5 (2 << 14) +#define SSI_NETWORK_DAC_RXSLOT_6_7 (3 << 14) +#define SSI_NETWORK_DAC_RXSLOT_MASK (3 << 14) +#define SSI_NETWORK_STDCRXSECSLOT(x) (((x) & 0x3) << 16) +#define SSI_NETWORK_STDCRXSECGAIN(x) (((x) & 0x3) << 18) +#define SSI_NETWORK_STDCSUMGAIN (1 << 20) + +/* + * MC13783_AUDIO_CODEC and MC13783_AUDIO_DAC mostly share the same + * register layout + */ +#define AUDIO_SSI_SEL (1 << 0) +#define AUDIO_CLK_SEL (1 << 1) +#define AUDIO_CSM (1 << 2) +#define AUDIO_BCL_INV (1 << 3) +#define AUDIO_CFS_INV (1 << 4) +#define AUDIO_CFS(x) (((x) & 0x3) << 5) +#define AUDIO_CLK(x) (((x) & 0x7) << 7) +#define AUDIO_C_EN (1 << 11) +#define AUDIO_C_CLK_EN (1 << 12) +#define AUDIO_C_RESET (1 << 15) + +#define AUDIO_CODEC_CDCFS8K16K (1 << 10) +#define AUDIO_DAC_CFS_DLY_B (1 << 10) + +struct mc13783_priv { + struct snd_soc_codec codec; + struct mc13xxx *mc13xxx; + + enum mc13783_ssi_port adc_ssi_port; + enum mc13783_ssi_port dac_ssi_port; +}; + +static unsigned int mc13783_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int value = 0; + + mc13xxx_lock(priv->mc13xxx); + + mc13xxx_reg_read(priv->mc13xxx, reg, &value); + + mc13xxx_unlock(priv->mc13xxx); + + return value; +} + +static int mc13783_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + mc13xxx_lock(priv->mc13xxx); + + ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); + + mc13xxx_unlock(priv->mc13xxx); + + return ret; +} + +/* Mapping between sample rates and register value */ +static unsigned int mc13783_rates[] = { + 8000, 11025, 12000, 16000, + 22050, 24000, 32000, 44100, + 48000, 64000, 96000 +}; + +static int mc13783_pcm_hw_params_dac(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + unsigned int rate = params_rate(params); + int i; + + for (i = 0; i < ARRAY_SIZE(mc13783_rates); i++) { + if (rate == mc13783_rates[i]) { + snd_soc_update_bits(codec, MC13783_AUDIO_DAC, + 0xf << 17, i << 17); + return 0; + } + } + + return -EINVAL; +} + +static int mc13783_pcm_hw_params_codec(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + unsigned int rate = params_rate(params); + unsigned int val; + + switch (rate) { + case 8000: + val = 0; + break; + case 16000: + val = AUDIO_CODEC_CDCFS8K16K; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, MC13783_AUDIO_CODEC, AUDIO_CODEC_CDCFS8K16K, + val); + + return 0; +} + +static int mc13783_pcm_hw_params_sync(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return mc13783_pcm_hw_params_dac(substream, params, dai); + else + return mc13783_pcm_hw_params_codec(substream, params, dai); +} + +static int mc13783_set_fmt(struct snd_soc_dai *dai, unsigned int fmt, + unsigned int reg) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + unsigned int mask = AUDIO_CFS(3) | AUDIO_BCL_INV | AUDIO_CFS_INV | + AUDIO_CSM | AUDIO_C_CLK_EN | AUDIO_C_RESET; + + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val |= AUDIO_CFS(2); + break; + case SND_SOC_DAIFMT_DSP_A: + val |= AUDIO_CFS(1); + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + val |= AUDIO_BCL_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + val |= AUDIO_BCL_INV | AUDIO_CFS_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + val |= AUDIO_CFS_INV; + break; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + val |= AUDIO_C_CLK_EN; + break; + case SND_SOC_DAIFMT_CBS_CFS: + val |= AUDIO_CSM; + break; + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + return -EINVAL; + } + + val |= AUDIO_C_RESET; + + snd_soc_update_bits(codec, reg, mask, val); + + return 0; +} + +static int mc13783_set_fmt_async(struct snd_soc_dai *dai, unsigned int fmt) +{ + if (dai->id == MC13783_ID_STEREO_DAC) + return mc13783_set_fmt(dai, fmt, MC13783_AUDIO_DAC); + else + return mc13783_set_fmt(dai, fmt, MC13783_AUDIO_CODEC); +} + +static int mc13783_set_fmt_sync(struct snd_soc_dai *dai, unsigned int fmt) +{ + int ret; + + ret = mc13783_set_fmt(dai, fmt, MC13783_AUDIO_DAC); + if (ret) + return ret; + + /* + * In synchronous mode force the voice codec into slave mode + * so that the clock / framesync from the stereo DAC is used + */ + fmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + fmt |= SND_SOC_DAIFMT_CBS_CFS; + ret = mc13783_set_fmt(dai, fmt, MC13783_AUDIO_CODEC); + + return ret; +} + +static int mc13783_sysclk[] = { + 13000000, + 15360000, + 16800000, + -1, + 26000000, + -1, /* 12000000, invalid for voice codec */ + -1, /* 3686400, invalid for voice codec */ + 33600000, +}; + +static int mc13783_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir, + unsigned int reg) +{ + struct snd_soc_codec *codec = dai->codec; + int clk; + unsigned int val = 0; + unsigned int mask = AUDIO_CLK(0x7) | AUDIO_CLK_SEL; + + for (clk = 0; clk < ARRAY_SIZE(mc13783_sysclk); clk++) { + if (mc13783_sysclk[clk] < 0) + continue; + if (mc13783_sysclk[clk] == freq) + break; + } + + if (clk == ARRAY_SIZE(mc13783_sysclk)) + return -EINVAL; + + if (clk_id == MC13783_CLK_CLIB) + val |= AUDIO_CLK_SEL; + + val |= AUDIO_CLK(clk); + + snd_soc_update_bits(codec, reg, mask, val); + + return 0; +} + +static int mc13783_set_sysclk_dac(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_DAC); +} + +static int mc13783_set_sysclk_codec(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_CODEC); +} + +static int mc13783_set_sysclk_sync(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + int ret; + + ret = mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_DAC); + if (ret) + return ret; + + return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_CODEC); +} + +static int mc13783_set_tdm_slot_dac(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, + int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + unsigned int mask = SSI_NETWORK_DAC_SLOT_MASK | + SSI_NETWORK_DAC_RXSLOT_MASK; + + switch (slots) { + case 2: + val |= SSI_NETWORK_DAC_SLOTS_2; + break; + case 4: + val |= SSI_NETWORK_DAC_SLOTS_4; + break; + case 8: + val |= SSI_NETWORK_DAC_SLOTS_8; + break; + default: + return -EINVAL; + } + + switch (rx_mask) { + case 0xfffffffc: + val |= SSI_NETWORK_DAC_RXSLOT_0_1; + break; + case 0xfffffff3: + val |= SSI_NETWORK_DAC_RXSLOT_2_3; + break; + case 0xffffffcf: + val |= SSI_NETWORK_DAC_RXSLOT_4_5; + break; + case 0xffffff3f: + val |= SSI_NETWORK_DAC_RXSLOT_6_7; + break; + default: + return -EINVAL; + }; + + snd_soc_update_bits(codec, MC13783_SSI_NETWORK, mask, val); + + return 0; +} + +static int mc13783_set_tdm_slot_codec(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, + int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + unsigned int mask = 0x3f; + + if (slots != 4) + return -EINVAL; + + if (tx_mask != 0xfffffffc) + return -EINVAL; + + val |= (0x00 << 2); /* primary timeslot RX/TX(?) is 0 */ + val |= (0x01 << 4); /* secondary timeslot TX is 1 */ + + snd_soc_update_bits(codec, MC13783_SSI_NETWORK, mask, val); + + return 0; +} + +static int mc13783_set_tdm_slot_sync(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, + int slot_width) +{ + int ret; + + ret = mc13783_set_tdm_slot_dac(dai, tx_mask, rx_mask, slots, + slot_width); + if (ret) + return ret; + + ret = mc13783_set_tdm_slot_codec(dai, tx_mask, rx_mask, slots, + slot_width); + + return ret; +} + +static const struct snd_kcontrol_new mc1l_amp_ctl = + SOC_DAPM_SINGLE("Switch", 38, 7, 1, 0); + +static const struct snd_kcontrol_new mc1r_amp_ctl = + SOC_DAPM_SINGLE("Switch", 38, 5, 1, 0); + +static const struct snd_kcontrol_new mc2_amp_ctl = + SOC_DAPM_SINGLE("Switch", 38, 9, 1, 0); + +static const struct snd_kcontrol_new atx_amp_ctl = + SOC_DAPM_SINGLE("Switch", 38, 11, 1, 0); + + +/* Virtual mux. The chip does the input selection automatically + * as soon as we enable one input. */ +static const char * const adcl_enum_text[] = { + "MC1L", "RXINL", +}; + +static const struct soc_enum adcl_enum = + SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcl_enum_text), adcl_enum_text); + +static const struct snd_kcontrol_new left_input_mux = + SOC_DAPM_ENUM_VIRT("Route", adcl_enum); + +static const char * const adcr_enum_text[] = { + "MC1R", "MC2", "RXINR", "TXIN", +}; + +static const struct soc_enum adcr_enum = + SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcr_enum_text), adcr_enum_text); + +static const struct snd_kcontrol_new right_input_mux = + SOC_DAPM_ENUM_VIRT("Route", adcr_enum); + +static const struct snd_kcontrol_new samp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 3, 1, 0); + +static const struct snd_kcontrol_new lamp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 5, 1, 0); + +static const struct snd_kcontrol_new hlamp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 10, 1, 0); + +static const struct snd_kcontrol_new hramp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 9, 1, 0); + +static const struct snd_kcontrol_new llamp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 16, 1, 0); + +static const struct snd_kcontrol_new lramp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 15, 1, 0); + +static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { +/* Input */ + SND_SOC_DAPM_INPUT("MC1LIN"), + SND_SOC_DAPM_INPUT("MC1RIN"), + SND_SOC_DAPM_INPUT("MC2IN"), + SND_SOC_DAPM_INPUT("RXINR"), + SND_SOC_DAPM_INPUT("RXINL"), + SND_SOC_DAPM_INPUT("TXIN"), + + SND_SOC_DAPM_SUPPLY("MC1 Bias", 38, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC2 Bias", 38, 1, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("MC1L Amp", 38, 7, 0, &mc1l_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1R Amp", 38, 5, 0, &mc1r_amp_ctl), + SND_SOC_DAPM_SWITCH("MC2 Amp", 38, 9, 0, &mc2_amp_ctl), + SND_SOC_DAPM_SWITCH("TXIN Amp", 38, 11, 0, &atx_amp_ctl), + + SND_SOC_DAPM_VIRT_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0, + &left_input_mux), + SND_SOC_DAPM_VIRT_MUX("PGA Right Input Mux", SND_SOC_NOPM, 0, 0, + &right_input_mux), + + SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_ADC("ADC", "Capture", 40, 11, 0), + SND_SOC_DAPM_SUPPLY("ADC_Reset", 40, 15, 0, NULL, 0), + +/* Output */ + SND_SOC_DAPM_SUPPLY("DAC_E", 41, 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_Reset", 41, 15, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("RXOUTL"), + SND_SOC_DAPM_OUTPUT("RXOUTR"), + SND_SOC_DAPM_OUTPUT("HSL"), + SND_SOC_DAPM_OUTPUT("HSR"), + SND_SOC_DAPM_OUTPUT("LSP"), + SND_SOC_DAPM_OUTPUT("SP"), + + SND_SOC_DAPM_SWITCH("Speaker Amp", 36, 3, 0, &samp_ctl), + SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), + SND_SOC_DAPM_SWITCH("Headset Amp Left", 36, 10, 0, &hlamp_ctl), + SND_SOC_DAPM_SWITCH("Headset Amp Right", 36, 9, 0, &hramp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Left", 36, 16, 0, &llamp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Right", 36, 15, 0, &lramp_ctl), + SND_SOC_DAPM_DAC("DAC", "Playback", 36, 22, 0), + SND_SOC_DAPM_PGA("DAC PGA", 37, 5, 0, NULL, 0), +}; + +static struct snd_soc_dapm_route mc13783_routes[] = { +/* Input */ + { "MC1L Amp", NULL, "MC1LIN"}, + { "MC1R Amp", NULL, "MC1RIN" }, + { "MC2 Amp", NULL, "MC2IN" }, + { "TXIN Amp", NULL, "TXIN"}, + + { "PGA Left Input Mux", "MC1L", "MC1L Amp" }, + { "PGA Left Input Mux", "RXINL", "RXINL"}, + { "PGA Right Input Mux", "MC1R", "MC1R Amp" }, + { "PGA Right Input Mux", "MC2", "MC2 Amp"}, + { "PGA Right Input Mux", "TXIN", "TXIN Amp"}, + { "PGA Right Input Mux", "RXINR", "RXINR"}, + + { "PGA Left Input", NULL, "PGA Left Input Mux"}, + { "PGA Right Input", NULL, "PGA Right Input Mux"}, + + { "ADC", NULL, "PGA Left Input"}, + { "ADC", NULL, "PGA Right Input"}, + { "ADC", NULL, "ADC_Reset"}, + +/* Output */ + { "HSL", NULL, "Headset Amp Left" }, + { "HSR", NULL, "Headset Amp Right"}, + { "RXOUTL", NULL, "Line out Amp Left"}, + { "RXOUTR", NULL, "Line out Amp Right"}, + { "SP", NULL, "Speaker Amp"}, + { "Speaker Amp", NULL, "DAC PGA"}, + { "LSP", NULL, "DAC PGA"}, + { "Headset Amp Left", NULL, "DAC PGA"}, + { "Headset Amp Right", NULL, "DAC PGA"}, + { "Line out Amp Left", NULL, "DAC PGA"}, + { "Line out Amp Right", NULL, "DAC PGA"}, + { "DAC PGA", NULL, "DAC"}, + { "DAC", NULL, "DAC_E"}, +}; + +static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix", + "Mono", "Mono Mix"}; + +static const struct soc_enum mc13783_enum_3d_mixer = + SOC_ENUM_SINGLE(MC13783_AUDIO_RX1, 16, ARRAY_SIZE(mc13783_3d_mixer), + mc13783_3d_mixer); + +static struct snd_kcontrol_new mc13783_control_list[] = { + SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0), + SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0), + SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), + SOC_ENUM("3D Control", mc13783_enum_3d_mixer), +}; + +static int mc13783_probe(struct snd_soc_codec *codec) +{ + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + + mc13xxx_lock(priv->mc13xxx); + + /* these are the reset values */ + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX0, 0x25893); + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX1, 0x00d35A); + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_TX, 0x420000); + mc13xxx_reg_write(priv->mc13xxx, MC13783_SSI_NETWORK, 0x013060); + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_CODEC, 0x180027); + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_DAC, 0x0e0004); + + if (priv->adc_ssi_port == MC13783_SSI1_PORT) + mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC, + AUDIO_SSI_SEL, 0); + else + mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC, + 0, AUDIO_SSI_SEL); + + if (priv->dac_ssi_port == MC13783_SSI1_PORT) + mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC, + AUDIO_SSI_SEL, 0); + else + mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC, + 0, AUDIO_SSI_SEL); + + mc13xxx_unlock(priv->mc13xxx); + + return 0; +} + +static int mc13783_remove(struct snd_soc_codec *codec) +{ + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + + mc13xxx_lock(priv->mc13xxx); + + /* Make sure VAUDIOON is off */ + mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_RX0, 0x3, 0); + + mc13xxx_unlock(priv->mc13xxx); + + return 0; +} + +#define MC13783_RATES_RECORD (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) + +#define MC13783_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops mc13783_ops_dac = { + .hw_params = mc13783_pcm_hw_params_dac, + .set_fmt = mc13783_set_fmt_async, + .set_sysclk = mc13783_set_sysclk_dac, + .set_tdm_slot = mc13783_set_tdm_slot_dac, +}; + +static struct snd_soc_dai_ops mc13783_ops_codec = { + .hw_params = mc13783_pcm_hw_params_codec, + .set_fmt = mc13783_set_fmt_async, + .set_sysclk = mc13783_set_sysclk_codec, + .set_tdm_slot = mc13783_set_tdm_slot_codec, +}; + +/* + * The mc13783 has two SSI ports, both of them can be routed either + * to the voice codec or the stereo DAC. When two different SSI ports + * are used for the voice codec and the stereo DAC we can do different + * formats and sysclock settings for playback and capture + * (mc13783-hifi-playback and mc13783-hifi-capture). Using the same port + * forces us to use symmetric rates (mc13783-hifi). + */ +static struct snd_soc_dai_driver mc13783_dai_async[] = { + { + .name = "mc13783-hifi-playback", + .id = MC13783_ID_STEREO_DAC, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = MC13783_FORMATS, + }, + .ops = &mc13783_ops_dac, + }, { + .name = "mc13783-hifi-capture", + .id = MC13783_ID_STEREO_CODEC, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MC13783_RATES_RECORD, + .formats = MC13783_FORMATS, + }, + .ops = &mc13783_ops_codec, + }, +}; + +static struct snd_soc_dai_ops mc13783_ops_sync = { + .hw_params = mc13783_pcm_hw_params_sync, + .set_fmt = mc13783_set_fmt_sync, + .set_sysclk = mc13783_set_sysclk_sync, + .set_tdm_slot = mc13783_set_tdm_slot_sync, +}; + +static struct snd_soc_dai_driver mc13783_dai_sync[] = { + { + .name = "mc13783-hifi", + .id = MC13783_ID_SYNC, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = MC13783_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MC13783_RATES_RECORD, + .formats = MC13783_FORMATS, + }, + .ops = &mc13783_ops_sync, + .symmetric_rates = 1, + } +}; + +static struct snd_soc_codec_driver soc_codec_dev_mc13783 = { + .probe = mc13783_probe, + .remove = mc13783_remove, + .read = mc13783_read, + .write = mc13783_write, + .controls = mc13783_control_list, + .num_controls = ARRAY_SIZE(mc13783_control_list), + .dapm_widgets = mc13783_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mc13783_dapm_widgets), + .dapm_routes = mc13783_routes, + .num_dapm_routes = ARRAY_SIZE(mc13783_routes), +}; + +static int mc13783_codec_probe(struct platform_device *pdev) +{ + struct mc13xxx *mc13xxx; + struct mc13783_priv *priv; + struct mc13xxx_codec_platform_data *pdata = pdev->dev.platform_data; + int ret; + + mc13xxx = dev_get_drvdata(pdev->dev.parent); + + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + dev_set_drvdata(&pdev->dev, priv); + priv->mc13xxx = mc13xxx; + if (pdata) { + priv->adc_ssi_port = pdata->adc_ssi_port; + priv->dac_ssi_port = pdata->dac_ssi_port; + } else { + priv->adc_ssi_port = MC13783_SSI1_PORT; + priv->dac_ssi_port = MC13783_SSI2_PORT; + } + + if (priv->adc_ssi_port == priv->dac_ssi_port) + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, + mc13783_dai_sync, ARRAY_SIZE(mc13783_dai_sync)); + else + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, + mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async)); + + if (ret) + goto err_register_codec; + + return 0; + +err_register_codec: + dev_err(&pdev->dev, "register codec failed with %d\n", ret); + + return ret; +} + +static int mc13783_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + + return 0; +} + +static struct platform_driver mc13783_codec_driver = { + .driver = { + .name = "mc13783-codec", + .owner = THIS_MODULE, + }, + .probe = mc13783_codec_probe, + .remove = __devexit_p(mc13783_codec_remove), +}; + +module_platform_driver(mc13783_codec_driver); + +MODULE_DESCRIPTION("ASoC MC13783 driver"); +MODULE_AUTHOR("Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>"); +MODULE_AUTHOR("Philippe Retornaz <philippe.retornaz@epfl.ch>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/mc13783.h b/sound/soc/codecs/mc13783.h new file mode 100644 index 000000000000..3a6d1993a217 --- /dev/null +++ b/sound/soc/codecs/mc13783.h @@ -0,0 +1,28 @@ +/* + * Copyright 2008 Juergen Beisert, kernel@pengutronix.de + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software Foundation, Inc. + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. + */ + +#ifndef MC13783_MIXER_H +#define MC13783_MIXER_H + +#define MC13783_CLK_CLIA 1 +#define MC13783_CLK_CLIB 2 + +#define MC13783_ID_STEREO_DAC 1 +#define MC13783_ID_STEREO_CODEC 2 +#define MC13783_ID_SYNC 3 + +#endif /* MC13783_MIXER_H */ diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c new file mode 100644 index 000000000000..96aa5fa05160 --- /dev/null +++ b/sound/soc/codecs/ml26124.c @@ -0,0 +1,678 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "ml26124.h" + +#define DVOL_CTL_DVMUTE_ON BIT(4) /* Digital volume MUTE On */ +#define DVOL_CTL_DVMUTE_OFF 0 /* Digital volume MUTE Off */ +#define ML26124_SAI_NO_DELAY BIT(1) +#define ML26124_SAI_FRAME_SYNC (BIT(5) | BIT(0)) /* For mono (Telecodec) */ +#define ML26134_CACHESIZE 212 +#define ML26124_VMID BIT(1) +#define ML26124_RATES (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_48000) +#define ML26124_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) +#define ML26124_NUM_REGISTER ML26134_CACHESIZE + +struct ml26124_priv { + u32 mclk; + u32 rate; + struct regmap *regmap; + int clk_in; + struct snd_pcm_substream *substream; +}; + +struct clk_coeff { + u32 mclk; + u32 rate; + u8 pllnl; + u8 pllnh; + u8 pllml; + u8 pllmh; + u8 plldiv; +}; + +/* ML26124 configuration */ +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7150, 50, 0); + +static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(mingain, -1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(maxgain, -675, 600, 0); +static const DECLARE_TLV_DB_SCALE(boost_vol, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); + +static const char * const ml26124_companding[] = {"16bit PCM", "u-law", + "A-law"}; + +static const struct soc_enum ml26124_adc_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_TRANS_CTL, 6, 3, ml26124_companding); + +static const struct soc_enum ml26124_dac_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding); + +static const struct snd_kcontrol_new ml26124_snd_controls[] = { + SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Digital Boost Volume", ML26124_DIGI_BOOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE_TLV("EQ Band0 Volume", ML26124_EQ_GAIN_BRAND0, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band1 Volume", ML26124_EQ_GAIN_BRAND1, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band2 Volume", ML26124_EQ_GAIN_BRAND2, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band3 Volume", ML26124_EQ_GAIN_BRAND3, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band4 Volume", ML26124_EQ_GAIN_BRAND4, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("ALC Target Level", ML26124_ALC_TARGET_LEV, 0, + 0xf, 1, alclvl), + SOC_SINGLE_TLV("ALC Min Input Volume", ML26124_ALC_MAXMIN_GAIN, 0, + 7, 0, mingain), + SOC_SINGLE_TLV("ALC Max Input Volume", ML26124_ALC_MAXMIN_GAIN, 4, + 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Limiter Min Input Volume", + ML26124_PL_MAXMIN_GAIN, 0, 7, 0, mingain), + SOC_SINGLE_TLV("Playback Limiter Max Input Volume", + ML26124_PL_MAXMIN_GAIN, 4, 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Boost Volume", ML26124_PLYBAK_BOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE("DC High Pass Filter Switch", ML26124_FILTER_EN, 0, 1, 0), + SOC_SINGLE("Noise High Pass Filter Switch", ML26124_FILTER_EN, 1, 1, 0), + SOC_SINGLE("ZC Switch", ML26124_PW_ZCCMP_PW_MNG, 1, + 1, 0), + SOC_SINGLE("EQ Band0 Switch", ML26124_FILTER_EN, 2, 1, 0), + SOC_SINGLE("EQ Band1 Switch", ML26124_FILTER_EN, 3, 1, 0), + SOC_SINGLE("EQ Band2 Switch", ML26124_FILTER_EN, 4, 1, 0), + SOC_SINGLE("EQ Band3 Switch", ML26124_FILTER_EN, 5, 1, 0), + SOC_SINGLE("EQ Band4 Switch", ML26124_FILTER_EN, 6, 1, 0), + SOC_SINGLE("Play Limiter", ML26124_DVOL_CTL, 0, 1, 0), + SOC_SINGLE("Capture Limiter", ML26124_DVOL_CTL, 1, 1, 0), + SOC_SINGLE("Digital Volume Fade Switch", ML26124_DVOL_CTL, 3, 1, 0), + SOC_SINGLE("Digital Switch", ML26124_DVOL_CTL, 4, 1, 0), + SOC_ENUM("DAC Companding", ml26124_dac_companding_enum), + SOC_ENUM("ADC Companding", ml26124_adc_companding_enum), +}; + +static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", ML26124_SPK_AMP_OUT, 1, 1, 0), + SOC_DAPM_SINGLE("Line in loopback Switch", ML26124_SPK_AMP_OUT, 3, 1, + 0), + SOC_DAPM_SINGLE("PGA Switch", ML26124_SPK_AMP_OUT, 5, 1, 0), +}; + +/* Input mux */ +static const char * const ml26124_input_select[] = {"Analog MIC SingleEnded in", + "Digital MIC in", "Analog MIC Differential in"}; + +static const struct soc_enum ml26124_insel_enum = + SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 3, ml26124_input_select); + +static const struct snd_kcontrol_new ml26124_input_mux_controls = + SOC_DAPM_ENUM("Input Select", ml26124_insel_enum); + +static const struct snd_kcontrol_new ml26124_line_control = + SOC_DAPM_SINGLE("Switch", ML26124_PW_LOUT_PW_MNG, 1, 1, 0); + +static const struct snd_soc_dapm_widget ml26124_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("MCLKEN", ML26124_CLK_EN, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLLEN", ML26124_CLK_EN, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLLOE", ML26124_CLK_EN, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS", ML26124_PW_REF_PW_MNG, 2, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, + &ml26124_output_mixer_controls[0], + ARRAY_SIZE(ml26124_output_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "Playback", ML26124_PW_DAC_PW_MNG, 1, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", ML26124_PW_IN_PW_MNG, 1, 0), + SND_SOC_DAPM_PGA("PGA", ML26124_PW_IN_PW_MNG, 3, 0, NULL, 0), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ml26124_input_mux_controls), + SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, + &ml26124_line_control), + SND_SOC_DAPM_INPUT("MDIN"), + SND_SOC_DAPM_INPUT("MIN"), + SND_SOC_DAPM_INPUT("LIN"), + SND_SOC_DAPM_OUTPUT("SPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), +}; + +static const struct snd_soc_dapm_route ml26124_intercon[] = { + /* Supply */ + {"DAC", NULL, "MCLKEN"}, + {"ADC", NULL, "MCLKEN"}, + {"DAC", NULL, "PLLEN"}, + {"ADC", NULL, "PLLEN"}, + {"DAC", NULL, "PLLOE"}, + {"ADC", NULL, "PLLOE"}, + + /* output mixer */ + {"Output Mixer", "DAC Switch", "DAC"}, + {"Output Mixer", "Line in loopback Switch", "LIN"}, + + /* outputs */ + {"LOUT", NULL, "Output Mixer"}, + {"SPOUT", NULL, "Output Mixer"}, + {"Line Out Enable", NULL, "LOUT"}, + + /* input */ + {"ADC", NULL, "Input Mux"}, + {"Input Mux", "Analog MIC SingleEnded in", "PGA"}, + {"Input Mux", "Analog MIC Differential in", "PGA"}, + {"PGA", NULL, "MIN"}, +}; + +/* PLLOutputFreq(Hz) = InputMclkFreq(Hz) * PLLM / (PLLN * PLLDIV) */ +static const struct clk_coeff coeff_div[] = { + {12288000, 16000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288000, 32000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288000, 48000, 0xc, 0x0, 0x30, 0x0, 0x4}, +}; + +static struct reg_default ml26124_reg[] = { + /* CLOCK control Register */ + {0x00, 0x00 }, /* Sampling Rate */ + {0x02, 0x00}, /* PLL NL */ + {0x04, 0x00}, /* PLLNH */ + {0x06, 0x00}, /* PLLML */ + {0x08, 0x00}, /* MLLMH */ + {0x0a, 0x00}, /* PLLDIV */ + {0x0c, 0x00}, /* Clock Enable */ + {0x0e, 0x00}, /* CLK Input/Output Control */ + + /* System Control Register */ + {0x10, 0x00}, /* Software RESET */ + {0x12, 0x00}, /* Record/Playback Run */ + {0x14, 0x00}, /* Mic Input/Output control */ + + /* Power Management Register */ + {0x20, 0x00}, /* Reference Power Management */ + {0x22, 0x00}, /* Input Power Management */ + {0x24, 0x00}, /* DAC Power Management */ + {0x26, 0x00}, /* SP-AMP Power Management */ + {0x28, 0x00}, /* LINEOUT Power Management */ + {0x2a, 0x00}, /* VIDEO Power Management */ + {0x2e, 0x00}, /* AC-CMP Power Management */ + + /* Analog reference Control Register */ + {0x30, 0x04}, /* MICBIAS Voltage Control */ + + /* Input/Output Amplifier Control Register */ + {0x32, 0x10}, /* MIC Input Volume */ + {0x38, 0x00}, /* Mic Boost Volume */ + {0x3a, 0x33}, /* Speaker AMP Volume */ + {0x48, 0x00}, /* AMP Volume Control Function Enable */ + {0x4a, 0x00}, /* Amplifier Volume Fader Control */ + + /* Analog Path Control Register */ + {0x54, 0x00}, /* Speaker AMP Output Control */ + {0x5a, 0x00}, /* Mic IF Control */ + {0xe8, 0x01}, /* Mic Select Control */ + + /* Audio Interface Control Register */ + {0x60, 0x00}, /* SAI-Trans Control */ + {0x62, 0x00}, /* SAI-Receive Control */ + {0x64, 0x00}, /* SAI Mode select */ + + /* DSP Control Register */ + {0x66, 0x01}, /* Filter Func Enable */ + {0x68, 0x00}, /* Volume Control Func Enable */ + {0x6A, 0x00}, /* Mixer & Volume Control*/ + {0x6C, 0xff}, /* Record Digital Volume */ + {0x70, 0xff}, /* Playback Digital Volume */ + {0x72, 0x10}, /* Digital Boost Volume */ + {0x74, 0xe7}, /* EQ gain Band0 */ + {0x76, 0xe7}, /* EQ gain Band1 */ + {0x78, 0xe7}, /* EQ gain Band2 */ + {0x7A, 0xe7}, /* EQ gain Band3 */ + {0x7C, 0xe7}, /* EQ gain Band4 */ + {0x7E, 0x00}, /* HPF2 CutOff*/ + {0x80, 0x00}, /* EQ Band0 Coef0L */ + {0x82, 0x00}, /* EQ Band0 Coef0H */ + {0x84, 0x00}, /* EQ Band0 Coef0L */ + {0x86, 0x00}, /* EQ Band0 Coef0H */ + {0x88, 0x00}, /* EQ Band1 Coef0L */ + {0x8A, 0x00}, /* EQ Band1 Coef0H */ + {0x8C, 0x00}, /* EQ Band1 Coef0L */ + {0x8E, 0x00}, /* EQ Band1 Coef0H */ + {0x90, 0x00}, /* EQ Band2 Coef0L */ + {0x92, 0x00}, /* EQ Band2 Coef0H */ + {0x94, 0x00}, /* EQ Band2 Coef0L */ + {0x96, 0x00}, /* EQ Band2 Coef0H */ + {0x98, 0x00}, /* EQ Band3 Coef0L */ + {0x9A, 0x00}, /* EQ Band3 Coef0H */ + {0x9C, 0x00}, /* EQ Band3 Coef0L */ + {0x9E, 0x00}, /* EQ Band3 Coef0H */ + {0xA0, 0x00}, /* EQ Band4 Coef0L */ + {0xA2, 0x00}, /* EQ Band4 Coef0H */ + {0xA4, 0x00}, /* EQ Band4 Coef0L */ + {0xA6, 0x00}, /* EQ Band4 Coef0H */ + + /* ALC Control Register */ + {0xb0, 0x00}, /* ALC Mode */ + {0xb2, 0x02}, /* ALC Attack Time */ + {0xb4, 0x03}, /* ALC Decay Time */ + {0xb6, 0x00}, /* ALC Hold Time */ + {0xb8, 0x0b}, /* ALC Target Level */ + {0xba, 0x70}, /* ALC Max/Min Gain */ + {0xbc, 0x00}, /* Noise Gate Threshold */ + {0xbe, 0x00}, /* ALC ZeroCross TimeOut */ + + /* Playback Limiter Control Register */ + {0xc0, 0x04}, /* PL Attack Time */ + {0xc2, 0x05}, /* PL Decay Time */ + {0xc4, 0x0d}, /* PL Target Level */ + {0xc6, 0x70}, /* PL Max/Min Gain */ + {0xc8, 0x10}, /* Playback Boost Volume */ + {0xca, 0x00}, /* PL ZeroCross TimeOut */ + + /* Video Amplifier Control Register */ + {0xd0, 0x01}, /* VIDEO AMP Gain Control */ + {0xd2, 0x01}, /* VIDEO AMP Setup 1 */ + {0xd4, 0x01}, /* VIDEO AMP Control2 */ +}; + +/* Get sampling rate value of sampling rate setting register (0x0) */ +static inline int get_srate(int rate) +{ + int srate; + + switch (rate) { + case 16000: + srate = 3; + break; + case 32000: + srate = 6; + break; + case 48000: + srate = 8; + break; + default: + return -EINVAL; + } + return srate; +} + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return -EINVAL; +} + +static int ml26124_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + int i = get_coeff(priv->mclk, params_rate(hw_params)); + + priv->substream = substream; + priv->rate = params_rate(hw_params); + + if (priv->clk_in) { + switch (priv->mclk / params_rate(hw_params)) { + case 256: + snd_soc_update_bits(codec, ML26124_CLK_CTL, + BIT(0) | BIT(1), 1); + break; + case 512: + snd_soc_update_bits(codec, ML26124_CLK_CTL, + BIT(0) | BIT(1), 2); + break; + case 1024: + snd_soc_update_bits(codec, ML26124_CLK_CTL, + BIT(0) | BIT(1), 3); + break; + default: + dev_err(codec->dev, "Unsupported MCLKI\n"); + break; + } + } else { + snd_soc_update_bits(codec, ML26124_CLK_CTL, + BIT(0) | BIT(1), 0); + } + + switch (params_rate(hw_params)) { + case 16000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(params_rate(hw_params))); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 32000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(params_rate(hw_params))); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 48000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(params_rate(hw_params))); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + default: + pr_err("%s:this rate is no support for ml26124\n", __func__); + return -EINVAL; + } + + return 0; +} + +static int ml26124_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + switch (priv->substream->stream) { + case SNDRV_PCM_STREAM_CAPTURE: + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, BIT(0), 1); + break; + case SNDRV_PCM_STREAM_PLAYBACK: + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, BIT(1), 2); + break; + } + + if (mute) + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_ON); + else + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_OFF); + + return 0; +} + +static int ml26124_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + unsigned char mode; + struct snd_soc_codec *codec = codec_dai->codec; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + mode = 0; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, ML26124_SAI_MODE_SEL, BIT(0), mode); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ml26124_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case ML26124_USE_PLLOUT: + priv->clk_in = ML26124_USE_PLLOUT; + break; + case ML26124_USE_MCLKI: + priv->clk_in = ML26124_USE_MCLKI; + break; + default: + return -EINVAL; + } + + priv->mclk = freq; + + return 0; +} + +static int ml26124_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, ML26124_PW_SPAMP_PW_MNG, + ML26124_R26_MASK, ML26124_BLT_PREAMP_ON); + msleep(100); + snd_soc_update_bits(codec, ML26124_PW_SPAMP_PW_MNG, + ML26124_R26_MASK, + ML26124_MICBEN_ON | ML26124_BLT_ALL_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* VMID ON */ + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, ML26124_VMID); + msleep(500); + regcache_sync(priv->regmap); + } + break; + case SND_SOC_BIAS_OFF: + /* VMID OFF */ + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const struct snd_soc_dai_ops ml26124_dai_ops = { + .hw_params = ml26124_hw_params, + .digital_mute = ml26124_mute, + .set_fmt = ml26124_set_dai_fmt, + .set_sysclk = ml26124_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver ml26124_dai = { + .name = "ml26124-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .ops = &ml26124_dai_ops, + .symmetric_rates = 1, +}; + +#ifdef CONFIG_PM +static int ml26124_suspend(struct snd_soc_codec *codec) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int ml26124_resume(struct snd_soc_codec *codec) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define ml26124_suspend NULL +#define ml26124_resume NULL +#endif + +static int ml26124_probe(struct snd_soc_codec *codec) +{ + int ret; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + codec->control_data = priv->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Software Reset */ + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0); + + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_ml26124 = { + .probe = ml26124_probe, + .suspend = ml26124_suspend, + .resume = ml26124_resume, + .set_bias_level = ml26124_set_bias_level, + .dapm_widgets = ml26124_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets), + .dapm_routes = ml26124_intercon, + .num_dapm_routes = ARRAY_SIZE(ml26124_intercon), + .controls = ml26124_snd_controls, + .num_controls = ARRAY_SIZE(ml26124_snd_controls), +}; + +static const struct regmap_config ml26124_i2c_regmap = { + .val_bits = 8, + .reg_bits = 8, + .max_register = ML26124_NUM_REGISTER, + .reg_defaults = ml26124_reg, + .num_reg_defaults = ARRAY_SIZE(ml26124_reg), + .cache_type = REGCACHE_RBTREE, + .write_flag_mask = 0x01, +}; + +static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ml26124_priv *priv; + int ret; + + priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + i2c_set_clientdata(i2c, priv); + + priv->regmap = devm_regmap_init_i2c(i2c, &ml26124_i2c_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&i2c->dev, "regmap_init_i2c() failed: %d\n", ret); + return ret; + } + + return snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_ml26124, &ml26124_dai, 1); +} + +static __devexit int ml26124_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ml26124_i2c_id[] = { + { "ml26124", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id); + +static struct i2c_driver ml26124_i2c_driver = { + .driver = { + .name = "ml26124", + .owner = THIS_MODULE, + }, + .probe = ml26124_i2c_probe, + .remove = __devexit_p(ml26124_i2c_remove), + .id_table = ml26124_i2c_id, +}; + +module_i2c_driver(ml26124_i2c_driver); + +MODULE_AUTHOR("Tomoya MORINAGA <tomoya.rohm@gmail.com>"); +MODULE_DESCRIPTION("LAPIS Semiconductor ML26124 ALSA SoC codec driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ml26124.h b/sound/soc/codecs/ml26124.h new file mode 100644 index 000000000000..5ea0cbb8c46c --- /dev/null +++ b/sound/soc/codecs/ml26124.h @@ -0,0 +1,184 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#ifndef ML26124_H +#define ML26124_H + +/* Clock Control Register */ +#define ML26124_SMPLING_RATE 0x00 +#define ML26124_PLLNL 0x02 +#define ML26124_PLLNH 0x04 +#define ML26124_PLLML 0x06 +#define ML26124_PLLMH 0x08 +#define ML26124_PLLDIV 0x0a +#define ML26124_CLK_EN 0x0c +#define ML26124_CLK_CTL 0x0e + +/* System Control Register */ +#define ML26124_SW_RST 0x10 +#define ML26124_REC_PLYBAK_RUN 0x12 +#define ML26124_MIC_TIM 0x14 + +/* Power Mnagement Register */ +#define ML26124_PW_REF_PW_MNG 0x20 +#define ML26124_PW_IN_PW_MNG 0x22 +#define ML26124_PW_DAC_PW_MNG 0x24 +#define ML26124_PW_SPAMP_PW_MNG 0x26 +#define ML26124_PW_LOUT_PW_MNG 0x28 +#define ML26124_PW_VOUT_PW_MNG 0x2a +#define ML26124_PW_ZCCMP_PW_MNG 0x2e + +/* Analog Reference Control Register */ +#define ML26124_PW_MICBIAS_VOL 0x30 + +/* Input/Output Amplifier Control Register */ +#define ML26124_PW_MIC_IN_VOL 0x32 +#define ML26124_PW_MIC_BOST_VOL 0x38 +#define ML26124_PW_SPK_AMP_VOL 0x3a +#define ML26124_PW_AMP_VOL_FUNC 0x48 +#define ML26124_PW_AMP_VOL_FADE 0x4a + +/* Analog Path Control Register */ +#define ML26124_SPK_AMP_OUT 0x54 +#define ML26124_MIC_IF_CTL 0x5a +#define ML26124_MIC_SELECT 0xe8 + +/* Audio Interface Control Register */ +#define ML26124_SAI_TRANS_CTL 0x60 +#define ML26124_SAI_RCV_CTL 0x62 +#define ML26124_SAI_MODE_SEL 0x64 + +/* DSP Control Register */ +#define ML26124_FILTER_EN 0x66 +#define ML26124_DVOL_CTL 0x68 +#define ML26124_MIXER_VOL_CTL 0x6a +#define ML26124_RECORD_DIG_VOL 0x6c +#define ML26124_PLBAK_DIG_VOL 0x70 +#define ML26124_DIGI_BOOST_VOL 0x72 +#define ML26124_EQ_GAIN_BRAND0 0x74 +#define ML26124_EQ_GAIN_BRAND1 0x76 +#define ML26124_EQ_GAIN_BRAND2 0x78 +#define ML26124_EQ_GAIN_BRAND3 0x7a +#define ML26124_EQ_GAIN_BRAND4 0x7c +#define ML26124_HPF2_CUTOFF 0x7e +#define ML26124_EQBRAND0_F0L 0x80 +#define ML26124_EQBRAND0_F0H 0x82 +#define ML26124_EQBRAND0_F1L 0x84 +#define ML26124_EQBRAND0_F1H 0x86 +#define ML26124_EQBRAND1_F0L 0x88 +#define ML26124_EQBRAND1_F0H 0x8a +#define ML26124_EQBRAND1_F1L 0x8c +#define ML26124_EQBRAND1_F1H 0x8e +#define ML26124_EQBRAND2_F0L 0x90 +#define ML26124_EQBRAND2_F0H 0x92 +#define ML26124_EQBRAND2_F1L 0x94 +#define ML26124_EQBRAND2_F1H 0x96 +#define ML26124_EQBRAND3_F0L 0x98 +#define ML26124_EQBRAND3_F0H 0x9a +#define ML26124_EQBRAND3_F1L 0x9c +#define ML26124_EQBRAND3_F1H 0x9e +#define ML26124_EQBRAND4_F0L 0xa0 +#define ML26124_EQBRAND4_F0H 0xa2 +#define ML26124_EQBRAND4_F1L 0xa4 +#define ML26124_EQBRAND4_F1H 0xa6 + +/* ALC Control Register */ +#define ML26124_ALC_MODE 0xb0 +#define ML26124_ALC_ATTACK_TIM 0xb2 +#define ML26124_ALC_DECAY_TIM 0xb4 +#define ML26124_ALC_HOLD_TIM 0xb6 +#define ML26124_ALC_TARGET_LEV 0xb8 +#define ML26124_ALC_MAXMIN_GAIN 0xba +#define ML26124_NOIS_GATE_THRSH 0xbc +#define ML26124_ALC_ZERO_TIMOUT 0xbe + +/* Playback Limiter Control Register */ +#define ML26124_PL_ATTACKTIME 0xc0 +#define ML26124_PL_DECAYTIME 0xc2 +#define ML26124_PL_TARGETTIME 0xc4 +#define ML26124_PL_MAXMIN_GAIN 0xc6 +#define ML26124_PLYBAK_BOST_VOL 0xc8 +#define ML26124_PL_0CROSS_TIMOUT 0xca + +/* Video Amplifer Control Register */ +#define ML26124_VIDEO_AMP_GAIN_CTL 0xd0 +#define ML26124_VIDEO_AMP_SETUP1 0xd2 +#define ML26124_VIDEO_AMP_CTL2 0xd4 + +/* Clock select for machine driver */ +#define ML26124_USE_PLL 0 +#define ML26124_USE_MCLKI_256FS 1 +#define ML26124_USE_MCLKI_512FS 2 +#define ML26124_USE_MCLKI_1024FS 3 + +/* Register Mask */ +#define ML26124_R0_MASK 0xf +#define ML26124_R2_MASK 0xff +#define ML26124_R4_MASK 0x1 +#define ML26124_R6_MASK 0xf +#define ML26124_R8_MASK 0x3f +#define ML26124_Ra_MASK 0x1f +#define ML26124_Rc_MASK 0x1f +#define ML26124_Re_MASK 0x7 +#define ML26124_R10_MASK 0x1 +#define ML26124_R12_MASK 0x17 +#define ML26124_R14_MASK 0x3f +#define ML26124_R20_MASK 0x47 +#define ML26124_R22_MASK 0xa +#define ML26124_R24_MASK 0x2 +#define ML26124_R26_MASK 0x1f +#define ML26124_R28_MASK 0x2 +#define ML26124_R2a_MASK 0x2 +#define ML26124_R2e_MASK 0x2 +#define ML26124_R30_MASK 0x7 +#define ML26124_R32_MASK 0x3f +#define ML26124_R38_MASK 0x38 +#define ML26124_R3a_MASK 0x3f +#define ML26124_R48_MASK 0x3 +#define ML26124_R4a_MASK 0x7 +#define ML26124_R54_MASK 0x2a +#define ML26124_R5a_MASK 0x3 +#define ML26124_Re8_MASK 0x3 +#define ML26124_R60_MASK 0xff +#define ML26124_R62_MASK 0xff +#define ML26124_R64_MASK 0x1 +#define ML26124_R66_MASK 0xff +#define ML26124_R68_MASK 0x3b +#define ML26124_R6a_MASK 0xf3 +#define ML26124_R6c_MASK 0xff +#define ML26124_R70_MASK 0xff + +#define ML26124_MCLKEN BIT(0) +#define ML26124_PLLEN BIT(1) +#define ML26124_PLLOE BIT(2) +#define ML26124_MCLKOE BIT(3) + +#define ML26124_BLT_ALL_ON 0x1f +#define ML26124_BLT_PREAMP_ON 0x13 + +#define ML26124_MICBEN_ON BIT(2) + +enum ml26124_regs { + ML26124_MCLK = 0, +}; + +enum ml26124_clk_in { + ML26124_USE_PLLOUT = 0, + ML26124_USE_MCLKI, +}; + +#endif diff --git a/sound/soc/codecs/omap-hdmi.c b/sound/soc/codecs/omap-hdmi.c new file mode 100644 index 000000000000..1bf5c74f5f96 --- /dev/null +++ b/sound/soc/codecs/omap-hdmi.c @@ -0,0 +1,69 @@ +/* + * ALSA SoC codec driver for HDMI audio on OMAP processors. + * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/ + * Author: Ricardo Neri <ricardo.neri@ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ +#include <linux/module.h> +#include <sound/soc.h> + +#define DRV_NAME "hdmi-audio-codec" + +static struct snd_soc_codec_driver omap_hdmi_codec; + +static struct snd_soc_dai_driver omap_hdmi_codec_dai = { + .name = "omap-hdmi-hifi", + .playback = { + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + }, +}; + +static __devinit int omap_hdmi_codec_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &omap_hdmi_codec, + &omap_hdmi_codec_dai, 1); +} + +static __devexit int omap_hdmi_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver omap_hdmi_codec_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + + .probe = omap_hdmi_codec_probe, + .remove = __devexit_p(omap_hdmi_codec_remove), +}; + +module_platform_driver(omap_hdmi_codec_driver); + +MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>"); +MODULE_DESCRIPTION("ASoC OMAP HDMI codec driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 20c324c7c349..960d0e93cce9 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -18,7 +18,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> -#include <linux/spi/spi.h> +#include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -30,6 +30,7 @@ #include "rt5631.h" struct rt5631_priv { + struct regmap *regmap; int codec_version; int master; int sysclk; @@ -38,33 +39,33 @@ struct rt5631_priv { int dmic_used_flag; }; -static const u16 rt5631_reg[RT5631_VENDOR_ID2 + 1] = { - [RT5631_SPK_OUT_VOL] = 0x8888, - [RT5631_HP_OUT_VOL] = 0x8080, - [RT5631_MONO_AXO_1_2_VOL] = 0xa080, - [RT5631_AUX_IN_VOL] = 0x0808, - [RT5631_ADC_REC_MIXER] = 0xf0f0, - [RT5631_VDAC_DIG_VOL] = 0x0010, - [RT5631_OUTMIXER_L_CTRL] = 0xffc0, - [RT5631_OUTMIXER_R_CTRL] = 0xffc0, - [RT5631_AXO1MIXER_CTRL] = 0x88c0, - [RT5631_AXO2MIXER_CTRL] = 0x88c0, - [RT5631_DIG_MIC_CTRL] = 0x3000, - [RT5631_MONO_INPUT_VOL] = 0x8808, - [RT5631_SPK_MIXER_CTRL] = 0xf8f8, - [RT5631_SPK_MONO_OUT_CTRL] = 0xfc00, - [RT5631_SPK_MONO_HP_OUT_CTRL] = 0x4440, - [RT5631_SDP_CTRL] = 0x8000, - [RT5631_MONO_SDP_CTRL] = 0x8000, - [RT5631_STEREO_AD_DA_CLK_CTRL] = 0x2010, - [RT5631_GEN_PUR_CTRL_REG] = 0x0e00, - [RT5631_INT_ST_IRQ_CTRL_2] = 0x071a, - [RT5631_MISC_CTRL] = 0x2040, - [RT5631_DEPOP_FUN_CTRL_2] = 0x8000, - [RT5631_SOFT_VOL_CTRL] = 0x07e0, - [RT5631_ALC_CTRL_1] = 0x0206, - [RT5631_ALC_CTRL_3] = 0x2000, - [RT5631_PSEUDO_SPATL_CTRL] = 0x0553, +static const struct reg_default rt5631_reg[] = { + { RT5631_SPK_OUT_VOL, 0x8888 }, + { RT5631_HP_OUT_VOL, 0x8080 }, + { RT5631_MONO_AXO_1_2_VOL, 0xa080 }, + { RT5631_AUX_IN_VOL, 0x0808 }, + { RT5631_ADC_REC_MIXER, 0xf0f0 }, + { RT5631_VDAC_DIG_VOL, 0x0010 }, + { RT5631_OUTMIXER_L_CTRL, 0xffc0 }, + { RT5631_OUTMIXER_R_CTRL, 0xffc0 }, + { RT5631_AXO1MIXER_CTRL, 0x88c0 }, + { RT5631_AXO2MIXER_CTRL, 0x88c0 }, + { RT5631_DIG_MIC_CTRL, 0x3000 }, + { RT5631_MONO_INPUT_VOL, 0x8808 }, + { RT5631_SPK_MIXER_CTRL, 0xf8f8 }, + { RT5631_SPK_MONO_OUT_CTRL, 0xfc00 }, + { RT5631_SPK_MONO_HP_OUT_CTRL, 0x4440 }, + { RT5631_SDP_CTRL, 0x8000 }, + { RT5631_MONO_SDP_CTRL, 0x8000 }, + { RT5631_STEREO_AD_DA_CLK_CTRL, 0x2010 }, + { RT5631_GEN_PUR_CTRL_REG, 0x0e00 }, + { RT5631_INT_ST_IRQ_CTRL_2, 0x071a }, + { RT5631_MISC_CTRL, 0x2040 }, + { RT5631_DEPOP_FUN_CTRL_2, 0x8000 }, + { RT5631_SOFT_VOL_CTRL, 0x07e0 }, + { RT5631_ALC_CTRL_1, 0x0206 }, + { RT5631_ALC_CTRL_3, 0x2000 }, + { RT5631_PSEUDO_SPATL_CTRL, 0x0553 }, }; /** @@ -96,8 +97,7 @@ static int rt5631_reset(struct snd_soc_codec *codec) return snd_soc_write(codec, RT5631_RESET, 0); } -static int rt5631_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool rt5631_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case RT5631_RESET: @@ -111,8 +111,7 @@ static int rt5631_volatile_register(struct snd_soc_codec *codec, } } -static int rt5631_readable_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool rt5631_readable_register(struct device *dev, unsigned int reg) { switch (reg) { case RT5631_RESET: @@ -1361,8 +1360,7 @@ static int get_coeff(int mclk, int rate, int timesofbclk) static int rt5631_hifi_pcm_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); int timesofbclk = 32, coeff; unsigned int iface = 0; @@ -1544,6 +1542,8 @@ static int rt5631_codec_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, static int rt5631_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: @@ -1561,8 +1561,8 @@ static int rt5631_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3, RT5631_PWR_FAST_VREF_CTRL, RT5631_PWR_FAST_VREF_CTRL); - codec->cache_only = false; - snd_soc_cache_sync(codec); + regcache_cache_only(rt5631->regmap, false); + regcache_sync(rt5631->regmap); } break; @@ -1587,7 +1587,9 @@ static int rt5631_probe(struct snd_soc_codec *codec) unsigned int val; int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + codec->control_data = rt5631->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -1698,12 +1700,6 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5631 = { .suspend = rt5631_suspend, .resume = rt5631_resume, .set_bias_level = rt5631_set_bias_level, - .reg_cache_size = RT5631_VENDOR_ID2 + 1, - .reg_word_size = sizeof(u16), - .reg_cache_default = rt5631_reg, - .volatile_register = rt5631_volatile_register, - .readable_register = rt5631_readable_register, - .reg_cache_step = 1, .controls = rt5631_snd_controls, .num_controls = ARRAY_SIZE(rt5631_snd_controls), .dapm_widgets = rt5631_dapm_widgets, @@ -1718,6 +1714,18 @@ static const struct i2c_device_id rt5631_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id); +static const struct regmap_config rt5631_regmap_config = { + .reg_bits = 8, + .val_bits = 16, + + .readable_reg = rt5631_readable_register, + .volatile_reg = rt5631_volatile_register, + .max_register = RT5631_VENDOR_ID2, + .reg_defaults = rt5631_reg, + .num_reg_defaults = ARRAY_SIZE(rt5631_reg), + .cache_type = REGCACHE_RBTREE, +}; + static int rt5631_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1731,6 +1739,10 @@ static int rt5631_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5631); + rt5631->regmap = devm_regmap_init_i2c(i2c, &rt5631_regmap_config); + if (IS_ERR(rt5631->regmap)) + return PTR_ERR(rt5631->regmap); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5631, rt5631_dai, ARRAY_SIZE(rt5631_dai)); return ret; @@ -1752,17 +1764,7 @@ static struct i2c_driver rt5631_i2c_driver = { .id_table = rt5631_i2c_id, }; -static int __init rt5631_modinit(void) -{ - return i2c_add_driver(&rt5631_i2c_driver); -} -module_init(rt5631_modinit); - -static void __exit rt5631_modexit(void) -{ - i2c_del_driver(&rt5631_i2c_driver); -} -module_exit(rt5631_modexit); +module_i2c_driver(rt5631_i2c_driver); MODULE_DESCRIPTION("ASoC RT5631 driver"); MODULE_AUTHOR("flove <flove@realtek.com>"); diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 8e92fb88ed09..8af6a5245b18 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -84,8 +84,8 @@ static struct regulator_consumer_supply ldo_consumer[] = { static struct regulator_init_data ldo_init_data = { .constraints = { - .min_uV = 850000, - .max_uV = 1600000, + .min_uV = 1200000, + .max_uV = 1200000, .valid_modes_mask = REGULATOR_MODE_NORMAL, .valid_ops_mask = REGULATOR_CHANGE_STATUS, }, @@ -197,9 +197,9 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HP_OUT"), SND_SOC_DAPM_OUTPUT("LINE_OUT"), - SND_SOC_DAPM_MICBIAS_E("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0, - mic_bias_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0, + mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0), SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0), @@ -665,8 +665,7 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); int channels = params_channels(params); int i2s_ctl = 0; @@ -809,6 +808,7 @@ static int ldo_regulator_register(struct snd_soc_codec *codec, { struct ldo_regulator *ldo; struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + struct regulator_config config = { }; ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL); @@ -832,8 +832,11 @@ static int ldo_regulator_register(struct snd_soc_codec *codec, ldo->codec_data = codec; ldo->voltage = voltage; - ldo->dev = regulator_register(&ldo->desc, codec->dev, - init_data, ldo, NULL); + config.dev = codec->dev; + config.driver_data = ldo; + config.init_data = init_data; + + ldo->dev = regulator_register(&ldo->desc, &config); if (IS_ERR(ldo->dev)) { int ret = PTR_ERR(ldo->dev); @@ -1451,17 +1454,7 @@ static struct i2c_driver sgtl5000_i2c_driver = { .id_table = sgtl5000_id, }; -static int __init sgtl5000_modinit(void) -{ - return i2c_add_driver(&sgtl5000_i2c_driver); -} -module_init(sgtl5000_modinit); - -static void __exit sgtl5000_exit(void) -{ - i2c_del_driver(&sgtl5000_i2c_driver); -} -module_exit(sgtl5000_exit); +module_i2c_driver(sgtl5000_i2c_driver); MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver"); MODULE_AUTHOR("Zeng Zhaoming <zengzm.kernel@gmail.com>"); diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c new file mode 100644 index 000000000000..dd8d856053fc --- /dev/null +++ b/sound/soc/codecs/spdif_receiver.c @@ -0,0 +1,67 @@ +/* + * ALSA SoC SPDIF DIR (Digital Interface Reciever) driver + * + * Based on ALSA SoC SPDIF DIT driver + * + * This driver is used by controllers which can operate in DIR (SPDI/F) where + * no codec is needed. This file provides stub codec that can be used + * in these configurations. SPEAr SPDIF IN Audio controller uses this driver. + * + * Author: Vipin Kumar, <vipin.kumar@st.com> + * Copyright: (C) 2012 ST Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/slab.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/initval.h> + +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) + +static struct snd_soc_codec_driver soc_codec_spdif_dir; + +static struct snd_soc_dai_driver dir_stub_dai = { + .name = "dir-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, +}; + +static int spdif_dir_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &soc_codec_spdif_dir, + &dir_stub_dai, 1); +} + +static int spdif_dir_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver spdif_dir_driver = { + .probe = spdif_dir_probe, + .remove = spdif_dir_remove, + .driver = { + .name = "spdif-dir", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(spdif_dir_driver); + +MODULE_DESCRIPTION("ASoC SPDIF DIR driver"); +MODULE_AUTHOR("Vipin Kumar <vipin.kumar@st.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index de2b20544ceb..079066fef425 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -33,6 +33,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/spi/spi.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -43,8 +44,6 @@ #include "ssm2602.h" -#define SSM2602_VERSION "0.1" - enum ssm2602_type { SSM2602, SSM2604, @@ -53,10 +52,12 @@ enum ssm2602_type { /* codec private data */ struct ssm2602_priv { unsigned int sysclk; - enum snd_soc_control_type control_type; + struct snd_pcm_hw_constraint_list *sysclk_constraints; struct snd_pcm_substream *master_substream; struct snd_pcm_substream *slave_substream; + struct regmap *regmap; + enum ssm2602_type type; unsigned int clk_out_pwr; }; @@ -73,7 +74,6 @@ static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = { 0x0000, 0x0000 }; -#define ssm2602_reset(c) snd_soc_write(c, SSM2602_RESET, 0) /*Appending several "None"s just for OSS mixer use*/ static const char *ssm2602_input_select[] = { @@ -195,6 +195,24 @@ static const struct snd_soc_dapm_route ssm2604_routes[] = { {"ADC", NULL, "Line Input"}, }; +static const unsigned int ssm2602_rates_12288000[] = { + 8000, 32000, 48000, 96000, +}; + +static struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = { + .list = ssm2602_rates_12288000, + .count = ARRAY_SIZE(ssm2602_rates_12288000), +}; + +static const unsigned int ssm2602_rates_11289600[] = { + 8000, 44100, 88200, +}; + +static struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = { + .list = ssm2602_rates_11289600, + .count = ARRAY_SIZE(ssm2602_rates_11289600), +}; + struct ssm2602_coeff { u32 mclk; u32 rate; @@ -254,11 +272,10 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - u16 iface = snd_soc_read(codec, SSM2602_IFACE) & 0xfff3; int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params)); + unsigned int iface; if (substream == ssm2602->slave_substream) { dev_dbg(codec->dev, "Ignoring hw_params for slave substream\n"); @@ -268,31 +285,34 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, if (srate < 0) return srate; - snd_soc_write(codec, SSM2602_SRATE, srate); + regmap_write(ssm2602->regmap, SSM2602_SRATE, srate); /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: + iface = 0x0; break; case SNDRV_PCM_FORMAT_S20_3LE: - iface |= 0x0004; + iface = 0x4; break; case SNDRV_PCM_FORMAT_S24_LE: - iface |= 0x0008; + iface = 0x8; break; case SNDRV_PCM_FORMAT_S32_LE: - iface |= 0x000c; + iface = 0xc; break; + default: + return -EINVAL; } - snd_soc_write(codec, SSM2602_IFACE, iface); + regmap_update_bits(ssm2602->regmap, SSM2602_IFACE, + IFACE_AUDIO_DATA_LEN, iface); return 0; } static int ssm2602_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); struct snd_pcm_runtime *master_runtime; @@ -322,14 +342,19 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, } else ssm2602->master_substream = substream; + if (ssm2602->sysclk_constraints) { + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + ssm2602->sysclk_constraints); + } + return 0; } static void ssm2602_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); if (ssm2602->master_substream == substream) @@ -341,14 +366,14 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream, static int ssm2602_mute(struct snd_soc_dai *dai, int mute) { - struct snd_soc_codec *codec = dai->codec; + struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(dai->codec); if (mute) - snd_soc_update_bits(codec, SSM2602_APDIGI, + regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI, APDIGI_ENABLE_DAC_MUTE, APDIGI_ENABLE_DAC_MUTE); else - snd_soc_update_bits(codec, SSM2602_APDIGI, + regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI, APDIGI_ENABLE_DAC_MUTE, 0); return 0; } @@ -364,16 +389,21 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai, return -EINVAL; switch (freq) { - case 11289600: - case 12000000: case 12288000: - case 16934400: case 18432000: - ssm2602->sysclk = freq; + ssm2602->sysclk_constraints = &ssm2602_constraints_12288000; + break; + case 11289600: + case 16934400: + ssm2602->sysclk_constraints = &ssm2602_constraints_11289600; + break; + case 12000000: + ssm2602->sysclk_constraints = NULL; break; default: return -EINVAL; } + ssm2602->sysclk = freq; } else { unsigned int mask; @@ -393,7 +423,7 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai, else ssm2602->clk_out_pwr &= ~mask; - snd_soc_update_bits(codec, SSM2602_PWR, + regmap_update_bits(ssm2602->regmap, SSM2602_PWR, PWR_CLK_OUT_PDN | PWR_OSC_PDN, ssm2602->clk_out_pwr); } @@ -403,8 +433,8 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; - u16 iface = 0; + struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec_dai->codec); + unsigned int iface = 0; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -455,7 +485,7 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, } /* set iface */ - snd_soc_write(codec, SSM2602_IFACE, iface); + regmap_write(ssm2602->regmap, SSM2602_IFACE, iface); return 0; } @@ -467,7 +497,7 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: /* vref/mid on, osc and clkout on if enabled */ - snd_soc_update_bits(codec, SSM2602_PWR, + regmap_update_bits(ssm2602->regmap, SSM2602_PWR, PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN, ssm2602->clk_out_pwr); break; @@ -475,13 +505,13 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ - snd_soc_update_bits(codec, SSM2602_PWR, + regmap_update_bits(ssm2602->regmap, SSM2602_PWR, PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN, PWR_CLK_OUT_PDN | PWR_OSC_PDN); break; case SND_SOC_BIAS_OFF: /* everything off */ - snd_soc_update_bits(codec, SSM2602_PWR, + regmap_update_bits(ssm2602->regmap, SSM2602_PWR, PWR_POWER_OFF, PWR_POWER_OFF); break; @@ -540,12 +570,13 @@ static int ssm2602_resume(struct snd_soc_codec *codec) static int ssm2602_probe(struct snd_soc_codec *codec) { + struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - snd_soc_update_bits(codec, SSM2602_LOUT1V, + regmap_update_bits(ssm2602->regmap, SSM2602_LOUT1V, LOUT1V_LRHP_BOTH, LOUT1V_LRHP_BOTH); - snd_soc_update_bits(codec, SSM2602_ROUT1V, + regmap_update_bits(ssm2602->regmap, SSM2602_ROUT1V, ROUT1V_RLHP_BOTH, ROUT1V_RLHP_BOTH); ret = snd_soc_add_codec_controls(codec, ssm2602_snd_controls, @@ -581,27 +612,26 @@ static int ssm260x_probe(struct snd_soc_codec *codec) struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); int ret; - pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION); - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, ssm2602->control_type); + codec->control_data = ssm2602->regmap; + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } - ret = ssm2602_reset(codec); + ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); return ret; } /* set the update bits */ - snd_soc_update_bits(codec, SSM2602_LINVOL, + regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL, LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH); - snd_soc_update_bits(codec, SSM2602_RINVOL, + regmap_update_bits(ssm2602->regmap, SSM2602_RINVOL, RINVOL_RLIN_BOTH, RINVOL_RLIN_BOTH); /*select Line in as default input*/ - snd_soc_write(codec, SSM2602_APANA, APANA_SELECT_DAC | + regmap_write(ssm2602->regmap, SSM2602_APANA, APANA_SELECT_DAC | APANA_ENABLE_MIC_BOOST); switch (ssm2602->type) { @@ -634,9 +664,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { .suspend = ssm2602_suspend, .resume = ssm2602_resume, .set_bias_level = ssm2602_set_bias_level, - .reg_cache_size = ARRAY_SIZE(ssm2602_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = ssm2602_reg, .controls = ssm260x_snd_controls, .num_controls = ARRAY_SIZE(ssm260x_snd_controls), @@ -646,6 +673,23 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { .num_dapm_routes = ARRAY_SIZE(ssm260x_routes), }; +static bool ssm2602_register_volatile(struct device *dev, unsigned int reg) +{ + return reg == SSM2602_RESET; +} + +static const struct regmap_config ssm2602_regmap_config = { + .val_bits = 9, + .reg_bits = 7, + + .max_register = SSM2602_RESET, + .volatile_reg = ssm2602_register_volatile, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults_raw = ssm2602_reg, + .num_reg_defaults_raw = ARRAY_SIZE(ssm2602_reg), +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit ssm2602_spi_probe(struct spi_device *spi) { @@ -658,9 +702,12 @@ static int __devinit ssm2602_spi_probe(struct spi_device *spi) return -ENOMEM; spi_set_drvdata(spi, ssm2602); - ssm2602->control_type = SND_SOC_SPI; ssm2602->type = SSM2602; + ssm2602->regmap = devm_regmap_init_spi(spi, &ssm2602_regmap_config); + if (IS_ERR(ssm2602->regmap)) + return PTR_ERR(ssm2602->regmap); + ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_ssm2602, &ssm2602_dai, 1); return ret; @@ -701,9 +748,12 @@ static int __devinit ssm2602_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, ssm2602); - ssm2602->control_type = SND_SOC_I2C; ssm2602->type = id->driver_data; + ssm2602->regmap = devm_regmap_init_i2c(i2c, &ssm2602_regmap_config); + if (IS_ERR(ssm2602->regmap)) + return PTR_ERR(ssm2602->regmap); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ssm2602, &ssm2602_dai, 1); return ret; diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 7db6fa515028..8d717f4b5a87 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -609,8 +609,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); unsigned int rate; int i, mcs = -1, ir = -1; diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c new file mode 100644 index 000000000000..0c225cd569d2 --- /dev/null +++ b/sound/soc/codecs/sta529.c @@ -0,0 +1,442 @@ +/* + * ASoC codec driver for spear platform + * + * sound/soc/codecs/sta529.c -- spear ALSA Soc codec driver + * + * Copyright (C) 2012 ST Microelectronics + * Rajeev Kumar <rajeev-dlh.kumar@st.com> + * + * This file is licensed under the terms of the GNU General Public + * License version 2. This program is licensed "as is" without any + * warranty of any kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/pm.h> +#include <linux/regmap.h> +#include <linux/slab.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +/* STA529 Register offsets */ +#define STA529_FFXCFG0 0x00 +#define STA529_FFXCFG1 0x01 +#define STA529_MVOL 0x02 +#define STA529_LVOL 0x03 +#define STA529_RVOL 0x04 +#define STA529_TTF0 0x05 +#define STA529_TTF1 0x06 +#define STA529_TTP0 0x07 +#define STA529_TTP1 0x08 +#define STA529_S2PCFG0 0x0A +#define STA529_S2PCFG1 0x0B +#define STA529_P2SCFG0 0x0C +#define STA529_P2SCFG1 0x0D +#define STA529_PLLCFG0 0x14 +#define STA529_PLLCFG1 0x15 +#define STA529_PLLCFG2 0x16 +#define STA529_PLLCFG3 0x17 +#define STA529_PLLPFE 0x18 +#define STA529_PLLST 0x19 +#define STA529_ADCCFG 0x1E /*mic_select*/ +#define STA529_CKOCFG 0x1F +#define STA529_MISC 0x20 +#define STA529_PADST0 0x21 +#define STA529_PADST1 0x22 +#define STA529_FFXST 0x23 +#define STA529_PWMIN1 0x2D +#define STA529_PWMIN2 0x2E +#define STA529_POWST 0x32 + +#define STA529_MAX_REGISTER 0x32 + +#define STA529_RATES (SNDRV_PCM_RATE_8000 | \ + SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +#define STA529_FORMAT (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) +#define S2PC_VALUE 0x98 +#define CLOCK_OUT 0x60 +#define LEFT_J_DATA_FORMAT 0x10 +#define I2S_DATA_FORMAT 0x12 +#define RIGHT_J_DATA_FORMAT 0x14 +#define CODEC_MUTE_VAL 0x80 + +#define POWER_CNTLMSAK 0x40 +#define POWER_STDBY 0x40 +#define FFX_MASK 0x80 +#define FFX_OFF 0x80 +#define POWER_UP 0x00 +#define FFX_CLK_ENB 0x01 +#define FFX_CLK_DIS 0x00 +#define FFX_CLK_MSK 0x01 +#define PLAY_FREQ_RANGE_MSK 0x70 +#define CAP_FREQ_RANGE_MSK 0x0C +#define PDATA_LEN_MSK 0xC0 +#define BCLK_TO_FS_MSK 0x30 +#define AUDIO_MUTE_MSK 0x80 + +static const struct reg_default sta529_reg_defaults[] = { + { 0, 0x35 }, /* R0 - FFX Configuration reg 0 */ + { 1, 0xc8 }, /* R1 - FFX Configuration reg 1 */ + { 2, 0x50 }, /* R2 - Master Volume */ + { 3, 0x00 }, /* R3 - Left Volume */ + { 4, 0x00 }, /* R4 - Right Volume */ + { 10, 0xb2 }, /* R10 - S2P Config Reg 0 */ + { 11, 0x41 }, /* R11 - S2P Config Reg 1 */ + { 12, 0x92 }, /* R12 - P2S Config Reg 0 */ + { 13, 0x41 }, /* R13 - P2S Config Reg 1 */ + { 30, 0xd2 }, /* R30 - ADC Config Reg */ + { 31, 0x40 }, /* R31 - clock Out Reg */ + { 32, 0x21 }, /* R32 - Misc Register */ +}; + +struct sta529 { + struct regmap *regmap; +}; + +static bool sta529_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + + case STA529_FFXCFG0: + case STA529_FFXCFG1: + case STA529_MVOL: + case STA529_LVOL: + case STA529_RVOL: + case STA529_S2PCFG0: + case STA529_S2PCFG1: + case STA529_P2SCFG0: + case STA529_P2SCFG1: + case STA529_ADCCFG: + case STA529_CKOCFG: + case STA529_MISC: + return true; + default: + return false; + } +} + + +static const char *pwm_mode_text[] = { "Binary", "Headphone", "Ternary", + "Phase-shift"}; + +static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -9150, 50, 0); +static const DECLARE_TLV_DB_SCALE(master_vol_tlv, -12750, 50, 0); +static const SOC_ENUM_SINGLE_DECL(pwm_src, STA529_FFXCFG1, 4, pwm_mode_text); + +static const struct snd_kcontrol_new sta529_snd_controls[] = { + SOC_DOUBLE_R_TLV("Digital Playback Volume", STA529_LVOL, STA529_RVOL, 0, + 127, 0, out_gain_tlv), + SOC_SINGLE_TLV("Master Playback Volume", STA529_MVOL, 0, 127, 1, + master_vol_tlv), + SOC_ENUM("PWM Select", pwm_src), +}; + +static int sta529_set_bias_level(struct snd_soc_codec *codec, enum + snd_soc_bias_level level) +{ + struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, STA529_FFXCFG0, POWER_CNTLMSAK, + POWER_UP); + snd_soc_update_bits(codec, STA529_MISC, FFX_CLK_MSK, + FFX_CLK_ENB); + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + regcache_sync(sta529->regmap); + snd_soc_update_bits(codec, STA529_FFXCFG0, + POWER_CNTLMSAK, POWER_STDBY); + /* Making FFX output to zero */ + snd_soc_update_bits(codec, STA529_FFXCFG0, FFX_MASK, + FFX_OFF); + snd_soc_update_bits(codec, STA529_MISC, FFX_CLK_MSK, + FFX_CLK_DIS); + break; + case SND_SOC_BIAS_OFF: + break; + } + + /* + * store the label for powers down audio subsystem for suspend.This is + * used by soc core layer + */ + codec->dapm.bias_level = level; + + return 0; + +} + +static int sta529_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + int pdata, play_freq_val, record_freq_val; + int bclk_to_fs_ratio; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + pdata = 1; + bclk_to_fs_ratio = 0; + break; + case SNDRV_PCM_FORMAT_S24_LE: + pdata = 2; + bclk_to_fs_ratio = 1; + break; + case SNDRV_PCM_FORMAT_S32_LE: + pdata = 3; + bclk_to_fs_ratio = 2; + break; + default: + dev_err(codec->dev, "Unsupported format\n"); + return -EINVAL; + } + + switch (params_rate(params)) { + case 8000: + case 11025: + play_freq_val = 0; + record_freq_val = 2; + break; + case 16000: + case 22050: + play_freq_val = 1; + record_freq_val = 0; + break; + + case 32000: + case 44100: + case 48000: + play_freq_val = 2; + record_freq_val = 0; + break; + default: + dev_err(codec->dev, "Unsupported rate\n"); + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + snd_soc_update_bits(codec, STA529_S2PCFG1, PDATA_LEN_MSK, + pdata << 6); + snd_soc_update_bits(codec, STA529_S2PCFG1, BCLK_TO_FS_MSK, + bclk_to_fs_ratio << 4); + snd_soc_update_bits(codec, STA529_MISC, PLAY_FREQ_RANGE_MSK, + play_freq_val << 4); + } else { + snd_soc_update_bits(codec, STA529_P2SCFG1, PDATA_LEN_MSK, + pdata << 6); + snd_soc_update_bits(codec, STA529_P2SCFG1, BCLK_TO_FS_MSK, + bclk_to_fs_ratio << 4); + snd_soc_update_bits(codec, STA529_MISC, CAP_FREQ_RANGE_MSK, + record_freq_val << 2); + } + + return 0; +} + +static int sta529_mute(struct snd_soc_dai *dai, int mute) +{ + u8 val = 0; + + if (mute) + val |= CODEC_MUTE_VAL; + + snd_soc_update_bits(dai->codec, STA529_FFXCFG0, AUDIO_MUTE_MSK, val); + + return 0; +} + +static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 mode = 0; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + mode = LEFT_J_DATA_FORMAT; + break; + case SND_SOC_DAIFMT_I2S: + mode = I2S_DATA_FORMAT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + mode = RIGHT_J_DATA_FORMAT; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode); + + return 0; +} + +static const struct snd_soc_dai_ops sta529_dai_ops = { + .hw_params = sta529_hw_params, + .set_fmt = sta529_set_dai_fmt, + .digital_mute = sta529_mute, +}; + +static struct snd_soc_dai_driver sta529_dai = { + .name = "sta529-audio", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = STA529_RATES, + .formats = STA529_FORMAT, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = STA529_RATES, + .formats = STA529_FORMAT, + }, + .ops = &sta529_dai_ops, +}; + +static int sta529_probe(struct snd_soc_codec *codec) +{ + struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = sta529->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +/* power down chip */ +static int sta529_remove(struct snd_soc_codec *codec) +{ + sta529_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int sta529_suspend(struct snd_soc_codec *codec) +{ + sta529_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int sta529_resume(struct snd_soc_codec *codec) +{ + sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +struct snd_soc_codec_driver sta529_codec_driver = { + .probe = sta529_probe, + .remove = sta529_remove, + .set_bias_level = sta529_set_bias_level, + .suspend = sta529_suspend, + .resume = sta529_resume, + .controls = sta529_snd_controls, + .num_controls = ARRAY_SIZE(sta529_snd_controls), +}; + +static const struct regmap_config sta529_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = STA529_MAX_REGISTER, + .readable_reg = sta529_readable, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = sta529_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(sta529_reg_defaults), +}; + +static __devinit int sta529_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct sta529 *sta529; + int ret; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) + return -EINVAL; + + sta529 = devm_kzalloc(&i2c->dev, sizeof(struct sta529), GFP_KERNEL); + if (sta529 == NULL) { + dev_err(&i2c->dev, "Can not allocate memory\n"); + return -ENOMEM; + } + + sta529->regmap = devm_regmap_init_i2c(i2c, &sta529_regmap); + if (IS_ERR(sta529->regmap)) { + ret = PTR_ERR(sta529->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(i2c, sta529); + + ret = snd_soc_register_codec(&i2c->dev, + &sta529_codec_driver, &sta529_dai, 1); + if (ret != 0) + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + + return ret; +} + +static int __devexit sta529_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id sta529_i2c_id[] = { + { "sta529", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, sta529_i2c_id); + +static struct i2c_driver sta529_i2c_driver = { + .driver = { + .name = "sta529", + .owner = THIS_MODULE, + }, + .probe = sta529_i2c_probe, + .remove = __devexit_p(sta529_i2c_remove), + .id_table = sta529_i2c_id, +}; + +module_i2c_driver(sta529_i2c_driver); + +MODULE_DESCRIPTION("ASoC STA529 codec driver"); +MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index df1e07ffac32..31762ebdd774 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -34,8 +34,6 @@ #include "tlv320aic23.h" -#define AIC23_VERSION "0.1" - /* * AIC23 register cache */ @@ -325,8 +323,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 iface_reg; int ret; struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); @@ -371,8 +368,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; /* set active */ snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001); @@ -383,8 +379,7 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); /* deactivate */ @@ -548,8 +543,6 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec) struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); int ret; - printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); - ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 802064b5030d..85944e953578 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -126,8 +126,7 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec); int fsref, divisor, wlen, pval, jval, dval, qval; u16 reg; diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 8d20f6ec20f3..dc78f5a4bcbf 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -118,7 +118,9 @@ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { 0x00, 0x00, 0x00, 0x00, /* 88 */ 0x00, 0x00, 0x00, 0x00, /* 92 */ 0x00, 0x00, 0x00, 0x00, /* 96 */ - 0x00, 0x00, 0x02, /* 100 */ + 0x00, 0x00, 0x02, 0x00, /* 100 */ + 0x00, 0x00, 0x00, 0x00, /* 104 */ + 0x00, 0x00, /* 108 */ }; #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ @@ -229,6 +231,25 @@ static const struct soc_enum aic3x_enum[] = { SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), }; +static const char *aic3x_agc_level[] = + { "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" }; +static const struct soc_enum aic3x_agc_level_enum[] = { + SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level), +}; + +static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" }; +static const struct soc_enum aic3x_agc_attack_enum[] = { + SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack), +}; + +static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" }; +static const struct soc_enum aic3x_agc_decay_enum[] = { + SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), +}; + /* * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps */ @@ -353,6 +374,15 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { * adjust PGA to max value when ADC is on and will never go back. */ SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), + SOC_ENUM("Left AGC Target level", aic3x_agc_level_enum[0]), + SOC_ENUM("Right AGC Target level", aic3x_agc_level_enum[1]), + SOC_ENUM("Left AGC Attack time", aic3x_agc_attack_enum[0]), + SOC_ENUM("Right AGC Attack time", aic3x_agc_attack_enum[1]), + SOC_ENUM("Left AGC Decay time", aic3x_agc_decay_enum[0]), + SOC_ENUM("Right AGC Decay time", aic3x_agc_decay_enum[1]), + + /* De-emphasis */ + SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0), /* Input */ SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL, @@ -368,7 +398,7 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { static DECLARE_TLV_DB_SCALE(classd_amp_tlv, 0, 600, 0); static const struct snd_kcontrol_new aic3x_classd_amp_gain_ctrl = - SOC_DOUBLE_TLV("Class-D Amplifier Gain", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv); + SOC_DOUBLE_TLV("Class-D Playback Volume", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv); /* Left DAC Mux */ static const struct snd_kcontrol_new aic3x_left_dac_mux_controls = @@ -802,8 +832,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; @@ -936,9 +965,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, } found: - data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); - snd_soc_write(codec, AIC3X_PLL_PROGA_REG, - data | (pll_p << PLLP_SHIFT)); + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLLP_MASK, pll_p); snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, pll_r << PLLR_SHIFT); snd_soc_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT); @@ -973,6 +1000,12 @@ static int aic3x_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + /* set clock on MCLK or GPIO2 or BCLK */ + snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, PLLCLK_IN_MASK, + clk_id << PLLCLK_IN_SHIFT); + snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, CLKDIV_IN_MASK, + clk_id << CLKDIV_IN_SHIFT); + aic3x->sysclk = freq; return 0; } @@ -1161,24 +1194,6 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, return 0; } -void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect, - int headset_debounce, int button_debounce) -{ - u8 val; - - val = ((detect & AIC3X_HEADSET_DETECT_MASK) - << AIC3X_HEADSET_DETECT_SHIFT) | - ((headset_debounce & AIC3X_HEADSET_DEBOUNCE_MASK) - << AIC3X_HEADSET_DEBOUNCE_SHIFT) | - ((button_debounce & AIC3X_BUTTON_DEBOUNCE_MASK) - << AIC3X_BUTTON_DEBOUNCE_SHIFT); - - if (detect & AIC3X_HEADSET_DETECT_MASK) - val |= AIC3X_HEADSET_DETECT_ENABLED; - - snd_soc_write(codec, AIC3X_HEADSET_DETECT_CTRL_A, val); -} - #define AIC3X_RATES SNDRV_PCM_RATE_8000_96000 #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 6f097fb60683..6db3c41b0163 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -13,7 +13,7 @@ #define _AIC3X_H /* AIC3X register space */ -#define AIC3X_CACHEREGNUM 103 +#define AIC3X_CACHEREGNUM 110 /* Page select register */ #define AIC3X_PAGE_SELECT 0 @@ -74,6 +74,8 @@ #define HPLCOM_CFG 37 /* Right High Power Output control registers */ #define HPRCOM_CFG 38 +/* High Power Output Stage Control Register */ +#define HPOUT_SC 40 /* DAC Output Switching control registers */ #define DAC_LINE_MUX 41 /* High Power Output Driver Pop Reduction registers */ @@ -148,6 +150,17 @@ #define AIC3X_GPIOB_REG 101 /* Clock generation control register */ #define AIC3X_CLKGEN_CTRL_REG 102 +/* New AGC registers */ +#define LAGCN_ATTACK 103 +#define LAGCN_DECAY 104 +#define RAGCN_ATTACK 105 +#define RAGCN_DECAY 106 +/* New Programmable ADC Digital Path and I2C Bus Condition Register */ +#define NEW_ADC_DIGITALPATH 107 +/* Passive Analog Signal Bypass Selection During Powerdown Register */ +#define PASSIVE_BYPASS 108 +/* DAC Quiescent Current Adjustment Register */ +#define DAC_ICC_ADJ 109 /* Page select register bits */ #define PAGE0_SELECT 0 @@ -163,9 +176,14 @@ #define DUAL_RATE_MODE ((1 << 5) | (1 << 6)) #define LDAC2LCH (0x1 << 3) #define RDAC2RCH (0x1 << 1) +#define LDAC2RCH (0x2 << 3) +#define RDAC2LCH (0x2 << 1) +#define LDAC2MONOMIX (0x3 << 3) +#define RDAC2MONOMIX (0x3 << 1) /* PLL registers bitfields */ #define PLLP_SHIFT 0 +#define PLLP_MASK 7 #define PLLQ_SHIFT 3 #define PLLR_SHIFT 0 #define PLLJ_SHIFT 2 @@ -178,6 +196,14 @@ #define PLL_CLKIN_SHIFT 4 #define MCLK_SOURCE 0x0 #define PLL_CLKDIV_SHIFT 0 +#define PLLCLK_IN_MASK 0x30 +#define PLLCLK_IN_SHIFT 4 +#define CLKDIV_IN_MASK 0xc0 +#define CLKDIV_IN_SHIFT 6 +/* clock in source */ +#define CLKIN_MCLK 0 +#define CLKIN_GPIO2 1 +#define CLKIN_BCLK 2 /* Software reset register bits */ #define SOFT_RESET 0x80 diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 4587ddd0fbf8..0dd41077ab79 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -62,8 +62,10 @@ #define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \ (((samples)*5000) / (((burstrate)*5000) / ((burstrate) - (playrate)))) -static void dac33_calculate_times(struct snd_pcm_substream *substream); -static int dac33_prepare_chip(struct snd_pcm_substream *substream); +static void dac33_calculate_times(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec); +static int dac33_prepare_chip(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec); enum dac33_state { DAC33_IDLE = 0, @@ -427,8 +429,8 @@ static int dac33_playback_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: if (likely(dac33->substream)) { - dac33_calculate_times(dac33->substream); - dac33_prepare_chip(dac33->substream); + dac33_calculate_times(dac33->substream, w->codec); + dac33_prepare_chip(dac33->substream, w->codec); } break; case SND_SOC_DAPM_POST_PMD: @@ -799,8 +801,7 @@ static void dac33_oscwait(struct snd_soc_codec *codec) static int dac33_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); /* Stream started, save the substream pointer */ @@ -812,8 +813,7 @@ static int dac33_startup(struct snd_pcm_substream *substream, static void dac33_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); dac33->substream = NULL; @@ -825,8 +825,7 @@ static int dac33_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); /* Check parameters for validity */ @@ -868,10 +867,9 @@ static int dac33_hw_params(struct snd_pcm_substream *substream, * writes happens in different order, than dac33 might end up in unknown state. * Use the known, working sequence of register writes to initialize the dac33. */ -static int dac33_prepare_chip(struct snd_pcm_substream *substream) +static int dac33_prepare_chip(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned int oscset, ratioset, pwr_ctrl, reg_tmp; u8 aictrl_a, aictrl_b, fifoctrl_a; @@ -1067,10 +1065,9 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) return 0; } -static void dac33_calculate_times(struct snd_pcm_substream *substream) +static void dac33_calculate_times(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned int period_size = substream->runtime->period_size; unsigned int rate = substream->runtime->rate; @@ -1128,8 +1125,7 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); int ret = 0; @@ -1161,8 +1157,7 @@ static snd_pcm_sframes_t dac33_dai_delay( struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned long long t0, t1, t_now; unsigned int time_delta, uthr; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 170cf9a8fc79..391fcfc7b63b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1685,8 +1685,7 @@ static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction, static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); if (twl4030->master_substream) { @@ -1715,8 +1714,7 @@ static int twl4030_startup(struct snd_pcm_substream *substream, static void twl4030_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); if (twl4030->master_substream == substream) @@ -1740,8 +1738,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 mode, old_mode, format, old_format; @@ -1974,8 +1971,7 @@ static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, static int twl4030_voice_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 mode; @@ -2007,8 +2003,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; /* Enable voice digital filters */ twl4030_voice_enable(codec, substream->stream, 0); @@ -2017,8 +2012,7 @@ static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 old_mode, mode; diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index dc7509b9d53a..c084c549942e 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -46,17 +46,6 @@ #define TWL6040_OUTHF_0dB 0x03 #define TWL6040_OUTHF_M52dB 0x1D -#define TWL6040_RAMP_NONE 0 -#define TWL6040_RAMP_UP 1 -#define TWL6040_RAMP_DOWN 2 - -#define TWL6040_HSL_VOL_MASK 0x0F -#define TWL6040_HSL_VOL_SHIFT 0 -#define TWL6040_HSR_VOL_MASK 0xF0 -#define TWL6040_HSR_VOL_SHIFT 4 -#define TWL6040_HF_VOL_MASK 0x1F -#define TWL6040_HF_VOL_SHIFT 0 - /* Shadow register used by the driver */ #define TWL6040_REG_SW_SHADOW 0x2F #define TWL6040_CACHEREGNUM (TWL6040_REG_SW_SHADOW + 1) @@ -64,18 +53,6 @@ /* TWL6040_REG_SW_SHADOW (0x2F) fields */ #define TWL6040_EAR_PATH_ENABLE 0x01 -struct twl6040_output { - u16 active; - u16 left_vol; - u16 right_vol; - u16 left_step; - u16 right_step; - unsigned int step_delay; - u16 ramp; - struct delayed_work work; - struct completion ramp_done; -}; - struct twl6040_jack_data { struct snd_soc_jack *jack; struct delayed_work work; @@ -100,8 +77,6 @@ struct twl6040_data { struct snd_soc_codec *codec; struct workqueue_struct *workqueue; struct mutex mutex; - struct twl6040_output headset; - struct twl6040_output handsfree; }; /* @@ -311,318 +286,6 @@ static void twl6040_restore_regs(struct snd_soc_codec *codec) } } -/* - * Ramp HS PGA volume to minimise pops at stream startup and shutdown. - */ -static inline int twl6040_hs_ramp_step(struct snd_soc_codec *codec, - unsigned int left_step, unsigned int right_step) -{ - - struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *headset = &priv->headset; - int left_complete = 0, right_complete = 0; - u8 reg, val; - - /* left channel */ - left_step = (left_step > 0xF) ? 0xF : left_step; - reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN); - val = (~reg & TWL6040_HSL_VOL_MASK); - - if (headset->ramp == TWL6040_RAMP_UP) { - /* ramp step up */ - if (val < headset->left_vol) { - if (val + left_step > headset->left_vol) - val = headset->left_vol; - else - val += left_step; - - reg &= ~TWL6040_HSL_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HSGAIN, - (reg | (~val & TWL6040_HSL_VOL_MASK))); - } else { - left_complete = 1; - } - } else if (headset->ramp == TWL6040_RAMP_DOWN) { - /* ramp step down */ - if (val > 0x0) { - if ((int)val - (int)left_step < 0) - val = 0; - else - val -= left_step; - - reg &= ~TWL6040_HSL_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HSGAIN, reg | - (~val & TWL6040_HSL_VOL_MASK)); - } else { - left_complete = 1; - } - } - - /* right channel */ - right_step = (right_step > 0xF) ? 0xF : right_step; - reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN); - val = (~reg & TWL6040_HSR_VOL_MASK) >> TWL6040_HSR_VOL_SHIFT; - - if (headset->ramp == TWL6040_RAMP_UP) { - /* ramp step up */ - if (val < headset->right_vol) { - if (val + right_step > headset->right_vol) - val = headset->right_vol; - else - val += right_step; - - reg &= ~TWL6040_HSR_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HSGAIN, - (reg | (~val << TWL6040_HSR_VOL_SHIFT))); - } else { - right_complete = 1; - } - } else if (headset->ramp == TWL6040_RAMP_DOWN) { - /* ramp step down */ - if (val > 0x0) { - if ((int)val - (int)right_step < 0) - val = 0; - else - val -= right_step; - - reg &= ~TWL6040_HSR_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HSGAIN, - reg | (~val << TWL6040_HSR_VOL_SHIFT)); - } else { - right_complete = 1; - } - } - - return left_complete & right_complete; -} - -/* - * Ramp HF PGA volume to minimise pops at stream startup and shutdown. - */ -static inline int twl6040_hf_ramp_step(struct snd_soc_codec *codec, - unsigned int left_step, unsigned int right_step) -{ - struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *handsfree = &priv->handsfree; - int left_complete = 0, right_complete = 0; - u16 reg, val; - - /* left channel */ - left_step = (left_step > 0x1D) ? 0x1D : left_step; - reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFLGAIN); - reg = 0x1D - reg; - val = (reg & TWL6040_HF_VOL_MASK); - if (handsfree->ramp == TWL6040_RAMP_UP) { - /* ramp step up */ - if (val < handsfree->left_vol) { - if (val + left_step > handsfree->left_vol) - val = handsfree->left_vol; - else - val += left_step; - - reg &= ~TWL6040_HF_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HFLGAIN, - reg | (0x1D - val)); - } else { - left_complete = 1; - } - } else if (handsfree->ramp == TWL6040_RAMP_DOWN) { - /* ramp step down */ - if (val > 0) { - if ((int)val - (int)left_step < 0) - val = 0; - else - val -= left_step; - - reg &= ~TWL6040_HF_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HFLGAIN, - reg | (0x1D - val)); - } else { - left_complete = 1; - } - } - - /* right channel */ - right_step = (right_step > 0x1D) ? 0x1D : right_step; - reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFRGAIN); - reg = 0x1D - reg; - val = (reg & TWL6040_HF_VOL_MASK); - if (handsfree->ramp == TWL6040_RAMP_UP) { - /* ramp step up */ - if (val < handsfree->right_vol) { - if (val + right_step > handsfree->right_vol) - val = handsfree->right_vol; - else - val += right_step; - - reg &= ~TWL6040_HF_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HFRGAIN, - reg | (0x1D - val)); - } else { - right_complete = 1; - } - } else if (handsfree->ramp == TWL6040_RAMP_DOWN) { - /* ramp step down */ - if (val > 0) { - if ((int)val - (int)right_step < 0) - val = 0; - else - val -= right_step; - - reg &= ~TWL6040_HF_VOL_MASK; - twl6040_write(codec, TWL6040_REG_HFRGAIN, - reg | (0x1D - val)); - } - } - - return left_complete & right_complete; -} - -/* - * This work ramps both output PGAs at stream start/stop time to - * minimise pop associated with DAPM power switching. - */ -static void twl6040_pga_hs_work(struct work_struct *work) -{ - struct twl6040_data *priv = - container_of(work, struct twl6040_data, headset.work.work); - struct snd_soc_codec *codec = priv->codec; - struct twl6040_output *headset = &priv->headset; - int i, headset_complete; - - /* do we need to ramp at all ? */ - if (headset->ramp == TWL6040_RAMP_NONE) - return; - - /* HS PGA gain range: 0x0 - 0xf (0 - 15) */ - for (i = 0; i < 16; i++) { - headset_complete = twl6040_hs_ramp_step(codec, - headset->left_step, - headset->right_step); - - /* ramp finished ? */ - if (headset_complete) - break; - - schedule_timeout_interruptible( - msecs_to_jiffies(headset->step_delay)); - } - - if (headset->ramp == TWL6040_RAMP_DOWN) { - headset->active = 0; - complete(&headset->ramp_done); - } else { - headset->active = 1; - } - headset->ramp = TWL6040_RAMP_NONE; -} - -static void twl6040_pga_hf_work(struct work_struct *work) -{ - struct twl6040_data *priv = - container_of(work, struct twl6040_data, handsfree.work.work); - struct snd_soc_codec *codec = priv->codec; - struct twl6040_output *handsfree = &priv->handsfree; - int i, handsfree_complete; - - /* do we need to ramp at all ? */ - if (handsfree->ramp == TWL6040_RAMP_NONE) - return; - - /* - * HF PGA gain range: 0x00 - 0x1d (0 - 29) */ - for (i = 0; i < 30; i++) { - handsfree_complete = twl6040_hf_ramp_step(codec, - handsfree->left_step, - handsfree->right_step); - - /* ramp finished ? */ - if (handsfree_complete) - break; - - schedule_timeout_interruptible( - msecs_to_jiffies(handsfree->step_delay)); - } - - - if (handsfree->ramp == TWL6040_RAMP_DOWN) { - handsfree->active = 0; - complete(&handsfree->ramp_done); - } else - handsfree->active = 1; - handsfree->ramp = TWL6040_RAMP_NONE; -} - -static int out_drv_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *out; - struct delayed_work *work; - - switch (w->shift) { - case 2: /* Headset output driver */ - out = &priv->headset; - work = &out->work; - /* - * Make sure, that we do not mess up variables for already - * executing work. - */ - cancel_delayed_work_sync(work); - - out->left_step = priv->hs_left_step; - out->right_step = priv->hs_right_step; - out->step_delay = 5; /* 5 ms between volume ramp steps */ - break; - case 4: /* Handsfree output driver */ - out = &priv->handsfree; - work = &out->work; - /* - * Make sure, that we do not mess up variables for already - * executing work. - */ - cancel_delayed_work_sync(work); - - out->left_step = priv->hf_left_step; - out->right_step = priv->hf_right_step; - out->step_delay = 5; /* 5 ms between volume ramp steps */ - break; - default: - return -1; - } - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - if (out->active) - break; - - /* don't use volume ramp for power-up */ - out->ramp = TWL6040_RAMP_UP; - out->left_step = out->left_vol; - out->right_step = out->right_vol; - - queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1)); - break; - - case SND_SOC_DAPM_PRE_PMD: - if (!out->active) - break; - - /* use volume ramp for power-down */ - out->ramp = TWL6040_RAMP_DOWN; - INIT_COMPLETION(out->ramp_done); - - queue_delayed_work(priv->workqueue, work, msecs_to_jiffies(1)); - - wait_for_completion_timeout(&out->ramp_done, - msecs_to_jiffies(2000)); - break; - } - - return 0; -} - /* set headset dac and driver power mode */ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) { @@ -747,71 +410,6 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data) return IRQ_HANDLED; } -static int twl6040_put_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *out = NULL; - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int ret; - - /* For HS and HF we shadow the values and only actually write - * them out when active in order to ensure the amplifier comes on - * as quietly as possible. */ - switch (mc->reg) { - case TWL6040_REG_HSGAIN: - out = &twl6040_priv->headset; - break; - case TWL6040_REG_HFLGAIN: - out = &twl6040_priv->handsfree; - break; - default: - dev_warn(codec->dev, "%s: Unexpected register: 0x%02x\n", - __func__, mc->reg); - return -EINVAL; - } - - out->left_vol = ucontrol->value.integer.value[0]; - out->right_vol = ucontrol->value.integer.value[1]; - if (!out->active) - return 1; - - ret = snd_soc_put_volsw(kcontrol, ucontrol); - if (ret < 0) - return ret; - - return 1; -} - -static int twl6040_get_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec); - struct twl6040_output *out = &twl6040_priv->headset; - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - - switch (mc->reg) { - case TWL6040_REG_HSGAIN: - out = &twl6040_priv->headset; - break; - case TWL6040_REG_HFLGAIN: - out = &twl6040_priv->handsfree; - break; - default: - dev_warn(codec->dev, "%s: Unexpected register: 0x%02x\n", - __func__, mc->reg); - return -EINVAL; - } - - ucontrol->value.integer.value[0] = out->left_vol; - ucontrol->value.integer.value[1] = out->right_vol; - return 0; -} - static int twl6040_soc_dapm_put_vibra_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -955,7 +553,7 @@ static const struct snd_kcontrol_new vibrar_mux_controls = /* Headset power mode */ static const char *twl6040_power_mode_texts[] = { - "Low-Power", "High-Perfomance", + "Low-Power", "High-Performance", }; static const struct soc_enum twl6040_power_mode_enum = @@ -1055,7 +653,7 @@ int twl6040_get_hs_step_size(struct snd_soc_codec *codec) { struct twl6040 *twl6040 = codec->control_data; - if (twl6040_get_revid(twl6040) < TWL6040_REV_ES1_2) + if (twl6040_get_revid(twl6040) < TWL6040_REV_ES1_3) /* For ES under ES_1.3 HS step is 2 mV */ return 2; else @@ -1076,12 +674,10 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { TWL6040_REG_LINEGAIN, 0, 3, 7, 0, afm_amp_tlv), /* Playback gains */ - SOC_DOUBLE_EXT_TLV("Headset Playback Volume", - TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, twl6040_get_volsw, - twl6040_put_volsw, hs_tlv), - SOC_DOUBLE_R_EXT_TLV("Handsfree Playback Volume", - TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, - twl6040_get_volsw, twl6040_put_volsw, hf_tlv), + SOC_DOUBLE_TLV("Headset Playback Volume", + TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), SOC_SINGLE_TLV("Earphone Playback Volume", TWL6040_REG_EARCTL, 1, 0xF, 1, ep_tlv), @@ -1180,22 +776,14 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { &auxr_switch_control), /* Analog playback drivers */ - SND_SOC_DAPM_OUT_DRV_E("HF Left Driver", - TWL6040_REG_HFLCTL, 4, 0, NULL, 0, - out_drv_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_OUT_DRV_E("HF Right Driver", - TWL6040_REG_HFRCTL, 4, 0, NULL, 0, - out_drv_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_OUT_DRV_E("HS Left Driver", - TWL6040_REG_HSLCTL, 2, 0, NULL, 0, - out_drv_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_OUT_DRV_E("HS Right Driver", - TWL6040_REG_HSRCTL, 2, 0, NULL, 0, - out_drv_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_OUT_DRV("HF Left Driver", + TWL6040_REG_HFLCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HF Right Driver", + TWL6040_REG_HFRCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HS Left Driver", + TWL6040_REG_HSLCTL, 2, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HS Right Driver", + TWL6040_REG_HSRCTL, 2, 0, NULL, 0), SND_SOC_DAPM_OUT_DRV_E("Earphone Driver", TWL6040_REG_EARCTL, 0, 0, NULL, 0, twl6040_ep_drv_event, @@ -1339,8 +927,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, static int twl6040_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); snd_pcm_hw_constraint_list(substream->runtime, 0, @@ -1354,8 +941,7 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); int rate; @@ -1391,8 +977,7 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, static int twl6040_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl6040 *twl6040 = codec->control_data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); int ret; @@ -1570,14 +1155,9 @@ static int twl6040_probe(struct snd_soc_codec *codec) } INIT_DELAYED_WORK(&priv->hs_jack.work, twl6040_accessory_work); - INIT_DELAYED_WORK(&priv->headset.work, twl6040_pga_hs_work); - INIT_DELAYED_WORK(&priv->handsfree.work, twl6040_pga_hf_work); mutex_init(&priv->mutex); - init_completion(&priv->headset.ramp_done); - init_completion(&priv->handsfree.ramp_done); - ret = request_threaded_irq(priv->plug_irq, NULL, twl6040_audio_handler, 0, "twl6040_irq_plug", codec); if (ret) { diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 797b0dde2c68..6c3d43b8ee85 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -159,8 +159,7 @@ static int uda134x_mute(struct snd_soc_dai *dai, int mute) static int uda134x_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); struct snd_pcm_runtime *master_runtime; @@ -191,8 +190,7 @@ static int uda134x_startup(struct snd_pcm_substream *substream, static void uda134x_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); if (uda134x->master_substream == substream) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 4f1b23d7e404..2502214b84ab 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -502,8 +502,7 @@ static int uda1380_set_dai_fmt_capture(struct snd_soc_dai *codec_dai, static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda1380_priv *uda1380 = snd_soc_codec_get_drvdata(codec); int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER); @@ -528,8 +527,7 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); /* set WSPLL power and divider if running from this clock */ diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 3d868dc40092..7b24d6d192e1 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -293,8 +293,7 @@ static const struct snd_kcontrol_new wl1273_controls[] = { static int wl1273_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); switch (wl1273->mode) { @@ -329,8 +328,7 @@ static int wl1273_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(rtd->codec); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(dai->codec); struct wl1273_core *core = wl1273->core; unsigned int rate, width, r; diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index aefb4f89be0e..951d7b49476a 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -79,22 +79,68 @@ static const struct snd_soc_dapm_route wm1250_ev1_dapm_routes[] = { { "WM1250 Output", NULL, "DAC" }, }; +static int wm1250_ev1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct wm1250_priv *wm1250 = snd_soc_codec_get_drvdata(dai->codec); + + switch (params_rate(params)) { + case 8000: + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio, + 1); + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio, + 1); + break; + case 16000: + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio, + 0); + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio, + 1); + break; + case 32000: + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio, + 1); + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio, + 0); + break; + case 64000: + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio, + 0); + gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio, + 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops wm1250_ev1_ops = { + .hw_params = wm1250_ev1_hw_params, +}; + +#define WM1250_EV1_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_64000) + static struct snd_soc_dai_driver wm1250_ev1_dai = { .name = "wm1250-ev1", .playback = { .stream_name = "Playback", .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, + .channels_max = 2, + .rates = WM1250_EV1_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "Capture", .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, + .channels_max = 2, + .rates = WM1250_EV1_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .ops = &wm1250_ev1_ops, }; static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = { @@ -215,23 +261,7 @@ static struct i2c_driver wm1250_ev1_i2c_driver = { .id_table = wm1250_ev1_i2c_id, }; -static int __init wm1250_ev1_modinit(void) -{ - int ret = 0; - - ret = i2c_add_driver(&wm1250_ev1_i2c_driver); - if (ret != 0) - pr_err("Failed to register WM1250-EV1 I2C driver: %d\n", ret); - - return ret; -} -module_init(wm1250_ev1_modinit); - -static void __exit wm1250_ev1_exit(void) -{ - i2c_del_driver(&wm1250_ev1_i2c_driver); -} -module_exit(wm1250_ev1_exit); +module_i2c_driver(wm1250_ev1_i2c_driver); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); MODULE_DESCRIPTION("WM1250-EV1 audio I/O module driver"); diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index a75c3766aede..3fd5b29dc933 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -1,7 +1,7 @@ /* * wm2000.c -- WM2000 ALSA Soc Audio driver * - * Copyright 2008-2010 Wolfson Microelectronics PLC. + * Copyright 2008-2011 Wolfson Microelectronics PLC. * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * @@ -99,8 +99,9 @@ static void wm2000_reset(struct wm2000_priv *wm2000) } static int wm2000_poll_bit(struct i2c_client *i2c, - unsigned int reg, u8 mask, int timeout) + unsigned int reg, u8 mask) { + int timeout = 4000; int val; val = wm2000_read(i2c, reg); @@ -119,7 +120,7 @@ static int wm2000_poll_bit(struct i2c_client *i2c, static int wm2000_power_up(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); - int ret, timeout; + int ret; BUG_ON(wm2000->anc_mode != ANC_OFF); @@ -140,13 +141,13 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) /* Wait for ANC engine to become ready */ if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, - WM2000_ANC_ENG_IDLE, 1)) { + WM2000_ANC_ENG_IDLE)) { dev_err(&i2c->dev, "ANC engine failed to reset\n"); return -ETIMEDOUT; } if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, - WM2000_STATUS_BOOT_COMPLETE, 1)) { + WM2000_STATUS_BOOT_COMPLETE)) { dev_err(&i2c->dev, "ANC engine failed to initialise\n"); return -ETIMEDOUT; } @@ -173,16 +174,13 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) dev_dbg(&i2c->dev, "Download complete\n"); if (analogue) { - timeout = 248; - wm2000_write(i2c, WM2000_REG_ANA_VMID_PU_TIME, timeout / 4); + wm2000_write(i2c, WM2000_REG_ANA_VMID_PU_TIME, 248 / 4); wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, WM2000_MODE_ANA_SEQ_INCLUDE | WM2000_MODE_MOUSE_ENABLE | WM2000_MODE_THERMAL_ENABLE); } else { - timeout = 10; - wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, WM2000_MODE_MOUSE_ENABLE | WM2000_MODE_THERMAL_ENABLE); @@ -201,9 +199,8 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_INT_N_CLR); if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, - WM2000_STATUS_MOUSE_ACTIVE, timeout)) { - dev_err(&i2c->dev, "Timed out waiting for device after %dms\n", - timeout * 10); + WM2000_STATUS_MOUSE_ACTIVE)) { + dev_err(&i2c->dev, "Timed out waiting for device\n"); return -ETIMEDOUT; } @@ -218,28 +215,25 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) static int wm2000_power_down(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); - int timeout; if (analogue) { - timeout = 248; - wm2000_write(i2c, WM2000_REG_ANA_VMID_PD_TIME, timeout / 4); + wm2000_write(i2c, WM2000_REG_ANA_VMID_PD_TIME, 248 / 4); wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, WM2000_MODE_ANA_SEQ_INCLUDE | WM2000_MODE_POWER_DOWN); } else { - timeout = 10; wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, WM2000_MODE_POWER_DOWN); } if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, - WM2000_STATUS_POWER_DOWN_COMPLETE, timeout)) { + WM2000_STATUS_POWER_DOWN_COMPLETE)) { dev_err(&i2c->dev, "Timeout waiting for ANC power down\n"); return -ETIMEDOUT; } if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, - WM2000_ANC_ENG_IDLE, 1)) { + WM2000_ANC_ENG_IDLE)) { dev_err(&i2c->dev, "Timeout waiting for ANC engine idle\n"); return -ETIMEDOUT; } @@ -268,13 +262,13 @@ static int wm2000_enter_bypass(struct i2c_client *i2c, int analogue) } if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, - WM2000_STATUS_ANC_DISABLED, 10)) { + WM2000_STATUS_ANC_DISABLED)) { dev_err(&i2c->dev, "Timeout waiting for ANC disable\n"); return -ETIMEDOUT; } if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, - WM2000_ANC_ENG_IDLE, 1)) { + WM2000_ANC_ENG_IDLE)) { dev_err(&i2c->dev, "Timeout waiting for ANC engine idle\n"); return -ETIMEDOUT; } @@ -311,7 +305,7 @@ static int wm2000_exit_bypass(struct i2c_client *i2c, int analogue) wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_INT_N_CLR); if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, - WM2000_STATUS_MOUSE_ACTIVE, 10)) { + WM2000_STATUS_MOUSE_ACTIVE)) { dev_err(&i2c->dev, "Timed out waiting for MOUSE\n"); return -ETIMEDOUT; } @@ -325,38 +319,32 @@ static int wm2000_exit_bypass(struct i2c_client *i2c, int analogue) static int wm2000_enter_standby(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); - int timeout; BUG_ON(wm2000->anc_mode != ANC_ACTIVE); if (analogue) { - timeout = 248; - wm2000_write(i2c, WM2000_REG_ANA_VMID_PD_TIME, timeout / 4); + wm2000_write(i2c, WM2000_REG_ANA_VMID_PD_TIME, 248 / 4); wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, WM2000_MODE_ANA_SEQ_INCLUDE | WM2000_MODE_THERMAL_ENABLE | WM2000_MODE_STANDBY_ENTRY); } else { - timeout = 10; - wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, WM2000_MODE_THERMAL_ENABLE | WM2000_MODE_STANDBY_ENTRY); } if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, - WM2000_STATUS_ANC_DISABLED, timeout)) { + WM2000_STATUS_ANC_DISABLED)) { dev_err(&i2c->dev, "Timed out waiting for ANC disable after 1ms\n"); return -ETIMEDOUT; } - if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, WM2000_ANC_ENG_IDLE, - 1)) { + if (!wm2000_poll_bit(i2c, WM2000_REG_ANC_STAT, WM2000_ANC_ENG_IDLE)) { dev_err(&i2c->dev, - "Timed out waiting for standby after %dms\n", - timeout * 10); + "Timed out waiting for standby\n"); return -ETIMEDOUT; } @@ -374,23 +362,19 @@ static int wm2000_enter_standby(struct i2c_client *i2c, int analogue) static int wm2000_exit_standby(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); - int timeout; BUG_ON(wm2000->anc_mode != ANC_STANDBY); wm2000_write(i2c, WM2000_REG_SYS_CTL1, 0); if (analogue) { - timeout = 248; - wm2000_write(i2c, WM2000_REG_ANA_VMID_PU_TIME, timeout / 4); + wm2000_write(i2c, WM2000_REG_ANA_VMID_PU_TIME, 248 / 4); wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, WM2000_MODE_ANA_SEQ_INCLUDE | WM2000_MODE_THERMAL_ENABLE | WM2000_MODE_MOUSE_ENABLE); } else { - timeout = 10; - wm2000_write(i2c, WM2000_REG_SYS_MODE_CNTRL, WM2000_MODE_THERMAL_ENABLE | WM2000_MODE_MOUSE_ENABLE); @@ -400,9 +384,8 @@ static int wm2000_exit_standby(struct i2c_client *i2c, int analogue) wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_ANC_INT_N_CLR); if (!wm2000_poll_bit(i2c, WM2000_REG_SYS_STATUS, - WM2000_STATUS_MOUSE_ACTIVE, timeout)) { - dev_err(&i2c->dev, "Timed out waiting for MOUSE after %dms\n", - timeout * 10); + WM2000_STATUS_MOUSE_ACTIVE)) { + dev_err(&i2c->dev, "Timed out waiting for MOUSE\n"); return -ETIMEDOUT; } @@ -691,9 +674,39 @@ static int wm2000_resume(struct snd_soc_codec *codec) #define wm2000_resume NULL #endif +static bool wm2000_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM2000_REG_SYS_START: + case WM2000_REG_SPEECH_CLARITY: + case WM2000_REG_SYS_WATCHDOG: + case WM2000_REG_ANA_VMID_PD_TIME: + case WM2000_REG_ANA_VMID_PU_TIME: + case WM2000_REG_CAT_FLTR_INDX: + case WM2000_REG_CAT_GAIN_0: + case WM2000_REG_SYS_STATUS: + case WM2000_REG_SYS_MODE_CNTRL: + case WM2000_REG_SYS_START0: + case WM2000_REG_SYS_START1: + case WM2000_REG_ID1: + case WM2000_REG_ID2: + case WM2000_REG_REVISON: + case WM2000_REG_SYS_CTL1: + case WM2000_REG_SYS_CTL2: + case WM2000_REG_ANC_STAT: + case WM2000_REG_IF_CTL: + return true; + default: + return false; + } +} + static const struct regmap_config wm2000_regmap = { .reg_bits = 8, .val_bits = 8, + + .max_register = WM2000_REG_IF_CTL, + .readable_reg = wm2000_readable_reg, }; static int wm2000_probe(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index acbdc5fde923..32682c1b7cde 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1491,6 +1491,7 @@ static int wm2200_bclk_rates_dat[WM2200_NUM_BCLK_RATES] = { static int wm2200_bclk_rates_cd[WM2200_NUM_BCLK_RATES] = { 5644800, + 3763200, 2882400, 1881600, 1411200, diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c index 9a18fae68204..e239f4bf2460 100644 --- a/sound/soc/codecs/wm5100-tables.c +++ b/sound/soc/codecs/wm5100-tables.c @@ -1,7 +1,7 @@ /* * wm5100-tables.c -- WM5100 ALSA SoC Audio driver data * - * Copyright 2011 Wolfson Microelectronics plc + * Copyright 2011-2 Wolfson Microelectronics plc * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * @@ -32,7 +32,18 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg) case WM5100_MIC_DETECT_3: return 1; default: - return 0; + if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) || + (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) || + (reg >= WM5100_DSP1_DM_0 && reg <= WM5100_DSP1_DM_511) || + (reg >= WM5100_DSP2_PM_0 && reg <= WM5100_DSP2_PM_1535) || + (reg >= WM5100_DSP2_ZM_0 && reg <= WM5100_DSP2_ZM_2047) || + (reg >= WM5100_DSP2_DM_0 && reg <= WM5100_DSP2_DM_511) || + (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) || + (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) || + (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511)) + return 1; + else + return 0; } } @@ -697,9 +708,110 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg) case WM5100_HPLPF3_2: case WM5100_HPLPF4_1: case WM5100_HPLPF4_2: + case WM5100_DSP1_CONTROL_1: + case WM5100_DSP1_CONTROL_2: + case WM5100_DSP1_CONTROL_3: + case WM5100_DSP1_CONTROL_4: + case WM5100_DSP1_CONTROL_5: + case WM5100_DSP1_CONTROL_6: + case WM5100_DSP1_CONTROL_7: + case WM5100_DSP1_CONTROL_8: + case WM5100_DSP1_CONTROL_9: + case WM5100_DSP1_CONTROL_10: + case WM5100_DSP1_CONTROL_11: + case WM5100_DSP1_CONTROL_12: + case WM5100_DSP1_CONTROL_13: + case WM5100_DSP1_CONTROL_14: + case WM5100_DSP1_CONTROL_15: + case WM5100_DSP1_CONTROL_16: + case WM5100_DSP1_CONTROL_17: + case WM5100_DSP1_CONTROL_18: + case WM5100_DSP1_CONTROL_19: + case WM5100_DSP1_CONTROL_20: + case WM5100_DSP1_CONTROL_21: + case WM5100_DSP1_CONTROL_22: + case WM5100_DSP1_CONTROL_23: + case WM5100_DSP1_CONTROL_24: + case WM5100_DSP1_CONTROL_25: + case WM5100_DSP1_CONTROL_26: + case WM5100_DSP1_CONTROL_27: + case WM5100_DSP1_CONTROL_28: + case WM5100_DSP1_CONTROL_29: + case WM5100_DSP1_CONTROL_30: + case WM5100_DSP2_CONTROL_1: + case WM5100_DSP2_CONTROL_2: + case WM5100_DSP2_CONTROL_3: + case WM5100_DSP2_CONTROL_4: + case WM5100_DSP2_CONTROL_5: + case WM5100_DSP2_CONTROL_6: + case WM5100_DSP2_CONTROL_7: + case WM5100_DSP2_CONTROL_8: + case WM5100_DSP2_CONTROL_9: + case WM5100_DSP2_CONTROL_10: + case WM5100_DSP2_CONTROL_11: + case WM5100_DSP2_CONTROL_12: + case WM5100_DSP2_CONTROL_13: + case WM5100_DSP2_CONTROL_14: + case WM5100_DSP2_CONTROL_15: + case WM5100_DSP2_CONTROL_16: + case WM5100_DSP2_CONTROL_17: + case WM5100_DSP2_CONTROL_18: + case WM5100_DSP2_CONTROL_19: + case WM5100_DSP2_CONTROL_20: + case WM5100_DSP2_CONTROL_21: + case WM5100_DSP2_CONTROL_22: + case WM5100_DSP2_CONTROL_23: + case WM5100_DSP2_CONTROL_24: + case WM5100_DSP2_CONTROL_25: + case WM5100_DSP2_CONTROL_26: + case WM5100_DSP2_CONTROL_27: + case WM5100_DSP2_CONTROL_28: + case WM5100_DSP2_CONTROL_29: + case WM5100_DSP2_CONTROL_30: + case WM5100_DSP3_CONTROL_1: + case WM5100_DSP3_CONTROL_2: + case WM5100_DSP3_CONTROL_3: + case WM5100_DSP3_CONTROL_4: + case WM5100_DSP3_CONTROL_5: + case WM5100_DSP3_CONTROL_6: + case WM5100_DSP3_CONTROL_7: + case WM5100_DSP3_CONTROL_8: + case WM5100_DSP3_CONTROL_9: + case WM5100_DSP3_CONTROL_10: + case WM5100_DSP3_CONTROL_11: + case WM5100_DSP3_CONTROL_12: + case WM5100_DSP3_CONTROL_13: + case WM5100_DSP3_CONTROL_14: + case WM5100_DSP3_CONTROL_15: + case WM5100_DSP3_CONTROL_16: + case WM5100_DSP3_CONTROL_17: + case WM5100_DSP3_CONTROL_18: + case WM5100_DSP3_CONTROL_19: + case WM5100_DSP3_CONTROL_20: + case WM5100_DSP3_CONTROL_21: + case WM5100_DSP3_CONTROL_22: + case WM5100_DSP3_CONTROL_23: + case WM5100_DSP3_CONTROL_24: + case WM5100_DSP3_CONTROL_25: + case WM5100_DSP3_CONTROL_26: + case WM5100_DSP3_CONTROL_27: + case WM5100_DSP3_CONTROL_28: + case WM5100_DSP3_CONTROL_29: + case WM5100_DSP3_CONTROL_30: return 1; default: - return 0; + if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) || + (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) || + (reg >= WM5100_DSP1_DM_0 && reg <= WM5100_DSP1_DM_511) || + (reg >= WM5100_DSP2_PM_0 && reg <= WM5100_DSP2_PM_1535) || + (reg >= WM5100_DSP2_ZM_0 && reg <= WM5100_DSP2_ZM_2047) || + (reg >= WM5100_DSP2_DM_0 && reg <= WM5100_DSP2_DM_511) || + (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) || + (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) || + (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511)) + return 1; + else + return 0; } } @@ -1361,4 +1473,13 @@ struct reg_default wm5100_reg_defaults[WM5100_REGISTER_COUNT] = { { 0x0EC9, 0x0000 }, /* R3785 - HPLPF3_2 */ { 0x0ECC, 0x0000 }, /* R3788 - HPLPF4_1 */ { 0x0ECD, 0x0000 }, /* R3789 - HPLPF4_2 */ + { 0x0F02, 0x0000 }, /* R3842 - DSP1 Control 2 */ + { 0x0F03, 0x0000 }, /* R3843 - DSP1 Control 3 */ + { 0x0F04, 0x0000 }, /* R3844 - DSP1 Control 4 */ + { 0x1002, 0x0000 }, /* R4098 - DSP2 Control 2 */ + { 0x1003, 0x0000 }, /* R4099 - DSP2 Control 3 */ + { 0x1004, 0x0000 }, /* R4100 - DSP2 Control 4 */ + { 0x1102, 0x0000 }, /* R4354 - DSP3 Control 2 */ + { 0x1103, 0x0000 }, /* R4355 - DSP3 Control 3 */ + { 0x1104, 0x0000 }, /* R4356 - DSP3 Control 4 */ }; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index b9c185ce64e4..f4817292ef45 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1,7 +1,7 @@ /* * wm5100.c -- WM5100 ALSA SoC Audio driver * - * Copyright 2011 Wolfson Microelectronics plc + * Copyright 2011-2 Wolfson Microelectronics plc * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * @@ -1265,29 +1265,12 @@ static const __devinitdata struct reg_default wm5100_reva_patches[] = { { WM5100_AUDIO_IF_3_19, 1 }, }; -static int wm5100_dai_to_base(struct snd_soc_dai *dai) -{ - switch (dai->id) { - case 0: - return WM5100_AUDIO_IF_1_1 - 1; - case 1: - return WM5100_AUDIO_IF_2_1 - 1; - case 2: - return WM5100_AUDIO_IF_3_1 - 1; - default: - BUG(); - return -EINVAL; - } -} - static int wm5100_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; int lrclk, bclk, mask, base; - base = wm5100_dai_to_base(dai); - if (base < 0) - return base; + base = dai->driver->base; lrclk = 0; bclk = 0; @@ -1414,9 +1397,7 @@ static int wm5100_hw_params(struct snd_pcm_substream *substream, int i, base, bclk, aif_rate, lrclk, wl, fl, sr; int *bclk_rates; - base = wm5100_dai_to_base(dai); - if (base < 0) - return base; + base = dai->driver->base; /* Data sizes if not using TDM */ wl = snd_pcm_format_width(params_format(params)); @@ -1897,6 +1878,7 @@ static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source, static struct snd_soc_dai_driver wm5100_dai[] = { { .name = "wm5100-aif1", + .base = WM5100_AUDIO_IF_1_1 - 1, .playback = { .stream_name = "AIF1 Playback", .channels_min = 2, @@ -1916,6 +1898,7 @@ static struct snd_soc_dai_driver wm5100_dai[] = { { .name = "wm5100-aif2", .id = 1, + .base = WM5100_AUDIO_IF_2_1 - 1, .playback = { .stream_name = "AIF2 Playback", .channels_min = 2, @@ -1935,6 +1918,7 @@ static struct snd_soc_dai_driver wm5100_dai[] = { { .name = "wm5100-aif3", .id = 2, + .base = WM5100_AUDIO_IF_3_1 - 1, .playback = { .stream_name = "AIF3 Playback", .channels_min = 2, @@ -2394,13 +2378,6 @@ static int wm5100_remove(struct snd_soc_codec *codec) return 0; } -static int wm5100_soc_volatile(struct snd_soc_codec *codec, - unsigned int reg) -{ - return true; -} - - static struct snd_soc_codec_driver soc_codec_dev_wm5100 = { .probe = wm5100_probe, .remove = wm5100_remove, @@ -2408,8 +2385,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm5100 = { .set_sysclk = wm5100_set_sysclk, .set_pll = wm5100_set_fll, .idle_bias_off = 1, - .reg_cache_size = WM5100_MAX_REGISTER, - .volatile_register = wm5100_soc_volatile, .seq_notifier = wm5100_seq_notifier, .controls = wm5100_snd_controls, @@ -2454,7 +2429,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, wm5100->dev = &i2c->dev; - wm5100->regmap = regmap_init_i2c(i2c, &wm5100_regmap); + wm5100->regmap = devm_regmap_init_i2c(i2c, &wm5100_regmap); if (IS_ERR(wm5100->regmap)) { ret = PTR_ERR(wm5100->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -2479,7 +2454,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, if (ret != 0) { dev_err(&i2c->dev, "Failed to request core supplies: %d\n", ret); - goto err_regmap; + goto err; } ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), @@ -2487,7 +2462,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, if (ret != 0) { dev_err(&i2c->dev, "Failed to enable core supplies: %d\n", ret); - goto err_regmap; + goto err; } if (wm5100->pdata.ldo_ena) { @@ -2660,8 +2635,6 @@ err_ldo: err_enable: regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); -err_regmap: - regmap_exit(wm5100->regmap); err: return ret; } @@ -2682,7 +2655,6 @@ static __devexit int wm5100_i2c_remove(struct i2c_client *i2c) gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); gpio_free(wm5100->pdata.ldo_ena); } - regmap_exit(wm5100->regmap); return 0; } @@ -2749,17 +2721,7 @@ static struct i2c_driver wm5100_i2c_driver = { .id_table = wm5100_i2c_id, }; -static int __init wm5100_modinit(void) -{ - return i2c_add_driver(&wm5100_i2c_driver); -} -module_init(wm5100_modinit); - -static void __exit wm5100_exit(void) -{ - i2c_del_driver(&wm5100_i2c_driver); -} -module_exit(wm5100_exit); +module_i2c_driver(wm5100_i2c_driver); MODULE_DESCRIPTION("ASoC WM5100 driver"); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); diff --git a/sound/soc/codecs/wm5100.h b/sound/soc/codecs/wm5100.h index 25cb6016f9d7..935a9b7fb274 100644 --- a/sound/soc/codecs/wm5100.h +++ b/sound/soc/codecs/wm5100.h @@ -709,6 +709,96 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); #define WM5100_HPLPF3_2 0xEC9 #define WM5100_HPLPF4_1 0xECC #define WM5100_HPLPF4_2 0xECD +#define WM5100_DSP1_CONTROL_1 0xF00 +#define WM5100_DSP1_CONTROL_2 0xF02 +#define WM5100_DSP1_CONTROL_3 0xF03 +#define WM5100_DSP1_CONTROL_4 0xF04 +#define WM5100_DSP1_CONTROL_5 0xF06 +#define WM5100_DSP1_CONTROL_6 0xF07 +#define WM5100_DSP1_CONTROL_7 0xF08 +#define WM5100_DSP1_CONTROL_8 0xF09 +#define WM5100_DSP1_CONTROL_9 0xF0A +#define WM5100_DSP1_CONTROL_10 0xF0B +#define WM5100_DSP1_CONTROL_11 0xF0C +#define WM5100_DSP1_CONTROL_12 0xF0D +#define WM5100_DSP1_CONTROL_13 0xF0F +#define WM5100_DSP1_CONTROL_14 0xF10 +#define WM5100_DSP1_CONTROL_15 0xF11 +#define WM5100_DSP1_CONTROL_16 0xF12 +#define WM5100_DSP1_CONTROL_17 0xF13 +#define WM5100_DSP1_CONTROL_18 0xF14 +#define WM5100_DSP1_CONTROL_19 0xF16 +#define WM5100_DSP1_CONTROL_20 0xF17 +#define WM5100_DSP1_CONTROL_21 0xF18 +#define WM5100_DSP1_CONTROL_22 0xF1A +#define WM5100_DSP1_CONTROL_23 0xF1B +#define WM5100_DSP1_CONTROL_24 0xF1C +#define WM5100_DSP1_CONTROL_25 0xF1E +#define WM5100_DSP1_CONTROL_26 0xF20 +#define WM5100_DSP1_CONTROL_27 0xF21 +#define WM5100_DSP1_CONTROL_28 0xF22 +#define WM5100_DSP1_CONTROL_29 0xF23 +#define WM5100_DSP1_CONTROL_30 0xF24 +#define WM5100_DSP2_CONTROL_1 0x1000 +#define WM5100_DSP2_CONTROL_2 0x1002 +#define WM5100_DSP2_CONTROL_3 0x1003 +#define WM5100_DSP2_CONTROL_4 0x1004 +#define WM5100_DSP2_CONTROL_5 0x1006 +#define WM5100_DSP2_CONTROL_6 0x1007 +#define WM5100_DSP2_CONTROL_7 0x1008 +#define WM5100_DSP2_CONTROL_8 0x1009 +#define WM5100_DSP2_CONTROL_9 0x100A +#define WM5100_DSP2_CONTROL_10 0x100B +#define WM5100_DSP2_CONTROL_11 0x100C +#define WM5100_DSP2_CONTROL_12 0x100D +#define WM5100_DSP2_CONTROL_13 0x100F +#define WM5100_DSP2_CONTROL_14 0x1010 +#define WM5100_DSP2_CONTROL_15 0x1011 +#define WM5100_DSP2_CONTROL_16 0x1012 +#define WM5100_DSP2_CONTROL_17 0x1013 +#define WM5100_DSP2_CONTROL_18 0x1014 +#define WM5100_DSP2_CONTROL_19 0x1016 +#define WM5100_DSP2_CONTROL_20 0x1017 +#define WM5100_DSP2_CONTROL_21 0x1018 +#define WM5100_DSP2_CONTROL_22 0x101A +#define WM5100_DSP2_CONTROL_23 0x101B +#define WM5100_DSP2_CONTROL_24 0x101C +#define WM5100_DSP2_CONTROL_25 0x101E +#define WM5100_DSP2_CONTROL_26 0x1020 +#define WM5100_DSP2_CONTROL_27 0x1021 +#define WM5100_DSP2_CONTROL_28 0x1022 +#define WM5100_DSP2_CONTROL_29 0x1023 +#define WM5100_DSP2_CONTROL_30 0x1024 +#define WM5100_DSP3_CONTROL_1 0x1100 +#define WM5100_DSP3_CONTROL_2 0x1102 +#define WM5100_DSP3_CONTROL_3 0x1103 +#define WM5100_DSP3_CONTROL_4 0x1104 +#define WM5100_DSP3_CONTROL_5 0x1106 +#define WM5100_DSP3_CONTROL_6 0x1107 +#define WM5100_DSP3_CONTROL_7 0x1108 +#define WM5100_DSP3_CONTROL_8 0x1109 +#define WM5100_DSP3_CONTROL_9 0x110A +#define WM5100_DSP3_CONTROL_10 0x110B +#define WM5100_DSP3_CONTROL_11 0x110C +#define WM5100_DSP3_CONTROL_12 0x110D +#define WM5100_DSP3_CONTROL_13 0x110F +#define WM5100_DSP3_CONTROL_14 0x1110 +#define WM5100_DSP3_CONTROL_15 0x1111 +#define WM5100_DSP3_CONTROL_16 0x1112 +#define WM5100_DSP3_CONTROL_17 0x1113 +#define WM5100_DSP3_CONTROL_18 0x1114 +#define WM5100_DSP3_CONTROL_19 0x1116 +#define WM5100_DSP3_CONTROL_20 0x1117 +#define WM5100_DSP3_CONTROL_21 0x1118 +#define WM5100_DSP3_CONTROL_22 0x111A +#define WM5100_DSP3_CONTROL_23 0x111B +#define WM5100_DSP3_CONTROL_24 0x111C +#define WM5100_DSP3_CONTROL_25 0x111E +#define WM5100_DSP3_CONTROL_26 0x1120 +#define WM5100_DSP3_CONTROL_27 0x1121 +#define WM5100_DSP3_CONTROL_28 0x1122 +#define WM5100_DSP3_CONTROL_29 0x1123 +#define WM5100_DSP3_CONTROL_30 0x1124 #define WM5100_DSP1_DM_0 0x4000 #define WM5100_DSP1_DM_1 0x4001 #define WM5100_DSP1_DM_2 0x4002 @@ -4561,6 +4651,75 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); #define WM5100_LHPF4_COEFF_WIDTH 16 /* LHPF4_COEFF - [15:0] */ /* + * R4132 (0x1024) - DSP2 Control 30 + */ +#define WM5100_DSP2_RATE_MASK 0xC000 /* DSP2_RATE - [15:14] */ +#define WM5100_DSP2_RATE_SHIFT 14 /* DSP2_RATE - [15:14] */ +#define WM5100_DSP2_RATE_WIDTH 2 /* DSP2_RATE - [15:14] */ +#define WM5100_DSP2_DBG_CLK_ENA 0x0008 /* DSP2_DBG_CLK_ENA */ +#define WM5100_DSP2_DBG_CLK_ENA_MASK 0x0008 /* DSP2_DBG_CLK_ENA */ +#define WM5100_DSP2_DBG_CLK_ENA_SHIFT 3 /* DSP2_DBG_CLK_ENA */ +#define WM5100_DSP2_DBG_CLK_ENA_WIDTH 1 /* DSP2_DBG_CLK_ENA */ +#define WM5100_DSP2_SYS_ENA 0x0004 /* DSP2_SYS_ENA */ +#define WM5100_DSP2_SYS_ENA_MASK 0x0004 /* DSP2_SYS_ENA */ +#define WM5100_DSP2_SYS_ENA_SHIFT 2 /* DSP2_SYS_ENA */ +#define WM5100_DSP2_SYS_ENA_WIDTH 1 /* DSP2_SYS_ENA */ +#define WM5100_DSP2_CORE_ENA 0x0002 /* DSP2_CORE_ENA */ +#define WM5100_DSP2_CORE_ENA_MASK 0x0002 /* DSP2_CORE_ENA */ +#define WM5100_DSP2_CORE_ENA_SHIFT 1 /* DSP2_CORE_ENA */ +#define WM5100_DSP2_CORE_ENA_WIDTH 1 /* DSP2_CORE_ENA */ +#define WM5100_DSP2_START 0x0001 /* DSP2_START */ +#define WM5100_DSP2_START_MASK 0x0001 /* DSP2_START */ +#define WM5100_DSP2_START_SHIFT 0 /* DSP2_START */ +#define WM5100_DSP2_START_WIDTH 1 /* DSP2_START */ + +/* + * R3876 (0xF24) - DSP1 Control 30 + */ +#define WM5100_DSP1_RATE_MASK 0xC000 /* DSP1_RATE - [15:14] */ +#define WM5100_DSP1_RATE_SHIFT 14 /* DSP1_RATE - [15:14] */ +#define WM5100_DSP1_RATE_WIDTH 2 /* DSP1_RATE - [15:14] */ +#define WM5100_DSP1_DBG_CLK_ENA 0x0008 /* DSP1_DBG_CLK_ENA */ +#define WM5100_DSP1_DBG_CLK_ENA_MASK 0x0008 /* DSP1_DBG_CLK_ENA */ +#define WM5100_DSP1_DBG_CLK_ENA_SHIFT 3 /* DSP1_DBG_CLK_ENA */ +#define WM5100_DSP1_DBG_CLK_ENA_WIDTH 1 /* DSP1_DBG_CLK_ENA */ +#define WM5100_DSP1_SYS_ENA 0x0004 /* DSP1_SYS_ENA */ +#define WM5100_DSP1_SYS_ENA_MASK 0x0004 /* DSP1_SYS_ENA */ +#define WM5100_DSP1_SYS_ENA_SHIFT 2 /* DSP1_SYS_ENA */ +#define WM5100_DSP1_SYS_ENA_WIDTH 1 /* DSP1_SYS_ENA */ +#define WM5100_DSP1_CORE_ENA 0x0002 /* DSP1_CORE_ENA */ +#define WM5100_DSP1_CORE_ENA_MASK 0x0002 /* DSP1_CORE_ENA */ +#define WM5100_DSP1_CORE_ENA_SHIFT 1 /* DSP1_CORE_ENA */ +#define WM5100_DSP1_CORE_ENA_WIDTH 1 /* DSP1_CORE_ENA */ +#define WM5100_DSP1_START 0x0001 /* DSP1_START */ +#define WM5100_DSP1_START_MASK 0x0001 /* DSP1_START */ +#define WM5100_DSP1_START_SHIFT 0 /* DSP1_START */ +#define WM5100_DSP1_START_WIDTH 1 /* DSP1_START */ + +/* + * R4388 (0x1124) - DSP3 Control 30 + */ +#define WM5100_DSP3_RATE_MASK 0xC000 /* DSP3_RATE - [15:14] */ +#define WM5100_DSP3_RATE_SHIFT 14 /* DSP3_RATE - [15:14] */ +#define WM5100_DSP3_RATE_WIDTH 2 /* DSP3_RATE - [15:14] */ +#define WM5100_DSP3_DBG_CLK_ENA 0x0008 /* DSP3_DBG_CLK_ENA */ +#define WM5100_DSP3_DBG_CLK_ENA_MASK 0x0008 /* DSP3_DBG_CLK_ENA */ +#define WM5100_DSP3_DBG_CLK_ENA_SHIFT 3 /* DSP3_DBG_CLK_ENA */ +#define WM5100_DSP3_DBG_CLK_ENA_WIDTH 1 /* DSP3_DBG_CLK_ENA */ +#define WM5100_DSP3_SYS_ENA 0x0004 /* DSP3_SYS_ENA */ +#define WM5100_DSP3_SYS_ENA_MASK 0x0004 /* DSP3_SYS_ENA */ +#define WM5100_DSP3_SYS_ENA_SHIFT 2 /* DSP3_SYS_ENA */ +#define WM5100_DSP3_SYS_ENA_WIDTH 1 /* DSP3_SYS_ENA */ +#define WM5100_DSP3_CORE_ENA 0x0002 /* DSP3_CORE_ENA */ +#define WM5100_DSP3_CORE_ENA_MASK 0x0002 /* DSP3_CORE_ENA */ +#define WM5100_DSP3_CORE_ENA_SHIFT 1 /* DSP3_CORE_ENA */ +#define WM5100_DSP3_CORE_ENA_WIDTH 1 /* DSP3_CORE_ENA */ +#define WM5100_DSP3_START 0x0001 /* DSP3_START */ +#define WM5100_DSP3_START_MASK 0x0001 /* DSP3_START */ +#define WM5100_DSP3_START_SHIFT 0 /* DSP3_START */ +#define WM5100_DSP3_START_WIDTH 1 /* DSP3_START */ + +/* * R16384 (0x4000) - DSP1 DM 0 */ #define WM5100_DSP1_DM_START_1_MASK 0x00FF /* DSP1_DM_START - [7:0] */ diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c new file mode 100644 index 000000000000..6537f16d383e --- /dev/null +++ b/sound/soc/codecs/wm5102.c @@ -0,0 +1,903 @@ +/* + * wm5102.c -- WM5102 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include <linux/mfd/arizona/core.h> +#include <linux/mfd/arizona/registers.h> + +#include "arizona.h" +#include "wm5102.h" + +struct wm5102_priv { + struct arizona_priv core; + struct arizona_fll fll[2]; +}; + +static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); + +static const struct snd_kcontrol_new wm5102_snd_controls[] = { +SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1R_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2R_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3R_CONTROL, + ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, + ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), +SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5, + ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA), + +ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE), + +SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode), +SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), +SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), +SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), + +ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv), + +ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT2L", ARIZONA_OUT2LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT2R", ARIZONA_OUT2RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUTL", ARIZONA_OUT4LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUTR", ARIZONA_OUT4RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), + +SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUT1_OSR_SHIFT, 1, 0), +SOC_SINGLE("OUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L, + ARIZONA_OUT2_OSR_SHIFT, 1, 0), +SOC_SINGLE("EPOUT High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3_OSR_SHIFT, 1, 0), +SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, + ARIZONA_OUT4_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, + ARIZONA_OUT5_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUTPUT_PATH_CONFIG_1R, + ARIZONA_OUT1L_PGA_VOL_SHIFT, + 0x34, 0x40, 0, ana_tlv), +SOC_DOUBLE_R_RANGE_TLV("OUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L, + ARIZONA_OUTPUT_PATH_CONFIG_2R, + ARIZONA_OUT2L_PGA_VOL_SHIFT, + 0x34, 0x40, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("EPOUT Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv), + +SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, + ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), + +ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), +}; + +ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT2L, ARIZONA_OUT2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT2R, ARIZONA_OUT2RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUTL, ARIZONA_OUT4LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUTR, ARIZONA_OUT4RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE); + +static const struct snd_soc_dapm_widget wm5102_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, + ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0), + +SND_SOC_DAPM_SIGGEN("TONE"), +SND_SOC_DAPM_SIGGEN("NOISE"), + +SND_SOC_DAPM_INPUT("IN1L"), +SND_SOC_DAPM_INPUT("IN1R"), +SND_SOC_DAPM_INPUT("IN2L"), +SND_SOC_DAPM_INPUT("IN2R"), +SND_SOC_DAPM_INPUT("IN3L"), +SND_SOC_DAPM_INPUT("IN3R"), + +SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1, + ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_PGA("ASRC1L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC1R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1R_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0, + ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0, + ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0, + ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, + ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT2R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"), +ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"), +ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"), +ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"), + +ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), +ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), +ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"), +ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"), + +ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), +ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), +ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"), +ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"), + +ARIZONA_MIXER_WIDGETS(Mic, "Mic"), +ARIZONA_MIXER_WIDGETS(Noise, "Noise"), + +ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"), +ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"), + +ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"), +ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"), +ARIZONA_MIXER_WIDGETS(OUT2L, "HPOUT2L"), +ARIZONA_MIXER_WIDGETS(OUT2R, "HPOUT2R"), +ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"), +ARIZONA_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"), +ARIZONA_MIXER_WIDGETS(SPKOUTR, "SPKOUTR"), +ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"), +ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"), + +ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"), +ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"), +ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"), +ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"), +ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"), +ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"), +ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"), +ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"), + +ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), +ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), + +ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), +ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), + +ARIZONA_MIXER_WIDGETS(ASRC1L, "ASRC1L"), +ARIZONA_MIXER_WIDGETS(ASRC1R, "ASRC1R"), +ARIZONA_MIXER_WIDGETS(ASRC2L, "ASRC2L"), +ARIZONA_MIXER_WIDGETS(ASRC2R, "ASRC2R"), + +SND_SOC_DAPM_OUTPUT("HPOUT1L"), +SND_SOC_DAPM_OUTPUT("HPOUT1R"), +SND_SOC_DAPM_OUTPUT("HPOUT2L"), +SND_SOC_DAPM_OUTPUT("HPOUT2R"), +SND_SOC_DAPM_OUTPUT("EPOUTN"), +SND_SOC_DAPM_OUTPUT("EPOUTP"), +SND_SOC_DAPM_OUTPUT("SPKOUTLN"), +SND_SOC_DAPM_OUTPUT("SPKOUTLP"), +SND_SOC_DAPM_OUTPUT("SPKOUTRN"), +SND_SOC_DAPM_OUTPUT("SPKOUTRP"), +SND_SOC_DAPM_OUTPUT("SPKDAT1L"), +SND_SOC_DAPM_OUTPUT("SPKDAT1R"), +}; + +#define ARIZONA_MIXER_INPUT_ROUTES(name) \ + { name, "Noise Generator", "Noise Generator" }, \ + { name, "Tone Generator 1", "Tone Generator 1" }, \ + { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "IN1L", "IN1L PGA" }, \ + { name, "IN1R", "IN1R PGA" }, \ + { name, "IN2L", "IN2L PGA" }, \ + { name, "IN2R", "IN2R PGA" }, \ + { name, "IN3L", "IN3L PGA" }, \ + { name, "IN3R", "IN3R PGA" }, \ + { name, "Mic Mute Mixer", "Mic Mute Mixer" }, \ + { name, "AIF1RX1", "AIF1RX1" }, \ + { name, "AIF1RX2", "AIF1RX2" }, \ + { name, "AIF1RX3", "AIF1RX3" }, \ + { name, "AIF1RX4", "AIF1RX4" }, \ + { name, "AIF1RX5", "AIF1RX5" }, \ + { name, "AIF1RX6", "AIF1RX6" }, \ + { name, "AIF1RX7", "AIF1RX7" }, \ + { name, "AIF1RX8", "AIF1RX8" }, \ + { name, "AIF2RX1", "AIF2RX1" }, \ + { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "AIF3RX1", "AIF3RX1" }, \ + { name, "AIF3RX2", "AIF3RX2" }, \ + { name, "EQ1", "EQ1" }, \ + { name, "EQ2", "EQ2" }, \ + { name, "EQ3", "EQ3" }, \ + { name, "EQ4", "EQ4" }, \ + { name, "DRC1L", "DRC1L" }, \ + { name, "DRC1R", "DRC1R" }, \ + { name, "DRC2L", "DRC2L" }, \ + { name, "DRC2R", "DRC2R" }, \ + { name, "LHPF1", "LHPF1" }, \ + { name, "LHPF2", "LHPF2" }, \ + { name, "LHPF3", "LHPF3" }, \ + { name, "LHPF4", "LHPF4" }, \ + { name, "ASRC1L", "ASRC1L" }, \ + { name, "ASRC1R", "ASRC1R" }, \ + { name, "ASRC2L", "ASRC2L" }, \ + { name, "ASRC2R", "ASRC2R" } + +static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { + { "AIF2 Capture", NULL, "DBVDD2" }, + { "AIF2 Playback", NULL, "DBVDD2" }, + + { "AIF3 Capture", NULL, "DBVDD3" }, + { "AIF3 Playback", NULL, "DBVDD3" }, + + { "OUT1L", NULL, "CPVDD" }, + { "OUT1R", NULL, "CPVDD" }, + { "OUT2L", NULL, "CPVDD" }, + { "OUT2R", NULL, "CPVDD" }, + { "OUT3L", NULL, "CPVDD" }, + + { "OUT4L", NULL, "SPKVDDL" }, + { "OUT4R", NULL, "SPKVDDR" }, + + { "OUT1L", NULL, "SYSCLK" }, + { "OUT1R", NULL, "SYSCLK" }, + { "OUT2L", NULL, "SYSCLK" }, + { "OUT2R", NULL, "SYSCLK" }, + { "OUT3L", NULL, "SYSCLK" }, + { "OUT4L", NULL, "SYSCLK" }, + { "OUT4R", NULL, "SYSCLK" }, + { "OUT5L", NULL, "SYSCLK" }, + { "OUT5R", NULL, "SYSCLK" }, + + { "MICBIAS1", NULL, "MICVDD" }, + { "MICBIAS2", NULL, "MICVDD" }, + { "MICBIAS3", NULL, "MICVDD" }, + + { "Noise Generator", NULL, "NOISE" }, + { "Tone Generator 1", NULL, "TONE" }, + { "Tone Generator 2", NULL, "TONE" }, + + { "Mic Mute Mixer", NULL, "Noise Mixer" }, + { "Mic Mute Mixer", NULL, "Mic Mixer" }, + + { "AIF1 Capture", NULL, "AIF1TX1" }, + { "AIF1 Capture", NULL, "AIF1TX2" }, + { "AIF1 Capture", NULL, "AIF1TX3" }, + { "AIF1 Capture", NULL, "AIF1TX4" }, + { "AIF1 Capture", NULL, "AIF1TX5" }, + { "AIF1 Capture", NULL, "AIF1TX6" }, + { "AIF1 Capture", NULL, "AIF1TX7" }, + { "AIF1 Capture", NULL, "AIF1TX8" }, + + { "AIF1RX1", NULL, "AIF1 Playback" }, + { "AIF1RX2", NULL, "AIF1 Playback" }, + { "AIF1RX3", NULL, "AIF1 Playback" }, + { "AIF1RX4", NULL, "AIF1 Playback" }, + { "AIF1RX5", NULL, "AIF1 Playback" }, + { "AIF1RX6", NULL, "AIF1 Playback" }, + { "AIF1RX7", NULL, "AIF1 Playback" }, + { "AIF1RX8", NULL, "AIF1 Playback" }, + + { "AIF2 Capture", NULL, "AIF2TX1" }, + { "AIF2 Capture", NULL, "AIF2TX2" }, + + { "AIF2RX1", NULL, "AIF2 Playback" }, + { "AIF2RX2", NULL, "AIF2 Playback" }, + + { "AIF3 Capture", NULL, "AIF3TX1" }, + { "AIF3 Capture", NULL, "AIF3TX2" }, + + { "AIF3RX1", NULL, "AIF3 Playback" }, + { "AIF3RX2", NULL, "AIF3 Playback" }, + + { "AIF1 Playback", NULL, "SYSCLK" }, + { "AIF2 Playback", NULL, "SYSCLK" }, + { "AIF3 Playback", NULL, "SYSCLK" }, + + { "AIF1 Capture", NULL, "SYSCLK" }, + { "AIF2 Capture", NULL, "SYSCLK" }, + { "AIF3 Capture", NULL, "SYSCLK" }, + + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), + ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), + ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), + ARIZONA_MIXER_ROUTES("OUT2R", "HPOUT2R"), + ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"), + + ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUTL"), + ARIZONA_MIXER_ROUTES("OUT4R", "SPKOUTR"), + ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"), + ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"), + + ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"), + ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"), + + ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"), + ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"), + ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"), + ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"), + ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"), + ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"), + ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"), + ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"), + + ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), + ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + + ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"), + ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"), + + ARIZONA_MIXER_ROUTES("EQ1", "EQ1"), + ARIZONA_MIXER_ROUTES("EQ2", "EQ2"), + ARIZONA_MIXER_ROUTES("EQ3", "EQ3"), + ARIZONA_MIXER_ROUTES("EQ4", "EQ4"), + + ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), + ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), + ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"), + ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"), + + ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), + ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), + ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), + ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + + ARIZONA_MIXER_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MIXER_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MIXER_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MIXER_ROUTES("ASRC2R", "ASRC2R"), + + { "HPOUT1L", NULL, "OUT1L" }, + { "HPOUT1R", NULL, "OUT1R" }, + + { "HPOUT2L", NULL, "OUT2L" }, + { "HPOUT2R", NULL, "OUT2R" }, + + { "EPOUTN", NULL, "OUT3L" }, + { "EPOUTP", NULL, "OUT3L" }, + + { "SPKOUTLN", NULL, "OUT4L" }, + { "SPKOUTLP", NULL, "OUT4L" }, + + { "SPKOUTRN", NULL, "OUT4R" }, + { "SPKOUTRP", NULL, "OUT4R" }, + + { "SPKDAT1L", NULL, "OUT5L" }, + { "SPKDAT1R", NULL, "OUT5R" }, +}; + +static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct wm5102_priv *wm5102 = snd_soc_codec_get_drvdata(codec); + + switch (fll_id) { + case WM5102_FLL1: + return arizona_set_fll(&wm5102->fll[0], source, Fref, Fout); + case WM5102_FLL2: + return arizona_set_fll(&wm5102->fll[1], source, Fref, Fout); + default: + return -EINVAL; + } +} + +#define WM5102_RATES SNDRV_PCM_RATE_8000_192000 + +#define WM5102_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver wm5102_dai[] = { + { + .name = "wm5102-aif1", + .id = 1, + .base = ARIZONA_AIF1_BCLK_CTRL, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm5102-aif2", + .id = 2, + .base = ARIZONA_AIF2_BCLK_CTRL, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm5102-aif3", + .id = 3, + .base = ARIZONA_AIF3_BCLK_CTRL, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, +}; + +static int wm5102_codec_probe(struct snd_soc_codec *codec) +{ + struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); + + codec->control_data = priv->core.arizona->regmap; + return snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); +} + +#define WM5102_DIG_VU 0x0200 + +static unsigned int wm5102_digital_vu[] = { + ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_ADC_DIGITAL_VOLUME_3R, + + ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, + ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, + ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_DAC_DIGITAL_VOLUME_3R, + ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, + ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm5102 = { + .probe = wm5102_codec_probe, + + .idle_bias_off = true, + + .set_sysclk = arizona_set_sysclk, + .set_pll = wm5102_set_fll, + + .controls = wm5102_snd_controls, + .num_controls = ARRAY_SIZE(wm5102_snd_controls), + .dapm_widgets = wm5102_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm5102_dapm_widgets), + .dapm_routes = wm5102_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm5102_dapm_routes), +}; + +static int __devinit wm5102_probe(struct platform_device *pdev) +{ + struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); + struct wm5102_priv *wm5102; + int i; + + wm5102 = devm_kzalloc(&pdev->dev, sizeof(struct wm5102_priv), + GFP_KERNEL); + if (wm5102 == NULL) + return -ENOMEM; + platform_set_drvdata(pdev, wm5102); + + wm5102->core.arizona = arizona; + + for (i = 0; i < ARRAY_SIZE(wm5102->fll); i++) + wm5102->fll[i].vco_mult = 1; + + arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1, + ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK, + &wm5102->fll[0]); + arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1, + ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK, + &wm5102->fll[1]); + + for (i = 0; i < ARRAY_SIZE(wm5102_dai); i++) + arizona_init_dai(&wm5102->core, i); + + /* Latch volume update bits */ + for (i = 0; i < ARRAY_SIZE(wm5102_digital_vu); i++) + regmap_update_bits(arizona->regmap, wm5102_digital_vu[i], + WM5102_DIG_VU, WM5102_DIG_VU); + + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5102, + wm5102_dai, ARRAY_SIZE(wm5102_dai)); +} + +static int __devexit wm5102_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + pm_runtime_disable(&pdev->dev); + + return 0; +} + +static struct platform_driver wm5102_codec_driver = { + .driver = { + .name = "wm5102-codec", + .owner = THIS_MODULE, + }, + .probe = wm5102_probe, + .remove = __devexit_p(wm5102_remove), +}; + +module_platform_driver(wm5102_codec_driver); + +MODULE_DESCRIPTION("ASoC WM5102 driver"); +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wm5102-codec"); diff --git a/sound/soc/codecs/wm5102.h b/sound/soc/codecs/wm5102.h new file mode 100644 index 000000000000..d30477f3070c --- /dev/null +++ b/sound/soc/codecs/wm5102.h @@ -0,0 +1,21 @@ +/* + * wm5102.h -- WM5102 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM5102_H +#define _WM5102_H + +#include "arizona.h" + +#define WM5102_FLL1 1 +#define WM5102_FLL2 2 + +#endif diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c new file mode 100644 index 000000000000..8033f7065189 --- /dev/null +++ b/sound/soc/codecs/wm5110.c @@ -0,0 +1,950 @@ +/* + * wm5110.c -- WM5110 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include <linux/mfd/arizona/core.h> +#include <linux/mfd/arizona/registers.h> + +#include "arizona.h" +#include "wm5110.h" + +struct wm5110_priv { + struct arizona_priv core; + struct arizona_fll fll[2]; +}; + +static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); + +static const struct snd_kcontrol_new wm5110_snd_controls[] = { +SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN4 High Performance Switch", ARIZONA_IN4L_CONTROL, + ARIZONA_IN4_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1R_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2R_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3R_CONTROL, + ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN4 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4L, + ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN4 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4L, + ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, + ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), +SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5, + ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA), + +ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE), + +SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode), +SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), +SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), +SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), + +ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv), + +ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT2L", ARIZONA_OUT2LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT2R", ARIZONA_OUT2RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUTL", ARIZONA_OUT4LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUTR", ARIZONA_OUT4RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT2L", ARIZONA_OUT6LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE), + +SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUT1_OSR_SHIFT, 1, 0), +SOC_SINGLE("OUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L, + ARIZONA_OUT2_OSR_SHIFT, 1, 0), +SOC_SINGLE("EPOUT High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3_OSR_SHIFT, 1, 0), +SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, + ARIZONA_OUT4_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, + ARIZONA_OUT5_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L, + ARIZONA_OUT6_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_6L, + ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_6L, + ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUTPUT_PATH_CONFIG_1R, + ARIZONA_OUT1L_PGA_VOL_SHIFT, + 0x34, 0x40, 0, ana_tlv), +SOC_DOUBLE_R_RANGE_TLV("OUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L, + ARIZONA_OUTPUT_PATH_CONFIG_2R, + ARIZONA_OUT2L_PGA_VOL_SHIFT, + 0x34, 0x40, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("EPOUT Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv), + +SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, + ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), +SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT, + ARIZONA_SPK2R_MUTE_SHIFT, 1, 1), + +ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), +}; + +ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT2L, ARIZONA_OUT2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT2R, ARIZONA_OUT2RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUTL, ARIZONA_OUT4LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUTR, ARIZONA_OUT4RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT2L, ARIZONA_OUT6LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT2R, ARIZONA_OUT6RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE); + +static const struct snd_soc_dapm_widget wm5110_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, + ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0), + +SND_SOC_DAPM_SIGGEN("TONE"), +SND_SOC_DAPM_SIGGEN("NOISE"), + +SND_SOC_DAPM_INPUT("IN1L"), +SND_SOC_DAPM_INPUT("IN1R"), +SND_SOC_DAPM_INPUT("IN2L"), +SND_SOC_DAPM_INPUT("IN2R"), +SND_SOC_DAPM_INPUT("IN3L"), +SND_SOC_DAPM_INPUT("IN3R"), +SND_SOC_DAPM_INPUT("IN4L"), +SND_SOC_DAPM_INPUT("IN4R"), + +SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN4L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN4R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1, + ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_PGA("ASRC1L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC1R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1R_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0, + ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0, + ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0, + ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, + ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT2R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT6L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT6L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT6R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT6R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"), +ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"), +ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"), +ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"), + +ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), +ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), +ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"), +ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"), + +ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), +ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), +ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"), +ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"), + +ARIZONA_MIXER_WIDGETS(Mic, "Mic"), +ARIZONA_MIXER_WIDGETS(Noise, "Noise"), + +ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"), +ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"), + +ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"), +ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"), +ARIZONA_MIXER_WIDGETS(OUT2L, "HPOUT2L"), +ARIZONA_MIXER_WIDGETS(OUT2R, "HPOUT2R"), +ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"), +ARIZONA_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"), +ARIZONA_MIXER_WIDGETS(SPKOUTR, "SPKOUTR"), +ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"), +ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"), +ARIZONA_MIXER_WIDGETS(SPKDAT2L, "SPKDAT2L"), +ARIZONA_MIXER_WIDGETS(SPKDAT2R, "SPKDAT2R"), + +ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"), +ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"), +ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"), +ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"), +ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"), +ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"), +ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"), +ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"), + +ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), +ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), + +ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), +ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), + +ARIZONA_MIXER_WIDGETS(ASRC1L, "ASRC1L"), +ARIZONA_MIXER_WIDGETS(ASRC1R, "ASRC1R"), +ARIZONA_MIXER_WIDGETS(ASRC2L, "ASRC2L"), +ARIZONA_MIXER_WIDGETS(ASRC2R, "ASRC2R"), + +SND_SOC_DAPM_OUTPUT("HPOUT1L"), +SND_SOC_DAPM_OUTPUT("HPOUT1R"), +SND_SOC_DAPM_OUTPUT("HPOUT2L"), +SND_SOC_DAPM_OUTPUT("HPOUT2R"), +SND_SOC_DAPM_OUTPUT("EPOUTN"), +SND_SOC_DAPM_OUTPUT("EPOUTP"), +SND_SOC_DAPM_OUTPUT("SPKOUTLN"), +SND_SOC_DAPM_OUTPUT("SPKOUTLP"), +SND_SOC_DAPM_OUTPUT("SPKOUTRN"), +SND_SOC_DAPM_OUTPUT("SPKOUTRP"), +SND_SOC_DAPM_OUTPUT("SPKDAT1L"), +SND_SOC_DAPM_OUTPUT("SPKDAT1R"), +SND_SOC_DAPM_OUTPUT("SPKDAT2L"), +SND_SOC_DAPM_OUTPUT("SPKDAT2R"), +}; + +#define ARIZONA_MIXER_INPUT_ROUTES(name) \ + { name, "Noise Generator", "Noise Generator" }, \ + { name, "Tone Generator 1", "Tone Generator 1" }, \ + { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "IN1L", "IN1L PGA" }, \ + { name, "IN1R", "IN1R PGA" }, \ + { name, "IN2L", "IN2L PGA" }, \ + { name, "IN2R", "IN2R PGA" }, \ + { name, "IN3L", "IN3L PGA" }, \ + { name, "IN3R", "IN3R PGA" }, \ + { name, "IN4L", "IN4L PGA" }, \ + { name, "IN4R", "IN4R PGA" }, \ + { name, "Mic Mute Mixer", "Mic Mute Mixer" }, \ + { name, "AIF1RX1", "AIF1RX1" }, \ + { name, "AIF1RX2", "AIF1RX2" }, \ + { name, "AIF1RX3", "AIF1RX3" }, \ + { name, "AIF1RX4", "AIF1RX4" }, \ + { name, "AIF1RX5", "AIF1RX5" }, \ + { name, "AIF1RX6", "AIF1RX6" }, \ + { name, "AIF1RX7", "AIF1RX7" }, \ + { name, "AIF1RX8", "AIF1RX8" }, \ + { name, "AIF2RX1", "AIF2RX1" }, \ + { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "AIF3RX1", "AIF3RX1" }, \ + { name, "AIF3RX2", "AIF3RX2" }, \ + { name, "EQ1", "EQ1" }, \ + { name, "EQ2", "EQ2" }, \ + { name, "EQ3", "EQ3" }, \ + { name, "EQ4", "EQ4" }, \ + { name, "DRC1L", "DRC1L" }, \ + { name, "DRC1R", "DRC1R" }, \ + { name, "DRC2L", "DRC2L" }, \ + { name, "DRC2R", "DRC2R" }, \ + { name, "LHPF1", "LHPF1" }, \ + { name, "LHPF2", "LHPF2" }, \ + { name, "LHPF3", "LHPF3" }, \ + { name, "LHPF4", "LHPF4" }, \ + { name, "ASRC1L", "ASRC1L" }, \ + { name, "ASRC1R", "ASRC1R" }, \ + { name, "ASRC2L", "ASRC2L" }, \ + { name, "ASRC2R", "ASRC2R" } + +static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { + { "AIF2 Capture", NULL, "DBVDD2" }, + { "AIF2 Playback", NULL, "DBVDD2" }, + + { "AIF3 Capture", NULL, "DBVDD3" }, + { "AIF3 Playback", NULL, "DBVDD3" }, + + { "OUT1L", NULL, "CPVDD" }, + { "OUT1R", NULL, "CPVDD" }, + { "OUT2L", NULL, "CPVDD" }, + { "OUT2R", NULL, "CPVDD" }, + { "OUT3L", NULL, "CPVDD" }, + + { "OUT4L", NULL, "SPKVDDL" }, + { "OUT4R", NULL, "SPKVDDR" }, + + { "OUT1L", NULL, "SYSCLK" }, + { "OUT1R", NULL, "SYSCLK" }, + { "OUT2L", NULL, "SYSCLK" }, + { "OUT2R", NULL, "SYSCLK" }, + { "OUT3L", NULL, "SYSCLK" }, + { "OUT4L", NULL, "SYSCLK" }, + { "OUT4R", NULL, "SYSCLK" }, + { "OUT5L", NULL, "SYSCLK" }, + { "OUT5R", NULL, "SYSCLK" }, + { "OUT6L", NULL, "SYSCLK" }, + { "OUT6R", NULL, "SYSCLK" }, + + { "MICBIAS1", NULL, "MICVDD" }, + { "MICBIAS2", NULL, "MICVDD" }, + { "MICBIAS3", NULL, "MICVDD" }, + + { "Noise Generator", NULL, "NOISE" }, + { "Tone Generator 1", NULL, "TONE" }, + { "Tone Generator 2", NULL, "TONE" }, + + { "Mic Mute Mixer", NULL, "Noise Mixer" }, + { "Mic Mute Mixer", NULL, "Mic Mixer" }, + + { "AIF1 Capture", NULL, "AIF1TX1" }, + { "AIF1 Capture", NULL, "AIF1TX2" }, + { "AIF1 Capture", NULL, "AIF1TX3" }, + { "AIF1 Capture", NULL, "AIF1TX4" }, + { "AIF1 Capture", NULL, "AIF1TX5" }, + { "AIF1 Capture", NULL, "AIF1TX6" }, + { "AIF1 Capture", NULL, "AIF1TX7" }, + { "AIF1 Capture", NULL, "AIF1TX8" }, + + { "AIF1RX1", NULL, "AIF1 Playback" }, + { "AIF1RX2", NULL, "AIF1 Playback" }, + { "AIF1RX3", NULL, "AIF1 Playback" }, + { "AIF1RX4", NULL, "AIF1 Playback" }, + { "AIF1RX5", NULL, "AIF1 Playback" }, + { "AIF1RX6", NULL, "AIF1 Playback" }, + { "AIF1RX7", NULL, "AIF1 Playback" }, + { "AIF1RX8", NULL, "AIF1 Playback" }, + + { "AIF2 Capture", NULL, "AIF2TX1" }, + { "AIF2 Capture", NULL, "AIF2TX2" }, + + { "AIF2RX1", NULL, "AIF2 Playback" }, + { "AIF2RX2", NULL, "AIF2 Playback" }, + + { "AIF3 Capture", NULL, "AIF3TX1" }, + { "AIF3 Capture", NULL, "AIF3TX2" }, + + { "AIF3RX1", NULL, "AIF3 Playback" }, + { "AIF3RX2", NULL, "AIF3 Playback" }, + + { "AIF1 Playback", NULL, "SYSCLK" }, + { "AIF2 Playback", NULL, "SYSCLK" }, + { "AIF3 Playback", NULL, "SYSCLK" }, + + { "AIF1 Capture", NULL, "SYSCLK" }, + { "AIF2 Capture", NULL, "SYSCLK" }, + { "AIF3 Capture", NULL, "SYSCLK" }, + + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), + ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), + ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), + ARIZONA_MIXER_ROUTES("OUT2R", "HPOUT2R"), + ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"), + + ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUTL"), + ARIZONA_MIXER_ROUTES("OUT4R", "SPKOUTR"), + ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"), + ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"), + ARIZONA_MIXER_ROUTES("OUT6L", "SPKDAT2L"), + ARIZONA_MIXER_ROUTES("OUT6R", "SPKDAT2R"), + + ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"), + ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"), + + ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"), + ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"), + ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"), + ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"), + ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"), + ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"), + ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"), + ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"), + + ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), + ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + + ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"), + ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"), + + ARIZONA_MIXER_ROUTES("EQ1", "EQ1"), + ARIZONA_MIXER_ROUTES("EQ2", "EQ2"), + ARIZONA_MIXER_ROUTES("EQ3", "EQ3"), + ARIZONA_MIXER_ROUTES("EQ4", "EQ4"), + + ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), + ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), + ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"), + ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"), + + ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), + ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), + ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), + ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + + ARIZONA_MIXER_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MIXER_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MIXER_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MIXER_ROUTES("ASRC2R", "ASRC2R"), + + { "HPOUT1L", NULL, "OUT1L" }, + { "HPOUT1R", NULL, "OUT1R" }, + + { "HPOUT2L", NULL, "OUT2L" }, + { "HPOUT2R", NULL, "OUT2R" }, + + { "EPOUTN", NULL, "OUT3L" }, + { "EPOUTP", NULL, "OUT3L" }, + + { "SPKOUTLN", NULL, "OUT4L" }, + { "SPKOUTLP", NULL, "OUT4L" }, + + { "SPKOUTRN", NULL, "OUT4R" }, + { "SPKOUTRP", NULL, "OUT4R" }, + + { "SPKDAT1L", NULL, "OUT5L" }, + { "SPKDAT1R", NULL, "OUT5R" }, + + { "SPKDAT2L", NULL, "OUT6L" }, + { "SPKDAT2R", NULL, "OUT6R" }, +}; + +static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct wm5110_priv *wm5110 = snd_soc_codec_get_drvdata(codec); + + switch (fll_id) { + case WM5110_FLL1: + return arizona_set_fll(&wm5110->fll[0], source, Fref, Fout); + case WM5110_FLL2: + return arizona_set_fll(&wm5110->fll[1], source, Fref, Fout); + default: + return -EINVAL; + } +} + +#define WM5110_RATES SNDRV_PCM_RATE_8000_192000 + +#define WM5110_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver wm5110_dai[] = { + { + .name = "wm5110-aif1", + .id = 1, + .base = ARIZONA_AIF1_BCLK_CTRL, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm5110-aif2", + .id = 2, + .base = ARIZONA_AIF2_BCLK_CTRL, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm5110-aif3", + .id = 3, + .base = ARIZONA_AIF3_BCLK_CTRL, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, +}; + +static int wm5110_codec_probe(struct snd_soc_codec *codec) +{ + struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); + + codec->control_data = priv->core.arizona->regmap; + return snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); +} + +#define WM5110_DIG_VU 0x0200 + +static unsigned int wm5110_digital_vu[] = { + ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_ADC_DIGITAL_VOLUME_3R, + + ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, + ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, + ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_DAC_DIGITAL_VOLUME_3R, + ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, + ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm5110 = { + .probe = wm5110_codec_probe, + + .idle_bias_off = true, + + .set_sysclk = arizona_set_sysclk, + .set_pll = wm5110_set_fll, + + .controls = wm5110_snd_controls, + .num_controls = ARRAY_SIZE(wm5110_snd_controls), + .dapm_widgets = wm5110_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm5110_dapm_widgets), + .dapm_routes = wm5110_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm5110_dapm_routes), +}; + +static int __devinit wm5110_probe(struct platform_device *pdev) +{ + struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); + struct wm5110_priv *wm5110; + int i; + + wm5110 = devm_kzalloc(&pdev->dev, sizeof(struct wm5110_priv), + GFP_KERNEL); + if (wm5110 == NULL) + return -ENOMEM; + platform_set_drvdata(pdev, wm5110); + + wm5110->core.arizona = arizona; + + for (i = 0; i < ARRAY_SIZE(wm5110->fll); i++) + wm5110->fll[i].vco_mult = 3; + + arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1, + ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK, + &wm5110->fll[0]); + arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1, + ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK, + &wm5110->fll[1]); + + for (i = 0; i < ARRAY_SIZE(wm5110_dai); i++) + arizona_init_dai(&wm5110->core, i); + + /* Latch volume update bits */ + for (i = 0; i < ARRAY_SIZE(wm5110_digital_vu); i++) + regmap_update_bits(arizona->regmap, wm5110_digital_vu[i], + WM5110_DIG_VU, WM5110_DIG_VU); + + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5110, + wm5110_dai, ARRAY_SIZE(wm5110_dai)); +} + +static int __devexit wm5110_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + pm_runtime_disable(&pdev->dev); + + return 0; +} + +static struct platform_driver wm5110_codec_driver = { + .driver = { + .name = "wm5110-codec", + .owner = THIS_MODULE, + }, + .probe = wm5110_probe, + .remove = __devexit_p(wm5110_remove), +}; + +module_platform_driver(wm5110_codec_driver); + +MODULE_DESCRIPTION("ASoC WM5110 driver"); +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wm5110-codec"); diff --git a/sound/soc/codecs/wm5110.h b/sound/soc/codecs/wm5110.h new file mode 100644 index 000000000000..75e9351ccab0 --- /dev/null +++ b/sound/soc/codecs/wm5110.h @@ -0,0 +1,21 @@ +/* + * wm5110.h -- WM5110 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM5110_H +#define _WM5110_H + +#include "arizona.h" + +#define WM5110_FLL1 1 +#define WM5110_FLL2 2 + +#endif diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index aa12c6b6beeb..d26c8ae4e6d9 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1,7 +1,7 @@ /* * wm8350.c -- WM8350 ALSA SoC audio driver * - * Copyright (C) 2007, 2008 Wolfson Microelectronics PLC. + * Copyright (C) 2007-12 Wolfson Microelectronics PLC. * * Author: Liam Girdwood <lrg@slimlogic.co.uk> * @@ -71,27 +71,6 @@ struct wm8350_data { int fll_freq_in; }; -static unsigned int wm8350_codec_cache_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct wm8350 *wm8350 = codec->control_data; - return wm8350->reg_cache[reg]; -} - -static unsigned int wm8350_codec_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct wm8350 *wm8350 = codec->control_data; - return wm8350_reg_read(wm8350, reg); -} - -static int wm8350_codec_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - struct wm8350 *wm8350 = codec->control_data; - return wm8350_reg_write(wm8350, reg, value); -} - /* * Ramp OUT1 PGA volume to minimise pops at stream startup and shutdown. */ @@ -99,7 +78,7 @@ static inline int wm8350_out1_ramp_step(struct snd_soc_codec *codec) { struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); struct wm8350_output *out1 = &wm8350_data->out1; - struct wm8350 *wm8350 = codec->control_data; + struct wm8350 *wm8350 = wm8350_data->wm8350; int left_complete = 0, right_complete = 0; u16 reg, val; @@ -165,7 +144,7 @@ static inline int wm8350_out2_ramp_step(struct snd_soc_codec *codec) { struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); struct wm8350_output *out2 = &wm8350_data->out2; - struct wm8350 *wm8350 = codec->control_data; + struct wm8350 *wm8350 = wm8350_data->wm8350; int left_complete = 0, right_complete = 0; u16 reg, val; @@ -360,8 +339,8 @@ static int wm8350_put_volsw_2r_vu(struct snd_kcontrol *kcontrol, return ret; /* now hit the volume update bits (always bit 8) */ - val = wm8350_codec_read(codec, reg); - wm8350_codec_write(codec, reg, val | WM8350_OUT1_VU); + val = snd_soc_read(codec, reg); + snd_soc_write(codec, reg, val | WM8350_OUT1_VU); return 1; } @@ -781,7 +760,8 @@ static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; - struct wm8350 *wm8350 = codec->control_data; + struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); + struct wm8350 *wm8350 = wm8350_data->wm8350; u16 fll_4; switch (clk_id) { @@ -795,9 +775,9 @@ static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai, case WM8350_MCLK_SEL_PLL_32K: wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_1, WM8350_MCLK_SEL); - fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) & + fll_4 = snd_soc_read(codec, WM8350_FLL_CONTROL_4) & ~WM8350_FLL_CLK_SRC_MASK; - wm8350_codec_write(codec, WM8350_FLL_CONTROL_4, fll_4 | clk_id); + snd_soc_write(codec, WM8350_FLL_CONTROL_4, fll_4 | clk_id); break; } @@ -819,39 +799,39 @@ static int wm8350_set_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) switch (div_id) { case WM8350_ADC_CLKDIV: - val = wm8350_codec_read(codec, WM8350_ADC_DIVIDER) & + val = snd_soc_read(codec, WM8350_ADC_DIVIDER) & ~WM8350_ADC_CLKDIV_MASK; - wm8350_codec_write(codec, WM8350_ADC_DIVIDER, val | div); + snd_soc_write(codec, WM8350_ADC_DIVIDER, val | div); break; case WM8350_DAC_CLKDIV: - val = wm8350_codec_read(codec, WM8350_DAC_CLOCK_CONTROL) & + val = snd_soc_read(codec, WM8350_DAC_CLOCK_CONTROL) & ~WM8350_DAC_CLKDIV_MASK; - wm8350_codec_write(codec, WM8350_DAC_CLOCK_CONTROL, val | div); + snd_soc_write(codec, WM8350_DAC_CLOCK_CONTROL, val | div); break; case WM8350_BCLK_CLKDIV: - val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) & + val = snd_soc_read(codec, WM8350_CLOCK_CONTROL_1) & ~WM8350_BCLK_DIV_MASK; - wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div); + snd_soc_write(codec, WM8350_CLOCK_CONTROL_1, val | div); break; case WM8350_OPCLK_CLKDIV: - val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) & + val = snd_soc_read(codec, WM8350_CLOCK_CONTROL_1) & ~WM8350_OPCLK_DIV_MASK; - wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div); + snd_soc_write(codec, WM8350_CLOCK_CONTROL_1, val | div); break; case WM8350_SYS_CLKDIV: - val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) & + val = snd_soc_read(codec, WM8350_CLOCK_CONTROL_1) & ~WM8350_MCLK_DIV_MASK; - wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div); + snd_soc_write(codec, WM8350_CLOCK_CONTROL_1, val | div); break; case WM8350_DACLR_CLKDIV: - val = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) & + val = snd_soc_read(codec, WM8350_DAC_LR_RATE) & ~WM8350_DACLRC_RATE_MASK; - wm8350_codec_write(codec, WM8350_DAC_LR_RATE, val | div); + snd_soc_write(codec, WM8350_DAC_LR_RATE, val | div); break; case WM8350_ADCLR_CLKDIV: - val = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) & + val = snd_soc_read(codec, WM8350_ADC_LR_RATE) & ~WM8350_ADCLRC_RATE_MASK; - wm8350_codec_write(codec, WM8350_ADC_LR_RATE, val | div); + snd_soc_write(codec, WM8350_ADC_LR_RATE, val | div); break; default: return -EINVAL; @@ -863,13 +843,13 @@ static int wm8350_set_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) & + u16 iface = snd_soc_read(codec, WM8350_AI_FORMATING) & ~(WM8350_AIF_BCLK_INV | WM8350_AIF_LRCLK_INV | WM8350_AIF_FMT_MASK); - u16 master = wm8350_codec_read(codec, WM8350_AI_DAC_CONTROL) & + u16 master = snd_soc_read(codec, WM8350_AI_DAC_CONTROL) & ~WM8350_BCLK_MSTR; - u16 dac_lrc = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) & + u16 dac_lrc = snd_soc_read(codec, WM8350_DAC_LR_RATE) & ~WM8350_DACLRC_ENA; - u16 adc_lrc = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) & + u16 adc_lrc = snd_soc_read(codec, WM8350_ADC_LR_RATE) & ~WM8350_ADCLRC_ENA; /* set master/slave audio interface */ @@ -922,42 +902,10 @@ static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return -EINVAL; } - wm8350_codec_write(codec, WM8350_AI_FORMATING, iface); - wm8350_codec_write(codec, WM8350_AI_DAC_CONTROL, master); - wm8350_codec_write(codec, WM8350_DAC_LR_RATE, dac_lrc); - wm8350_codec_write(codec, WM8350_ADC_LR_RATE, adc_lrc); - return 0; -} - -static int wm8350_pcm_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *codec_dai) -{ - struct snd_soc_codec *codec = codec_dai->codec; - int master = wm8350_codec_cache_read(codec, WM8350_AI_DAC_CONTROL) & - WM8350_BCLK_MSTR; - int enabled = 0; - - /* Check that the DACs or ADCs are enabled since they are - * required for LRC in master mode. The DACs or ADCs need a - * valid audio path i.e. pin -> ADC or DAC -> pin before - * the LRC will be enabled in master mode. */ - if (!master || cmd != SNDRV_PCM_TRIGGER_START) - return 0; - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { - enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) & - (WM8350_ADCR_ENA | WM8350_ADCL_ENA); - } else { - enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) & - (WM8350_DACR_ENA | WM8350_DACL_ENA); - } - - if (!enabled) { - dev_err(codec->dev, - "%s: invalid audio path - no clocks available\n", - __func__); - return -EINVAL; - } + snd_soc_write(codec, WM8350_AI_FORMATING, iface); + snd_soc_write(codec, WM8350_AI_DAC_CONTROL, master); + snd_soc_write(codec, WM8350_DAC_LR_RATE, dac_lrc); + snd_soc_write(codec, WM8350_ADC_LR_RATE, adc_lrc); return 0; } @@ -966,8 +914,9 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai) { struct snd_soc_codec *codec = codec_dai->codec; - struct wm8350 *wm8350 = codec->control_data; - u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) & + struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); + struct wm8350 *wm8350 = wm8350_data->wm8350; + u16 iface = snd_soc_read(codec, WM8350_AI_FORMATING) & ~WM8350_AIF_WL_MASK; /* bit size */ @@ -985,7 +934,7 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream, break; } - wm8350_codec_write(codec, WM8350_AI_FORMATING, iface); + snd_soc_write(codec, WM8350_AI_FORMATING, iface); /* The sloping stopband filter is recommended for use with * lower sample rates to improve performance. @@ -1005,12 +954,15 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream, static int wm8350_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - struct wm8350 *wm8350 = codec->control_data; + unsigned int val; if (mute) - wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA); + val = WM8350_DAC_MUTE_ENA; else - wm8350_clear_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA); + val = 0; + + snd_soc_update_bits(codec, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA, val); + return 0; } @@ -1079,8 +1031,8 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; - struct wm8350 *wm8350 = codec->control_data; struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec); + struct wm8350 *wm8350 = priv->wm8350; struct _fll_div fll_div; int ret = 0; u16 fll_1, fll_4; @@ -1104,17 +1056,17 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai, fll_div.ratio); /* set up N.K & dividers */ - fll_1 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_1) & + fll_1 = snd_soc_read(codec, WM8350_FLL_CONTROL_1) & ~(WM8350_FLL_OUTDIV_MASK | WM8350_FLL_RSP_RATE_MASK | 0xc000); - wm8350_codec_write(codec, WM8350_FLL_CONTROL_1, + snd_soc_write(codec, WM8350_FLL_CONTROL_1, fll_1 | (fll_div.div << 8) | 0x50); - wm8350_codec_write(codec, WM8350_FLL_CONTROL_2, + snd_soc_write(codec, WM8350_FLL_CONTROL_2, (fll_div.ratio << 11) | (fll_div. n & WM8350_FLL_N_MASK)); - wm8350_codec_write(codec, WM8350_FLL_CONTROL_3, fll_div.k); - fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) & + snd_soc_write(codec, WM8350_FLL_CONTROL_3, fll_div.k); + fll_4 = snd_soc_read(codec, WM8350_FLL_CONTROL_4) & ~(WM8350_FLL_FRAC | WM8350_FLL_SLOW_LOCK_REF); - wm8350_codec_write(codec, WM8350_FLL_CONTROL_4, + snd_soc_write(codec, WM8350_FLL_CONTROL_4, fll_4 | (fll_div.k ? WM8350_FLL_FRAC : 0) | (fll_div.ratio == 8 ? WM8350_FLL_SLOW_LOCK_REF : 0)); @@ -1131,8 +1083,8 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai, static int wm8350_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - struct wm8350 *wm8350 = codec->control_data; struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec); + struct wm8350 *wm8350 = priv->wm8350; struct wm8350_audio_platform_data *platform = wm8350->codec.platform_data; u16 pm1; @@ -1339,35 +1291,36 @@ static void wm8350_hpr_work(struct work_struct *work) wm8350_hp_work(priv, &priv->hpr, WM8350_JACK_R_LVL); } -static irqreturn_t wm8350_hp_jack_handler(int irq, void *data) +static irqreturn_t wm8350_hpl_jack_handler(int irq, void *data) { struct wm8350_data *priv = data; struct wm8350 *wm8350 = priv->wm8350; - struct wm8350_jack_data *jack = NULL; - switch (irq - wm8350->irq_base) { - case WM8350_IRQ_CODEC_JCK_DET_L: #ifndef CONFIG_SND_SOC_WM8350_MODULE - trace_snd_soc_jack_irq("WM8350 HPL"); + trace_snd_soc_jack_irq("WM8350 HPL"); #endif - jack = &priv->hpl; - break; - case WM8350_IRQ_CODEC_JCK_DET_R: + if (device_may_wakeup(wm8350->dev)) + pm_wakeup_event(wm8350->dev, 250); + + schedule_delayed_work(&priv->hpl.work, 200); + + return IRQ_HANDLED; +} + +static irqreturn_t wm8350_hpr_jack_handler(int irq, void *data) +{ + struct wm8350_data *priv = data; + struct wm8350 *wm8350 = priv->wm8350; + #ifndef CONFIG_SND_SOC_WM8350_MODULE - trace_snd_soc_jack_irq("WM8350 HPR"); + trace_snd_soc_jack_irq("WM8350 HPR"); #endif - jack = &priv->hpr; - break; - - default: - BUG(); - } if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&jack->work, 200); + schedule_delayed_work(&priv->hpr.work, 200); return IRQ_HANDLED; } @@ -1387,7 +1340,7 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, struct snd_soc_jack *jack, int report) { struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec); - struct wm8350 *wm8350 = codec->control_data; + struct wm8350 *wm8350 = priv->wm8350; int irq; int ena; @@ -1418,7 +1371,14 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, } /* Sync status */ - wm8350_hp_jack_handler(irq + wm8350->irq_base, priv); + switch (which) { + case WM8350_JDL: + wm8350_hpl_jack_handler(0, priv); + break; + case WM8350_JDR: + wm8350_hpr_jack_handler(0, priv); + break; + } return 0; } @@ -1463,7 +1423,7 @@ int wm8350_mic_jack_detect(struct snd_soc_codec *codec, int detect_report, int short_report) { struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec); - struct wm8350 *wm8350 = codec->control_data; + struct wm8350 *wm8350 = priv->wm8350; priv->mic.jack = jack; priv->mic.report = detect_report; @@ -1491,7 +1451,6 @@ EXPORT_SYMBOL_GPL(wm8350_mic_jack_detect); static const struct snd_soc_dai_ops wm8350_dai_ops = { .hw_params = wm8350_pcm_hw_params, .digital_mute = wm8350_mute, - .trigger = wm8350_pcm_trigger, .set_fmt = wm8350_set_dai_fmt, .set_sysclk = wm8350_set_dai_sysclk, .set_pll = wm8350_set_fll, @@ -1546,7 +1505,9 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; - codec->control_data = wm8350; + codec->control_data = wm8350->regmap; + + snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); /* Put the codec into reset if it wasn't already */ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); @@ -1559,9 +1520,9 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) wm8350_set_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); /* Enable robust clocking mode in ADC */ - wm8350_codec_write(codec, WM8350_SECURITY, 0xa7); - wm8350_codec_write(codec, 0xde, 0x13); - wm8350_codec_write(codec, WM8350_SECURITY, 0); + snd_soc_write(codec, WM8350_SECURITY, 0xa7); + snd_soc_write(codec, 0xde, 0x13); + snd_soc_write(codec, WM8350_SECURITY, 0); /* read OUT1 & OUT2 volumes */ out1 = &priv->out1; @@ -1601,10 +1562,10 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) WM8350_JDL_ENA | WM8350_JDR_ENA); wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, - wm8350_hp_jack_handler, 0, "Left jack detect", + wm8350_hpl_jack_handler, 0, "Left jack detect", priv); wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, - wm8350_hp_jack_handler, 0, "Right jack detect", + wm8350_hpr_jack_handler, 0, "Right jack detect", priv); wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, wm8350_mic_handler, 0, "Microphone short", priv); @@ -1656,8 +1617,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8350 = { .remove = wm8350_codec_remove, .suspend = wm8350_suspend, .resume = wm8350_resume, - .read = wm8350_codec_read, - .write = wm8350_codec_write, .set_bias_level = wm8350_set_bias_level, .controls = wm8350_snd_controls, diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 898979d23010..5d277a915f81 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1,7 +1,7 @@ /* * wm8400.c -- WM8400 ALSA Soc Audio driver * - * Copyright 2008, 2009 Wolfson Microelectronics PLC. + * Copyright 2008-11 Wolfson Microelectronics PLC. * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * * This program is free software; you can redistribute it and/or modify it @@ -138,8 +138,8 @@ static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, return ret; /* now hit the volume update bits (always bit 8) */ - val = wm8400_read(codec, reg); - return wm8400_write(codec, reg, val | 0x0100); + val = snd_soc_read(codec, reg); + return snd_soc_write(codec, reg, val | 0x0100); } #define WM8400_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert, tlv_array) \ @@ -362,8 +362,8 @@ static int inmixer_event (struct snd_soc_dapm_widget *w, { u16 reg, fakepower; - reg = wm8400_read(w->codec, WM8400_POWER_MANAGEMENT_2); - fakepower = wm8400_read(w->codec, WM8400_INTDRIVBITS); + reg = snd_soc_read(w->codec, WM8400_POWER_MANAGEMENT_2); + fakepower = snd_soc_read(w->codec, WM8400_INTDRIVBITS); if (fakepower & ((1 << WM8400_INMIXL_PWR) | (1 << WM8400_AINLMUX_PWR))) { @@ -378,7 +378,7 @@ static int inmixer_event (struct snd_soc_dapm_widget *w, } else { reg &= ~WM8400_AINR_ENA; } - wm8400_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg); + snd_soc_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg); return 0; } @@ -394,7 +394,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, switch (reg_shift) { case WM8400_SPEAKER_MIXER | (WM8400_LDSPK << 8) : - reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER1); + reg = snd_soc_read(w->codec, WM8400_OUTPUT_MIXER1); if (reg & WM8400_LDLO) { printk(KERN_WARNING "Cannot set as Output Mixer 1 LDLO Set\n"); @@ -402,7 +402,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, } break; case WM8400_SPEAKER_MIXER | (WM8400_RDSPK << 8): - reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER2); + reg = snd_soc_read(w->codec, WM8400_OUTPUT_MIXER2); if (reg & WM8400_RDRO) { printk(KERN_WARNING "Cannot set as Output Mixer 2 RDRO Set\n"); @@ -410,7 +410,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, } break; case WM8400_OUTPUT_MIXER1 | (WM8400_LDLO << 8): - reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER); + reg = snd_soc_read(w->codec, WM8400_SPEAKER_MIXER); if (reg & WM8400_LDSPK) { printk(KERN_WARNING "Cannot set as Speaker Mixer LDSPK Set\n"); @@ -418,7 +418,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w, } break; case WM8400_OUTPUT_MIXER2 | (WM8400_RDRO << 8): - reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER); + reg = snd_soc_read(w->codec, WM8400_SPEAKER_MIXER); if (reg & WM8400_RDSPK) { printk(KERN_WARNING "Cannot set as Speaker Mixer RDSPK Set\n"); @@ -1021,13 +1021,13 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, wm8400->fll_in = freq_in; /* We *must* disable the FLL before any changes */ - reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_2); + reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_2); reg &= ~WM8400_FLL_ENA; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_2, reg); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_2, reg); - reg = wm8400_read(codec, WM8400_FLL_CONTROL_1); + reg = snd_soc_read(codec, WM8400_FLL_CONTROL_1); reg &= ~WM8400_FLL_OSC_ENA; - wm8400_write(codec, WM8400_FLL_CONTROL_1, reg); + snd_soc_write(codec, WM8400_FLL_CONTROL_1, reg); if (!freq_out) return 0; @@ -1035,15 +1035,15 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, reg &= ~(WM8400_FLL_REF_FREQ | WM8400_FLL_FRATIO_MASK); reg |= WM8400_FLL_FRAC | factors.fratio; reg |= factors.freq_ref << WM8400_FLL_REF_FREQ_SHIFT; - wm8400_write(codec, WM8400_FLL_CONTROL_1, reg); + snd_soc_write(codec, WM8400_FLL_CONTROL_1, reg); - wm8400_write(codec, WM8400_FLL_CONTROL_2, factors.k); - wm8400_write(codec, WM8400_FLL_CONTROL_3, factors.n); + snd_soc_write(codec, WM8400_FLL_CONTROL_2, factors.k); + snd_soc_write(codec, WM8400_FLL_CONTROL_3, factors.n); - reg = wm8400_read(codec, WM8400_FLL_CONTROL_4); + reg = snd_soc_read(codec, WM8400_FLL_CONTROL_4); reg &= ~WM8400_FLL_OUTDIV_MASK; reg |= factors.outdiv; - wm8400_write(codec, WM8400_FLL_CONTROL_4, reg); + snd_soc_write(codec, WM8400_FLL_CONTROL_4, reg); return 0; } @@ -1057,8 +1057,8 @@ static int wm8400_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; u16 audio1, audio3; - audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1); - audio3 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_3); + audio1 = snd_soc_read(codec, WM8400_AUDIO_INTERFACE_1); + audio3 = snd_soc_read(codec, WM8400_AUDIO_INTERFACE_3); /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -1099,8 +1099,8 @@ static int wm8400_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); - wm8400_write(codec, WM8400_AUDIO_INTERFACE_3, audio3); + snd_soc_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); + snd_soc_write(codec, WM8400_AUDIO_INTERFACE_3, audio3); return 0; } @@ -1112,24 +1112,24 @@ static int wm8400_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div_id) { case WM8400_MCLK_DIV: - reg = wm8400_read(codec, WM8400_CLOCKING_2) & + reg = snd_soc_read(codec, WM8400_CLOCKING_2) & ~WM8400_MCLK_DIV_MASK; - wm8400_write(codec, WM8400_CLOCKING_2, reg | div); + snd_soc_write(codec, WM8400_CLOCKING_2, reg | div); break; case WM8400_DACCLK_DIV: - reg = wm8400_read(codec, WM8400_CLOCKING_2) & + reg = snd_soc_read(codec, WM8400_CLOCKING_2) & ~WM8400_DAC_CLKDIV_MASK; - wm8400_write(codec, WM8400_CLOCKING_2, reg | div); + snd_soc_write(codec, WM8400_CLOCKING_2, reg | div); break; case WM8400_ADCCLK_DIV: - reg = wm8400_read(codec, WM8400_CLOCKING_2) & + reg = snd_soc_read(codec, WM8400_CLOCKING_2) & ~WM8400_ADC_CLKDIV_MASK; - wm8400_write(codec, WM8400_CLOCKING_2, reg | div); + snd_soc_write(codec, WM8400_CLOCKING_2, reg | div); break; case WM8400_BCLK_DIV: - reg = wm8400_read(codec, WM8400_CLOCKING_1) & + reg = snd_soc_read(codec, WM8400_CLOCKING_1) & ~WM8400_BCLK_DIV_MASK; - wm8400_write(codec, WM8400_CLOCKING_1, reg | div); + snd_soc_write(codec, WM8400_CLOCKING_1, reg | div); break; default: return -EINVAL; @@ -1145,9 +1145,8 @@ static int wm8400_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; - u16 audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1); + struct snd_soc_codec *codec = dai->codec; + u16 audio1 = snd_soc_read(codec, WM8400_AUDIO_INTERFACE_1); audio1 &= ~WM8400_AIF_WL_MASK; /* bit size */ @@ -1165,19 +1164,19 @@ static int wm8400_hw_params(struct snd_pcm_substream *substream, break; } - wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); + snd_soc_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); return 0; } static int wm8400_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 val = wm8400_read(codec, WM8400_DAC_CTRL) & ~WM8400_DAC_MUTE; + u16 val = snd_soc_read(codec, WM8400_DAC_CTRL) & ~WM8400_DAC_MUTE; if (mute) - wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); + snd_soc_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); else - wm8400_write(codec, WM8400_DAC_CTRL, val); + snd_soc_write(codec, WM8400_DAC_CTRL, val); return 0; } @@ -1196,9 +1195,9 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: /* VMID=2*50k */ - val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) & + val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1) & ~WM8400_VMID_MODE_MASK; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x2); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x2); break; case SND_SOC_BIAS_STANDBY: @@ -1212,74 +1211,74 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec, return ret; } - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, WM8400_CODEC_ENA | WM8400_SYSCLK_ENA); /* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | WM8400_BUFDCOPEN | WM8400_POBCTRL); msleep(50); /* Enable VREF & VMID at 2x50k */ - val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); + val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1); val |= 0x2 | WM8400_VREF_ENA; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val); /* Enable BUFIOEN */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | WM8400_BUFDCOPEN | WM8400_POBCTRL | WM8400_BUFIOEN); /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN); + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN); } /* VMID=2*300k */ - val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) & + val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1) & ~WM8400_VMID_MODE_MASK; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x4); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x4); break; case SND_SOC_BIAS_OFF: /* Enable POBCTRL and SOFT_ST */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | WM8400_POBCTRL | WM8400_BUFIOEN); /* Enable POBCTRL, SOFT_ST and BUFDCOPEN */ - wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + snd_soc_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | WM8400_BUFDCOPEN | WM8400_POBCTRL | WM8400_BUFIOEN); /* mute DAC */ - val = wm8400_read(codec, WM8400_DAC_CTRL); - wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); + val = snd_soc_read(codec, WM8400_DAC_CTRL); + snd_soc_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); /* Enable any disabled outputs */ - val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); + val = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1); val |= WM8400_SPK_ENA | WM8400_OUT3_ENA | WM8400_OUT4_ENA | WM8400_LOUT_ENA | WM8400_ROUT_ENA; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val); /* Disable VMID */ val &= ~WM8400_VMID_MODE_MASK; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val); msleep(300); /* Enable all output discharge bits */ - wm8400_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE | + snd_soc_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE | WM8400_DIS_RLINE | WM8400_DIS_OUT3 | WM8400_DIS_OUT4 | WM8400_DIS_LOUT | WM8400_DIS_ROUT); /* Disable VREF */ val &= ~WM8400_VREF_ENA; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, val); /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ - wm8400_write(codec, WM8400_ANTIPOP2, 0x0); + snd_soc_write(codec, WM8400_ANTIPOP2, 0x0); ret = regulator_bulk_disable(ARRAY_SIZE(power), &power[0]); @@ -1385,19 +1384,19 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec) wm8400_codec_reset(codec); - reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, reg | WM8400_CODEC_ENA); + reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, reg | WM8400_CODEC_ENA); /* Latch volume update bits */ - reg = wm8400_read(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME); - wm8400_write(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME, + reg = snd_soc_read(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME); + snd_soc_write(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME, reg & WM8400_IPVU); - reg = wm8400_read(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME); - wm8400_write(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME, + reg = snd_soc_read(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME); + snd_soc_write(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME, reg & WM8400_IPVU); - wm8400_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); - wm8400_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); + snd_soc_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); + snd_soc_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); if (!schedule_work(&priv->work)) { ret = -EINVAL; @@ -1414,8 +1413,8 @@ static int wm8400_codec_remove(struct snd_soc_codec *codec) { u16 reg; - reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, + reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1); + snd_soc_write(codec, WM8400_POWER_MANAGEMENT_1, reg & (~WM8400_CODEC_ENA)); regulator_bulk_free(ARRAY_SIZE(power), power); @@ -1428,7 +1427,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8400 = { .remove = wm8400_codec_remove, .suspend = wm8400_suspend, .resume = wm8400_resume, - .read = wm8400_read, + .read = snd_soc_read, .write = wm8400_write, .set_bias_level = wm8400_set_bias_level, diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 9166126bd312..56a049555e2c 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -392,8 +392,7 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 iface = snd_soc_read(codec, WM8510_IFACE) & 0x19f; u16 adn = snd_soc_read(codec, WM8510_ADD) & 0x1f1; diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 7fea2c3bf7e7..1c3ffb290cdc 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -145,8 +145,7 @@ static int wm8523_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); int i; u16 aifctrl1 = snd_soc_read(codec, WM8523_AIF_CTRL1); diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 211285164d70..7c68226376e4 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -1,7 +1,7 @@ /* * wm8580.c -- WM8580 ALSA Soc Audio driver * - * Copyright 2008, 2009 Wolfson Microelectronics PLC. + * Copyright 2008-11 Wolfson Microelectronics PLC. * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index fc3d59e49084..1467f97dce21 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -88,8 +88,7 @@ static int wm8728_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 dac = snd_soc_read(codec, WM8728_DACCTL); dac &= ~0x18; diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index a32caa72bd7d..bb1d26919b10 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -2,6 +2,7 @@ * wm8731.c -- WM8731 ALSA SoC Audio driver * * Copyright 2005 Openedhand Ltd. + * Copyright 2006-12 Wolfson Microelectronics, plc * * Author: Richard Purdie <richard@openedhand.com> * @@ -635,16 +636,17 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi) struct wm8731_priv *wm8731; int ret; - wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL); + wm8731 = devm_kzalloc(&spi->dev, sizeof(struct wm8731_priv), + GFP_KERNEL); if (wm8731 == NULL) return -ENOMEM; - wm8731->regmap = regmap_init_spi(spi, &wm8731_regmap); + wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); dev_err(&spi->dev, "Failed to allocate register map: %d\n", ret); - goto err; + return ret; } spi_set_drvdata(spi, wm8731); @@ -653,25 +655,15 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi) &soc_codec_dev_wm8731, &wm8731_dai, 1); if (ret != 0) { dev_err(&spi->dev, "Failed to register CODEC: %d\n", ret); - goto err_regmap; + return ret; } return 0; - -err_regmap: - regmap_exit(wm8731->regmap); -err: - kfree(wm8731); - return ret; } static int __devexit wm8731_spi_remove(struct spi_device *spi) { - struct wm8731_priv *wm8731 = spi_get_drvdata(spi); - snd_soc_unregister_codec(&spi->dev); - regmap_exit(wm8731->regmap); - kfree(wm8731); return 0; } @@ -693,16 +685,17 @@ static __devinit int wm8731_i2c_probe(struct i2c_client *i2c, struct wm8731_priv *wm8731; int ret; - wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL); + wm8731 = devm_kzalloc(&i2c->dev, sizeof(struct wm8731_priv), + GFP_KERNEL); if (wm8731 == NULL) return -ENOMEM; - wm8731->regmap = regmap_init_i2c(i2c, &wm8731_regmap); + wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); - goto err; + return ret; } i2c_set_clientdata(i2c, wm8731); @@ -711,24 +704,15 @@ static __devinit int wm8731_i2c_probe(struct i2c_client *i2c, &soc_codec_dev_wm8731, &wm8731_dai, 1); if (ret != 0) { dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); - goto err_regmap; + return ret; } return 0; - -err_regmap: - regmap_exit(wm8731->regmap); -err: - kfree(wm8731); - return ret; } static __devexit int wm8731_i2c_remove(struct i2c_client *client) { - struct wm8731_priv *wm8731 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); - regmap_exit(wm8731->regmap); - kfree(wm8731); return 0; } diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 4fe9d191e277..d0520124616d 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -329,8 +329,7 @@ static int wm8737_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); int i; u16 clocking = 0; diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 3941f50bf187..35f3d23200e0 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -1,7 +1,7 @@ /* * wm8741.c -- WM8741 ALSA SoC Audio driver * - * Copyright 2010 Wolfson Microelectronics plc + * Copyright 2010-1 Wolfson Microelectronics plc * * Author: Ian Lartey <ian@opensource.wolfsonmicro.com> * @@ -203,8 +203,7 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1FC; int i; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index e4c50ce7d9c0..89151ca5e776 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -547,8 +547,7 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8750_priv *wm8750 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8750_IFACE) & 0x1f3; u16 srate = snd_soc_read(codec, WM8750_SRATE) & 0x1c0; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index e27e7b62b365..13bff87ddcf5 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1,7 +1,7 @@ /* * wm8753.c -- WM8753 ALSA Soc Audio driver * - * Copyright 2003 Wolfson Microelectronics PLC. + * Copyright 2003-11 Wolfson Microelectronics PLC. * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it @@ -931,8 +931,7 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 voice = snd_soc_read(codec, WM8753_PCM) & 0x01f3; u16 srate = snd_soc_read(codec, WM8753_SRATE1) & 0x017f; @@ -1161,8 +1160,7 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 srate = snd_soc_read(codec, WM8753_SRATE1) & 0x01c0; u16 hifi = snd_soc_read(codec, WM8753_HIFI) & 0x01f3; diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index a19db5a0a17a..879c356a9045 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -1,7 +1,7 @@ /* * wm8776.c -- WM8776 ALSA SoC Audio driver * - * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2009-12 Wolfson Microelectronics plc * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 6bd1b767b138..c088020172ab 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -1,7 +1,7 @@ /* * wm8804.c -- WM8804 S/PDIF transceiver driver * - * Copyright 2010 Wolfson Microelectronics plc + * Copyright 2010-11 Wolfson Microelectronics plc * * Author: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> * diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index f18c554efc98..077c9628c70d 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -610,8 +610,7 @@ static int wm8900_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 reg; reg = snd_soc_read(codec, WM8900_REG_AUDIO1) & ~0x60; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index c91fb2f99c13..73f1c8d7bafb 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1,8 +1,8 @@ /* * wm8903.c -- WM8903 ALSA SoC Audio driver * - * Copyright 2008 Wolfson Microelectronics - * Copyright 2011 NVIDIA, Inc. + * Copyright 2008-12 Wolfson Microelectronics + * Copyright 2011-2012 NVIDIA, Inc. * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * @@ -116,6 +116,7 @@ static const struct reg_default wm8903_reg_defaults[] = { struct wm8903_priv { struct wm8903_platform_data *pdata; + struct device *dev; struct snd_soc_codec *codec; struct regmap *regmap; @@ -1432,8 +1433,7 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); int fs = params_rate(params); int bclk; @@ -1636,17 +1636,27 @@ EXPORT_SYMBOL_GPL(wm8903_mic_detect); static irqreturn_t wm8903_irq(int irq, void *data) { - struct snd_soc_codec *codec = data; - struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - int mic_report; - int int_pol; - int int_val = 0; - int mask = ~snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1_MASK); + struct wm8903_priv *wm8903 = data; + int mic_report, ret; + unsigned int int_val, mask, int_pol; - int_val = snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1) & mask; + ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_STATUS_1_MASK, + &mask); + if (ret != 0) { + dev_err(wm8903->dev, "Failed to read IRQ mask: %d\n", ret); + return IRQ_NONE; + } + + ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_STATUS_1, &int_val); + if (ret != 0) { + dev_err(wm8903->dev, "Failed to read IRQ status: %d\n", ret); + return IRQ_NONE; + } + + int_val &= ~mask; if (int_val & WM8903_WSEQ_BUSY_EINT) { - dev_warn(codec->dev, "Write sequencer done\n"); + dev_warn(wm8903->dev, "Write sequencer done\n"); } /* @@ -1657,22 +1667,28 @@ static irqreturn_t wm8903_irq(int irq, void *data) * the polarity register. */ mic_report = wm8903->mic_last_report; - int_pol = snd_soc_read(codec, WM8903_INTERRUPT_POLARITY_1); + ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_POLARITY_1, + &int_pol); + if (ret != 0) { + dev_err(wm8903->dev, "Failed to read interrupt polarity: %d\n", + ret); + return IRQ_HANDLED; + } #ifndef CONFIG_SND_SOC_WM8903_MODULE if (int_val & (WM8903_MICSHRT_EINT | WM8903_MICDET_EINT)) - trace_snd_soc_jack_irq(dev_name(codec->dev)); + trace_snd_soc_jack_irq(dev_name(wm8903->dev)); #endif if (int_val & WM8903_MICSHRT_EINT) { - dev_dbg(codec->dev, "Microphone short (pol=%x)\n", int_pol); + dev_dbg(wm8903->dev, "Microphone short (pol=%x)\n", int_pol); mic_report ^= wm8903->mic_short; int_pol ^= WM8903_MICSHRT_INV; } if (int_val & WM8903_MICDET_EINT) { - dev_dbg(codec->dev, "Microphone detect (pol=%x)\n", int_pol); + dev_dbg(wm8903->dev, "Microphone detect (pol=%x)\n", int_pol); mic_report ^= wm8903->mic_det; int_pol ^= WM8903_MICDET_INV; @@ -1680,8 +1696,8 @@ static irqreturn_t wm8903_irq(int irq, void *data) msleep(wm8903->mic_delay); } - snd_soc_update_bits(codec, WM8903_INTERRUPT_POLARITY_1, - WM8903_MICSHRT_INV | WM8903_MICDET_INV, int_pol); + regmap_update_bits(wm8903->regmap, WM8903_INTERRUPT_POLARITY_1, + WM8903_MICSHRT_INV | WM8903_MICDET_INV, int_pol); snd_soc_jack_report(wm8903->mic_jack, mic_report, wm8903->mic_short | wm8903->mic_det); @@ -1775,7 +1791,6 @@ static int wm8903_gpio_request(struct gpio_chip *chip, unsigned offset) static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); - struct snd_soc_codec *codec = wm8903->codec; unsigned int mask, val; int ret; @@ -1783,8 +1798,8 @@ static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) val = (WM8903_GPn_FN_GPIO_INPUT << WM8903_GP1_FN_SHIFT) | WM8903_GP1_DIR; - ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - mask, val); + ret = regmap_update_bits(wm8903->regmap, + WM8903_GPIO_CONTROL_1 + offset, mask, val); if (ret < 0) return ret; @@ -1794,10 +1809,9 @@ static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset) { struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); - struct snd_soc_codec *codec = wm8903->codec; - int reg; + unsigned int reg; - reg = snd_soc_read(codec, WM8903_GPIO_CONTROL_1 + offset); + regmap_read(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset, ®); return (reg & WM8903_GP1_LVL_MASK) >> WM8903_GP1_LVL_SHIFT; } @@ -1806,7 +1820,6 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); - struct snd_soc_codec *codec = wm8903->codec; unsigned int mask, val; int ret; @@ -1814,8 +1827,8 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip, val = (WM8903_GPn_FN_GPIO_OUTPUT << WM8903_GP1_FN_SHIFT) | (value << WM8903_GP2_LVL_SHIFT); - ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - mask, val); + ret = regmap_update_bits(wm8903->regmap, + WM8903_GPIO_CONTROL_1 + offset, mask, val); if (ret < 0) return ret; @@ -1825,11 +1838,10 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip, static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); - struct snd_soc_codec *codec = wm8903->codec; - snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - WM8903_GP1_LVL_MASK, - !!value << WM8903_GP1_LVL_SHIFT); + regmap_update_bits(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset, + WM8903_GP1_LVL_MASK, + !!value << WM8903_GP1_LVL_SHIFT); } static struct gpio_chip wm8903_template_chip = { @@ -1843,15 +1855,14 @@ static struct gpio_chip wm8903_template_chip = { .can_sleep = 1, }; -static void wm8903_init_gpio(struct snd_soc_codec *codec) +static void wm8903_init_gpio(struct wm8903_priv *wm8903) { - struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); struct wm8903_platform_data *pdata = wm8903->pdata; int ret; wm8903->gpio_chip = wm8903_template_chip; wm8903->gpio_chip.ngpio = WM8903_NUM_GPIO; - wm8903->gpio_chip.dev = codec->dev; + wm8903->gpio_chip.dev = wm8903->dev; if (pdata->gpio_base) wm8903->gpio_chip.base = pdata->gpio_base; @@ -1860,24 +1871,23 @@ static void wm8903_init_gpio(struct snd_soc_codec *codec) ret = gpiochip_add(&wm8903->gpio_chip); if (ret != 0) - dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); + dev_err(wm8903->dev, "Failed to add GPIOs: %d\n", ret); } -static void wm8903_free_gpio(struct snd_soc_codec *codec) +static void wm8903_free_gpio(struct wm8903_priv *wm8903) { - struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); int ret; ret = gpiochip_remove(&wm8903->gpio_chip); if (ret != 0) - dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret); + dev_err(wm8903->dev, "Failed to remove GPIOs: %d\n", ret); } #else -static void wm8903_init_gpio(struct snd_soc_codec *codec) +static void wm8903_init_gpio(struct wm8903_priv *wm8903) { } -static void wm8903_free_gpio(struct snd_soc_codec *codec) +static void wm8903_free_gpio(struct wm8903_priv *wm8903) { } #endif @@ -1885,11 +1895,7 @@ static void wm8903_free_gpio(struct snd_soc_codec *codec) static int wm8903_probe(struct snd_soc_codec *codec) { struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - struct wm8903_platform_data *pdata = wm8903->pdata; - int ret, i; - int trigger, irq_pol; - u16 val; - bool mic_gpio = false; + int ret; wm8903->codec = codec; codec->control_data = wm8903->regmap; @@ -1900,121 +1906,16 @@ static int wm8903_probe(struct snd_soc_codec *codec) return ret; } - /* Set up GPIOs, detect if any are MIC detect outputs */ - for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { - if ((!pdata->gpio_cfg[i]) || - (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO)) - continue; - - snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i, - pdata->gpio_cfg[i] & 0x7fff); - - val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK) - >> WM8903_GP1_FN_SHIFT; - - switch (val) { - case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT: - case WM8903_GPn_FN_MICBIAS_SHORT_DETECT: - mic_gpio = true; - break; - default: - break; - } - } - - /* Set up microphone detection */ - snd_soc_write(codec, WM8903_MIC_BIAS_CONTROL_0, - pdata->micdet_cfg); - - /* Microphone detection needs the WSEQ clock */ - if (pdata->micdet_cfg) - snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0, - WM8903_WSEQ_ENA, WM8903_WSEQ_ENA); - - /* If microphone detection is enabled by pdata but - * detected via IRQ then interrupts can be lost before - * the machine driver has set up microphone detection - * IRQs as the IRQs are clear on read. The detection - * will be enabled when the machine driver configures. - */ - WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA)); - - wm8903->mic_delay = pdata->micdet_delay; - - if (wm8903->irq) { - if (pdata->irq_active_low) { - trigger = IRQF_TRIGGER_LOW; - irq_pol = WM8903_IRQ_POL; - } else { - trigger = IRQF_TRIGGER_HIGH; - irq_pol = 0; - } - - snd_soc_update_bits(codec, WM8903_INTERRUPT_CONTROL, - WM8903_IRQ_POL, irq_pol); - - ret = request_threaded_irq(wm8903->irq, NULL, wm8903_irq, - trigger | IRQF_ONESHOT, - "wm8903", codec); - if (ret != 0) { - dev_err(codec->dev, "Failed to request IRQ: %d\n", - ret); - return ret; - } - - /* Enable write sequencer interrupts */ - snd_soc_update_bits(codec, WM8903_INTERRUPT_STATUS_1_MASK, - WM8903_IM_WSEQ_BUSY_EINT, 0); - } - /* power on device */ wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* Latch volume update bits */ - val = snd_soc_read(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT); - val |= WM8903_ADCVU; - snd_soc_write(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT, val); - snd_soc_write(codec, WM8903_ADC_DIGITAL_VOLUME_RIGHT, val); - - val = snd_soc_read(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT); - val |= WM8903_DACVU; - snd_soc_write(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT, val); - snd_soc_write(codec, WM8903_DAC_DIGITAL_VOLUME_RIGHT, val); - - val = snd_soc_read(codec, WM8903_ANALOGUE_OUT1_LEFT); - val |= WM8903_HPOUTVU; - snd_soc_write(codec, WM8903_ANALOGUE_OUT1_LEFT, val); - snd_soc_write(codec, WM8903_ANALOGUE_OUT1_RIGHT, val); - - val = snd_soc_read(codec, WM8903_ANALOGUE_OUT2_LEFT); - val |= WM8903_LINEOUTVU; - snd_soc_write(codec, WM8903_ANALOGUE_OUT2_LEFT, val); - snd_soc_write(codec, WM8903_ANALOGUE_OUT2_RIGHT, val); - - val = snd_soc_read(codec, WM8903_ANALOGUE_OUT3_LEFT); - val |= WM8903_SPKVU; - snd_soc_write(codec, WM8903_ANALOGUE_OUT3_LEFT, val); - snd_soc_write(codec, WM8903_ANALOGUE_OUT3_RIGHT, val); - - /* Enable DAC soft mute by default */ - snd_soc_update_bits(codec, WM8903_DAC_DIGITAL_1, - WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE, - WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE); - - wm8903_init_gpio(codec); - return ret; } /* power down chip */ static int wm8903_remove(struct snd_soc_codec *codec) { - struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - - wm8903_free_gpio(codec); wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (wm8903->irq) - free_irq(wm8903->irq, codec); return 0; } @@ -2124,15 +2025,18 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, { struct wm8903_platform_data *pdata = dev_get_platdata(&i2c->dev); struct wm8903_priv *wm8903; - unsigned int val; - int ret; + int trigger; + bool mic_gpio = false; + unsigned int val, irq_pol; + int ret, i; wm8903 = devm_kzalloc(&i2c->dev, sizeof(struct wm8903_priv), GFP_KERNEL); if (wm8903 == NULL) return -ENOMEM; + wm8903->dev = &i2c->dev; - wm8903->regmap = regmap_init_i2c(i2c, &wm8903_regmap); + wm8903->regmap = devm_regmap_init_i2c(i2c, &wm8903_regmap); if (IS_ERR(wm8903->regmap)) { ret = PTR_ERR(wm8903->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -2141,7 +2045,6 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, } i2c_set_clientdata(i2c, wm8903); - wm8903->irq = i2c->irq; /* If no platform data was supplied, create storage for defaults */ if (pdata) { @@ -2168,6 +2071,8 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, } } + pdata = wm8903->pdata; + ret = regmap_read(wm8903->regmap, WM8903_SW_RESET_AND_ID, &val); if (ret != 0) { dev_err(&i2c->dev, "Failed to read chip ID: %d\n", ret); @@ -2190,6 +2095,107 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, /* Reset the device */ regmap_write(wm8903->regmap, WM8903_SW_RESET_AND_ID, 0x8903); + wm8903_init_gpio(wm8903); + + /* Set up GPIO pin state, detect if any are MIC detect outputs */ + for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { + if ((!pdata->gpio_cfg[i]) || + (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO)) + continue; + + regmap_write(wm8903->regmap, WM8903_GPIO_CONTROL_1 + i, + pdata->gpio_cfg[i] & 0x7fff); + + val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK) + >> WM8903_GP1_FN_SHIFT; + + switch (val) { + case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT: + case WM8903_GPn_FN_MICBIAS_SHORT_DETECT: + mic_gpio = true; + break; + default: + break; + } + } + + /* Set up microphone detection */ + regmap_write(wm8903->regmap, WM8903_MIC_BIAS_CONTROL_0, + pdata->micdet_cfg); + + /* Microphone detection needs the WSEQ clock */ + if (pdata->micdet_cfg) + regmap_update_bits(wm8903->regmap, WM8903_WRITE_SEQUENCER_0, + WM8903_WSEQ_ENA, WM8903_WSEQ_ENA); + + /* If microphone detection is enabled by pdata but + * detected via IRQ then interrupts can be lost before + * the machine driver has set up microphone detection + * IRQs as the IRQs are clear on read. The detection + * will be enabled when the machine driver configures. + */ + WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA)); + + wm8903->mic_delay = pdata->micdet_delay; + + if (i2c->irq) { + if (pdata->irq_active_low) { + trigger = IRQF_TRIGGER_LOW; + irq_pol = WM8903_IRQ_POL; + } else { + trigger = IRQF_TRIGGER_HIGH; + irq_pol = 0; + } + + regmap_update_bits(wm8903->regmap, WM8903_INTERRUPT_CONTROL, + WM8903_IRQ_POL, irq_pol); + + ret = request_threaded_irq(i2c->irq, NULL, wm8903_irq, + trigger | IRQF_ONESHOT, + "wm8903", wm8903); + if (ret != 0) { + dev_err(wm8903->dev, "Failed to request IRQ: %d\n", + ret); + return ret; + } + + /* Enable write sequencer interrupts */ + regmap_update_bits(wm8903->regmap, + WM8903_INTERRUPT_STATUS_1_MASK, + WM8903_IM_WSEQ_BUSY_EINT, 0); + } + + /* Latch volume update bits */ + regmap_update_bits(wm8903->regmap, WM8903_ADC_DIGITAL_VOLUME_LEFT, + WM8903_ADCVU, WM8903_ADCVU); + regmap_update_bits(wm8903->regmap, WM8903_ADC_DIGITAL_VOLUME_RIGHT, + WM8903_ADCVU, WM8903_ADCVU); + + regmap_update_bits(wm8903->regmap, WM8903_DAC_DIGITAL_VOLUME_LEFT, + WM8903_DACVU, WM8903_DACVU); + regmap_update_bits(wm8903->regmap, WM8903_DAC_DIGITAL_VOLUME_RIGHT, + WM8903_DACVU, WM8903_DACVU); + + regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT1_LEFT, + WM8903_HPOUTVU, WM8903_HPOUTVU); + regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT1_RIGHT, + WM8903_HPOUTVU, WM8903_HPOUTVU); + + regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT2_LEFT, + WM8903_LINEOUTVU, WM8903_LINEOUTVU); + regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT2_RIGHT, + WM8903_LINEOUTVU, WM8903_LINEOUTVU); + + regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT3_LEFT, + WM8903_SPKVU, WM8903_SPKVU); + regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT3_RIGHT, + WM8903_SPKVU, WM8903_SPKVU); + + /* Enable DAC soft mute by default */ + regmap_update_bits(wm8903->regmap, WM8903_DAC_DIGITAL_1, + WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE, + WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8903, &wm8903_dai, 1); if (ret != 0) @@ -2197,7 +2203,6 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, return 0; err: - regmap_exit(wm8903->regmap); return ret; } @@ -2205,7 +2210,9 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client) { struct wm8903_priv *wm8903 = i2c_get_clientdata(client); - regmap_exit(wm8903->regmap); + if (client->irq) + free_irq(client->irq, wm8903); + wm8903_free_gpio(wm8903); snd_soc_unregister_codec(&client->dev); return 0; diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 65d525d74c54..0013afe48e66 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1,7 +1,7 @@ /* * wm8904.c -- WM8904 ALSA SoC Audio driver * - * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2009-12 Wolfson Microelectronics plc * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * @@ -314,11 +314,6 @@ static bool wm8904_readable_register(struct device *dev, unsigned int reg) } } -static int wm8904_reset(struct snd_soc_codec *codec) -{ - return snd_soc_write(codec, WM8904_SW_RESET_AND_ID, 0); -} - static int wm8904_configure_clocking(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); @@ -1863,6 +1858,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, return ret; } + regcache_cache_only(wm8904->regmap, false); regcache_sync(wm8904->regmap); /* Enable bias */ @@ -1899,14 +1895,8 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, WM8904_BIAS_ENA, 0); -#ifdef CONFIG_REGULATOR - /* Post 2.6.34 we will be able to get a callback when - * the regulators are disabled which we can use but - * for now just assume that the power will be cut if - * the regulator API is in use. - */ - codec->cache_sync = 1; -#endif + regcache_cache_only(wm8904->regmap, true); + regcache_mark_dirty(wm8904->regmap); regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); @@ -1950,25 +1940,6 @@ static struct snd_soc_dai_driver wm8904_dai = { .symmetric_rates = 1, }; -#ifdef CONFIG_PM -static int wm8904_suspend(struct snd_soc_codec *codec) -{ - wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int wm8904_resume(struct snd_soc_codec *codec) -{ - wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} -#else -#define wm8904_suspend NULL -#define wm8904_resume NULL -#endif - static void wm8904_handle_retune_mobile_pdata(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); @@ -2083,11 +2054,8 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec) static int wm8904_probe(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - struct wm8904_pdata *pdata = wm8904->pdata; - u16 *reg_cache = codec->reg_cache; - int ret, i; + int ret; - codec->cache_sync = 1; codec->control_data = wm8904->regmap; switch (wm8904->devtype) { @@ -2108,122 +2076,17 @@ static int wm8904_probe(struct snd_soc_codec *codec) return ret; } - for (i = 0; i < ARRAY_SIZE(wm8904->supplies); i++) - wm8904->supplies[i].supply = wm8904_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8904->supplies), - wm8904->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - return ret; - } - - ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies), - wm8904->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); - goto err_get; - } - - ret = snd_soc_read(codec, WM8904_SW_RESET_AND_ID); - if (ret < 0) { - dev_err(codec->dev, "Failed to read ID register\n"); - goto err_enable; - } - if (ret != 0x8904) { - dev_err(codec->dev, "Device is not a WM8904, ID is %x\n", ret); - ret = -EINVAL; - goto err_enable; - } - - ret = snd_soc_read(codec, WM8904_REVISION); - if (ret < 0) { - dev_err(codec->dev, "Failed to read device revision: %d\n", - ret); - goto err_enable; - } - dev_info(codec->dev, "revision %c\n", ret + 'A'); - - ret = wm8904_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset\n"); - goto err_enable; - } - - /* Change some default settings - latch VU and enable ZC */ - snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT, - WM8904_ADC_VU, WM8904_ADC_VU); - snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_RIGHT, - WM8904_ADC_VU, WM8904_ADC_VU); - snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_LEFT, - WM8904_DAC_VU, WM8904_DAC_VU); - snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_RIGHT, - WM8904_DAC_VU, WM8904_DAC_VU); - snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_LEFT, - WM8904_HPOUT_VU | WM8904_HPOUTLZC, - WM8904_HPOUT_VU | WM8904_HPOUTLZC); - snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_RIGHT, - WM8904_HPOUT_VU | WM8904_HPOUTRZC, - WM8904_HPOUT_VU | WM8904_HPOUTRZC); - snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_LEFT, - WM8904_LINEOUT_VU | WM8904_LINEOUTLZC, - WM8904_LINEOUT_VU | WM8904_LINEOUTLZC); - snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_RIGHT, - WM8904_LINEOUT_VU | WM8904_LINEOUTRZC, - WM8904_LINEOUT_VU | WM8904_LINEOUTRZC); - snd_soc_update_bits(codec, WM8904_CLOCK_RATES_0, - WM8904_SR_MODE, 0); - - /* Apply configuration from the platform data. */ - if (wm8904->pdata) { - for (i = 0; i < WM8904_GPIO_REGS; i++) { - if (!pdata->gpio_cfg[i]) - continue; - - reg_cache[WM8904_GPIO_CONTROL_1 + i] - = pdata->gpio_cfg[i] & 0xffff; - } - - /* Zero is the default value for these anyway */ - for (i = 0; i < WM8904_MIC_REGS; i++) - reg_cache[WM8904_MIC_BIAS_CONTROL_0 + i] - = pdata->mic_cfg[i]; - } - - /* Set Class W by default - this will be managed by the Class - * G widget at runtime where bypass paths are available. - */ - snd_soc_update_bits(codec, WM8904_CLASS_W_0, - WM8904_CP_DYN_PWR, WM8904_CP_DYN_PWR); - - /* Use normal bias source */ - snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, - WM8904_POBCTRL, 0); - - wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - /* Bias level configuration will have done an extra enable */ - regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); - wm8904_handle_pdata(codec); wm8904_add_widgets(codec); return 0; - -err_enable: - regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); -err_get: - regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); - return ret; } static int wm8904_remove(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); kfree(wm8904->retune_mobile_texts); kfree(wm8904->drc_texts); @@ -2233,8 +2096,6 @@ static int wm8904_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_wm8904 = { .probe = wm8904_probe, .remove = wm8904_remove, - .suspend = wm8904_suspend, - .resume = wm8904_resume, .set_bias_level = wm8904_set_bias_level, .idle_bias_off = true, }; @@ -2256,14 +2117,15 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8904_priv *wm8904; - int ret; + unsigned int val; + int ret, i; wm8904 = devm_kzalloc(&i2c->dev, sizeof(struct wm8904_priv), GFP_KERNEL); if (wm8904 == NULL) return -ENOMEM; - wm8904->regmap = regmap_init_i2c(i2c, &wm8904_regmap); + wm8904->regmap = devm_regmap_init_i2c(i2c, &wm8904_regmap); if (IS_ERR(wm8904->regmap)) { ret = PTR_ERR(wm8904->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -2275,23 +2137,121 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8904); wm8904->pdata = i2c->dev.platform_data; + for (i = 0; i < ARRAY_SIZE(wm8904->supplies); i++) + wm8904->supplies[i].supply = wm8904_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8904->supplies), + wm8904->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies), + wm8904->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + ret = regmap_read(wm8904->regmap, WM8904_SW_RESET_AND_ID, &val); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret); + goto err_enable; + } + if (val != 0x8904) { + dev_err(&i2c->dev, "Device is not a WM8904, ID is %x\n", val); + ret = -EINVAL; + goto err_enable; + } + + ret = regmap_read(wm8904->regmap, WM8904_REVISION, &val); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read device revision: %d\n", + ret); + goto err_enable; + } + dev_info(&i2c->dev, "revision %c\n", val + 'A'); + + ret = regmap_write(wm8904->regmap, WM8904_SW_RESET_AND_ID, 0); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret); + goto err_enable; + } + + /* Change some default settings - latch VU and enable ZC */ + regmap_update_bits(wm8904->regmap, WM8904_ADC_DIGITAL_VOLUME_LEFT, + WM8904_ADC_VU, WM8904_ADC_VU); + regmap_update_bits(wm8904->regmap, WM8904_ADC_DIGITAL_VOLUME_RIGHT, + WM8904_ADC_VU, WM8904_ADC_VU); + regmap_update_bits(wm8904->regmap, WM8904_DAC_DIGITAL_VOLUME_LEFT, + WM8904_DAC_VU, WM8904_DAC_VU); + regmap_update_bits(wm8904->regmap, WM8904_DAC_DIGITAL_VOLUME_RIGHT, + WM8904_DAC_VU, WM8904_DAC_VU); + regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT1_LEFT, + WM8904_HPOUT_VU | WM8904_HPOUTLZC, + WM8904_HPOUT_VU | WM8904_HPOUTLZC); + regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT1_RIGHT, + WM8904_HPOUT_VU | WM8904_HPOUTRZC, + WM8904_HPOUT_VU | WM8904_HPOUTRZC); + regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT2_LEFT, + WM8904_LINEOUT_VU | WM8904_LINEOUTLZC, + WM8904_LINEOUT_VU | WM8904_LINEOUTLZC); + regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT2_RIGHT, + WM8904_LINEOUT_VU | WM8904_LINEOUTRZC, + WM8904_LINEOUT_VU | WM8904_LINEOUTRZC); + regmap_update_bits(wm8904->regmap, WM8904_CLOCK_RATES_0, + WM8904_SR_MODE, 0); + + /* Apply configuration from the platform data. */ + if (wm8904->pdata) { + for (i = 0; i < WM8904_GPIO_REGS; i++) { + if (!wm8904->pdata->gpio_cfg[i]) + continue; + + regmap_update_bits(wm8904->regmap, + WM8904_GPIO_CONTROL_1 + i, + 0xffff, + wm8904->pdata->gpio_cfg[i]); + } + + /* Zero is the default value for these anyway */ + for (i = 0; i < WM8904_MIC_REGS; i++) + regmap_update_bits(wm8904->regmap, + WM8904_MIC_BIAS_CONTROL_0 + i, + 0xffff, + wm8904->pdata->mic_cfg[i]); + } + + /* Set Class W by default - this will be managed by the Class + * G widget at runtime where bypass paths are available. + */ + regmap_update_bits(wm8904->regmap, WM8904_CLASS_W_0, + WM8904_CP_DYN_PWR, WM8904_CP_DYN_PWR); + + /* Use normal bias source */ + regmap_update_bits(wm8904->regmap, WM8904_BIAS_CONTROL_0, + WM8904_POBCTRL, 0); + + /* Can leave the device powered off until we need it */ + regcache_cache_only(wm8904->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8904, &wm8904_dai, 1); if (ret != 0) - goto err; + return ret; return 0; -err: - regmap_exit(wm8904->regmap); +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); return ret; } static __devexit int wm8904_i2c_remove(struct i2c_client *client) { - struct wm8904_priv *wm8904 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); - regmap_exit(wm8904->regmap); return 0; } @@ -2313,23 +2273,7 @@ static struct i2c_driver wm8904_i2c_driver = { .id_table = wm8904_i2c_id, }; -static int __init wm8904_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8904_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register wm8904 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8904_modinit); - -static void __exit wm8904_exit(void) -{ - i2c_del_driver(&wm8904_i2c_driver); -} -module_exit(wm8904_exit); +module_i2c_driver(wm8904_i2c_driver); MODULE_DESCRIPTION("ASoC WM8904 driver"); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index d2883affea3b..481a3d9cfe48 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -371,8 +371,7 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 iface = snd_soc_read(codec, WM8940_IFACE) & 0xFD9F; u16 addcntrl = snd_soc_read(codec, WM8940_ADDCNTRL) & 0xFFF1; u16 companding = snd_soc_read(codec, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 840d72086d04..96518ac8e24c 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1,6 +1,8 @@ /* * wm8960.c -- WM8960 ALSA SoC Audio driver * + * Copyright 2007-11 Wolfson Microelectronics, plc + * * Author: Liam Girdwood * * This program is free software; you can redistribute it and/or modify @@ -505,8 +507,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; int i; diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 05ea7c274093..01edbcc754d2 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1,6 +1,8 @@ /* * wm8961.c -- WM8961 ALSA SoC Audio driver * + * Copyright 2009-10 Wolfson Microelectronics, plc + * * Author: Mark Brown * * This program is free software; you can redistribute it and/or modify diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 15d467ff91b4..eaf65863ec21 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1,7 +1,7 @@ /* * wm8962.c -- WM8962 ALSA SoC Audio driver * - * Copyright 2010 Wolfson Microelectronics plc + * Copyright 2010-2 Wolfson Microelectronics plc * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * @@ -1478,7 +1478,8 @@ static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static int wm8962_dsp2_write_config(struct snd_soc_codec *codec) { - return 0; + return regcache_sync_region(codec->control_data, + WM8962_HDBASS_AI_1, WM8962_MAX_REGISTER); } static int wm8962_dsp2_set_enable(struct snd_soc_codec *codec, u16 val) @@ -1755,10 +1756,22 @@ SOC_DOUBLE_R_TLV("EQ4 Volume", WM8962_EQ3, WM8962_EQ23, SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23, WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv), +SOC_SINGLE("3D Switch", WM8962_THREED1, 0, 1, 0), +SND_SOC_BYTES_MASK("3D Coefficients", WM8962_THREED1, 4, WM8962_THREED_ENA), + +SOC_SINGLE("DF1 Switch", WM8962_DF1, 0, 1, 0), +SND_SOC_BYTES_MASK("DF1 Coefficients", WM8962_DF1, 7, WM8962_DF1_ENA), + +SOC_SINGLE("DRC Switch", WM8962_DRC_1, 0, 1, 0), +SND_SOC_BYTES_MASK("DRC Coefficients", WM8962_DRC_1, 5, WM8962_DRC_ENA), + WM8962_DSP2_ENABLE("VSS Switch", WM8962_VSS_ENA_SHIFT), +SND_SOC_BYTES("VSS Coefficients", WM8962_VSS_XHD2_1, 148), WM8962_DSP2_ENABLE("HPF1 Switch", WM8962_HPF1_ENA_SHIFT), WM8962_DSP2_ENABLE("HPF2 Switch", WM8962_HPF2_ENA_SHIFT), +SND_SOC_BYTES("HPF Coefficients", WM8962_LHPF2, 1), WM8962_DSP2_ENABLE("HD Bass Switch", WM8962_HDBASS_ENA_SHIFT), +SND_SOC_BYTES("HD Bass Coefficients", WM8962_HDBASS_AI_1, 30), }; static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = { @@ -2519,8 +2532,7 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); int i; int aif0 = 0; @@ -2568,6 +2580,9 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, WM8962_SAMPLE_RATE_INT_MODE | WM8962_SAMPLE_RATE_MASK, adctl3); + dev_dbg(codec->dev, "hw_params set BCLK %dHz LRCLK %dHz\n", + wm8962->bclk, wm8962->lrclk); + if (codec->dapm.bias_level == SND_SOC_BIAS_ON) wm8962_configure_bclk(codec); @@ -3710,6 +3725,9 @@ static int wm8962_runtime_resume(struct device *dev) } regcache_cache_only(wm8962->regmap, false); + + wm8962_reset(wm8962); + regcache_sync(wm8962->regmap); regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 28fe59e3ce01..eef783f6b6d6 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -478,8 +478,7 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8971_IFACE) & 0x1f3; u16 srate = snd_soc_read(codec, WM8971_SRATE) & 0x1c0; diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 72d5fdcd3cc2..a5be3adecf75 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -723,8 +723,7 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec); /* Word length mask = 0x60 */ u16 iface_ctl = snd_soc_read(codec, WM8978_AUDIO_INTERFACE) & ~0x60; diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 6cdf6a2bc283..1d4c5cf47b06 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -668,8 +668,7 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8988_IFACE) & 0x1f3; u16 srate = snd_soc_read(codec, WM8988_SRATE) & 0x180; diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 9d242351e6e8..db63c97ddf51 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1112,8 +1112,7 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 audio1 = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_1); audio1 &= ~WM8990_AIF_WL_MASK; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index d256a9340644..9fd80d688979 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1,7 +1,7 @@ /* * wm8993.c -- WM8993 ALSA SoC audio driver * - * Copyright 2009, 2010 Wolfson Microelectronics plc + * Copyright 2009-12 Wolfson Microelectronics plc * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * @@ -218,7 +218,6 @@ struct wm8993_priv { unsigned int sysclk_rate; unsigned int fs; unsigned int bclk; - int class_w_users; unsigned int fll_fref; unsigned int fll_fout; int fll_src; @@ -824,84 +823,6 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w, return 0; } -/* - * When used with DAC outputs only the WM8993 charge pump supports - * operation in class W mode, providing very low power consumption - * when used with digital sources. Enable and disable this mode - * automatically depending on the mixer configuration. - * - * Currently the only supported paths are the direct DAC->headphone - * paths (which provide minimum power consumption anyway). - */ -static int class_w_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; - struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); - int ret; - - /* Turn it off if we're using the main output mixer */ - if (ucontrol->value.integer.value[0] == 0) { - if (wm8993->class_w_users == 0) { - dev_dbg(codec->dev, "Disabling Class W\n"); - snd_soc_update_bits(codec, WM8993_CLASS_W_0, - WM8993_CP_DYN_FREQ | - WM8993_CP_DYN_V, - 0); - } - wm8993->class_w_users++; - wm8993->hubs_data.class_w = true; - } - - /* Implement the change */ - ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol); - - /* Enable it if we're using the direct DAC path */ - if (ucontrol->value.integer.value[0] == 1) { - if (wm8993->class_w_users == 1) { - dev_dbg(codec->dev, "Enabling Class W\n"); - snd_soc_update_bits(codec, WM8993_CLASS_W_0, - WM8993_CP_DYN_FREQ | - WM8993_CP_DYN_V, - WM8993_CP_DYN_FREQ | - WM8993_CP_DYN_V); - } - wm8993->class_w_users--; - wm8993->hubs_data.class_w = false; - } - - dev_dbg(codec->dev, "Indirect DAC use count now %d\n", - wm8993->class_w_users); - - return ret; -} - -#define SOC_DAPM_ENUM_W(xname, xenum) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_enum_double, \ - .get = snd_soc_dapm_get_enum_double, \ - .put = class_w_put, \ - .private_value = (unsigned long)&xenum } - -static const char *hp_mux_text[] = { - "Mixer", - "DAC", -}; - -static const struct soc_enum hpl_enum = - SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER1, 8, 2, hp_mux_text); - -static const struct snd_kcontrol_new hpl_mux = - SOC_DAPM_ENUM_W("Left Headphone Mux", hpl_enum); - -static const struct soc_enum hpr_enum = - SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER2, 8, 2, hp_mux_text); - -static const struct snd_kcontrol_new hpr_mux = - SOC_DAPM_ENUM_W("Right Headphone Mux", hpr_enum); - static const struct snd_kcontrol_new left_speaker_mixer[] = { SOC_DAPM_SINGLE("Input Switch", WM8993_SPEAKER_MIXER, 7, 1, 0), SOC_DAPM_SINGLE("IN1LP Switch", WM8993_SPEAKER_MIXER, 5, 1, 0), @@ -988,8 +909,8 @@ SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &sidetoner_mux), SND_SOC_DAPM_DAC("DACL", NULL, WM8993_POWER_MANAGEMENT_3, 1, 0), SND_SOC_DAPM_DAC("DACR", NULL, WM8993_POWER_MANAGEMENT_3, 0, 0), -SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), -SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpl_mux), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpr_mux), SND_SOC_DAPM_MIXER("SPKL", WM8993_POWER_MANAGEMENT_3, 8, 0, left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), @@ -1579,9 +1500,6 @@ static int wm8993_probe(struct snd_soc_codec *codec) return ret; } - /* By default we're using the output mixers */ - wm8993->class_w_users = 2; - /* Latch volume update bits and default ZC on */ snd_soc_update_bits(codec, WM8993_RIGHT_DAC_DIGITAL_VOLUME, WM8993_DAC_VU, WM8993_DAC_VU); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 6c1fe3afd4b5..bb62f4b3d563 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1,7 +1,7 @@ /* * wm8994.c -- WM8994 ALSA SoC Audio driver * - * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2009-12 Wolfson Microelectronics plc * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * @@ -46,6 +46,39 @@ #define WM8994_NUM_DRC 3 #define WM8994_NUM_EQ 3 +static struct { + unsigned int reg; + unsigned int mask; +} wm8994_vu_bits[] = { + { WM8994_LEFT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU }, + { WM8994_RIGHT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU }, + { WM8994_LEFT_LINE_INPUT_3_4_VOLUME, WM8994_IN2_VU }, + { WM8994_RIGHT_LINE_INPUT_3_4_VOLUME, WM8994_IN2_VU }, + { WM8994_SPEAKER_VOLUME_LEFT, WM8994_SPKOUT_VU }, + { WM8994_SPEAKER_VOLUME_RIGHT, WM8994_SPKOUT_VU }, + { WM8994_LEFT_OUTPUT_VOLUME, WM8994_HPOUT1_VU }, + { WM8994_RIGHT_OUTPUT_VOLUME, WM8994_HPOUT1_VU }, + { WM8994_LEFT_OPGA_VOLUME, WM8994_MIXOUT_VU }, + { WM8994_RIGHT_OPGA_VOLUME, WM8994_MIXOUT_VU }, + + { WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1DAC1_VU }, + { WM8994_AIF1_DAC1_RIGHT_VOLUME, WM8994_AIF1DAC1_VU }, + { WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU }, + { WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU }, + { WM8994_AIF2_DAC_LEFT_VOLUME, WM8994_AIF2DAC_VU }, + { WM8994_AIF2_DAC_RIGHT_VOLUME, WM8994_AIF2DAC_VU }, + { WM8994_AIF1_ADC1_LEFT_VOLUME, WM8994_AIF1ADC1_VU }, + { WM8994_AIF1_ADC1_RIGHT_VOLUME, WM8994_AIF1ADC1_VU }, + { WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU }, + { WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, + { WM8994_AIF2_ADC_LEFT_VOLUME, WM8994_AIF2ADC_VU }, + { WM8994_AIF2_ADC_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, + { WM8994_DAC1_LEFT_VOLUME, WM8994_DAC1_VU }, + { WM8994_DAC1_RIGHT_VOLUME, WM8994_DAC1_VU }, + { WM8994_DAC2_LEFT_VOLUME, WM8994_DAC2_VU }, + { WM8994_DAC2_RIGHT_VOLUME, WM8994_DAC2_VU }, +}; + static int wm8994_drc_base[] = { WM8994_AIF1_DRC1_1, WM8994_AIF1_DRC2_1, @@ -70,8 +103,8 @@ static const struct wm8958_micd_rate micdet_rates[] = { static const struct wm8958_micd_rate jackdet_rates[] = { { 32768, true, 0, 1 }, { 32768, false, 0, 1 }, - { 44100 * 256, true, 7, 10 }, - { 44100 * 256, false, 7, 10 }, + { 44100 * 256, true, 10, 10 }, + { 44100 * 256, false, 7, 8 }, }; static void wm8958_micd_set_rate(struct snd_soc_codec *codec) @@ -82,7 +115,8 @@ static void wm8958_micd_set_rate(struct snd_soc_codec *codec) const struct wm8958_micd_rate *rates; int num_rates; - if (wm8994->jack_cb != wm8958_default_micdet) + if (!(wm8994->pdata && wm8994->pdata->micd_rates) && + wm8994->jack_cb != wm8958_default_micdet) return; idle = !wm8994->jack_mic; @@ -118,6 +152,10 @@ static void wm8958_micd_set_rate(struct snd_soc_codec *codec) val = rates[best].start << WM8958_MICD_BIAS_STARTTIME_SHIFT | rates[best].rate << WM8958_MICD_RATE_SHIFT; + dev_dbg(codec->dev, "MICD rate %d,%d for %dHz %s\n", + rates[best].start, rates[best].rate, sysclk, + idle ? "idle" : "active"); + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_BIAS_STARTTIME_MASK | WM8958_MICD_RATE_MASK, val); @@ -398,7 +436,7 @@ static void wm8994_set_retune_mobile(struct snd_soc_codec *codec, int block) wm8994->dac_rates[iface]); /* The EQ will be disabled while reconfiguring it, remember the - * current configuration. + * current configuration. */ save = snd_soc_read(codec, base); save &= WM8994_AIF1DAC1_EQ_ENA; @@ -784,7 +822,7 @@ static void vmid_reference(struct snd_soc_codec *codec) switch (wm8994->vmid_mode) { default: - WARN_ON(0 == "Invalid VMID mode"); + WARN_ON(NULL == "Invalid VMID mode"); case WM8994_VMID_NORMAL: /* Startup bias, VMID ramp & buffer */ snd_soc_update_bits(codec, WM8994_ANTIPOP_2, @@ -937,27 +975,12 @@ static int vmid_event(struct snd_soc_dapm_widget *w, return 0; } -static void wm8994_update_class_w(struct snd_soc_codec *codec) +static bool wm8994_check_class_w_digital(struct snd_soc_codec *codec) { - struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - int enable = 1; int source = 0; /* GCC flow analysis can't track enable */ int reg, reg_r; - /* Only support direct DAC->headphone paths */ - reg = snd_soc_read(codec, WM8994_OUTPUT_MIXER_1); - if (!(reg & WM8994_DAC1L_TO_HPOUT1L)) { - dev_vdbg(codec->dev, "HPL connected to output mixer\n"); - enable = 0; - } - - reg = snd_soc_read(codec, WM8994_OUTPUT_MIXER_2); - if (!(reg & WM8994_DAC1R_TO_HPOUT1R)) { - dev_vdbg(codec->dev, "HPR connected to output mixer\n"); - enable = 0; - } - - /* We also need the same setting for L/R and only one path */ + /* We also need the same AIF source for L/R and only one path */ reg = snd_soc_read(codec, WM8994_DAC1_LEFT_MIXER_ROUTING); switch (reg) { case WM8994_AIF2DACL_TO_DAC1L: @@ -974,30 +997,20 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec) break; default: dev_vdbg(codec->dev, "DAC mixer setting: %x\n", reg); - enable = 0; - break; + return false; } reg_r = snd_soc_read(codec, WM8994_DAC1_RIGHT_MIXER_ROUTING); if (reg_r != reg) { dev_vdbg(codec->dev, "Left and right DAC mixers different\n"); - enable = 0; + return false; } - if (enable) { - dev_dbg(codec->dev, "Class W enabled\n"); - snd_soc_update_bits(codec, WM8994_CLASS_W_1, - WM8994_CP_DYN_PWR | - WM8994_CP_DYN_SRC_SEL_MASK, - source | WM8994_CP_DYN_PWR); - wm8994->hubs.class_w = true; - - } else { - dev_dbg(codec->dev, "Class W disabled\n"); - snd_soc_update_bits(codec, WM8994_CLASS_W_1, - WM8994_CP_DYN_PWR, 0); - wm8994->hubs.class_w = false; - } + /* Set the source up */ + snd_soc_update_bits(codec, WM8994_CLASS_W_1, + WM8994_CP_DYN_SRC_SEL_MASK, source); + + return true; } static int aif1clk_ev(struct snd_soc_dapm_widget *w, @@ -1006,6 +1019,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = w->codec; struct wm8994 *control = codec->control_data; int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; + int i; int dac; int adc; int val; @@ -1064,6 +1078,13 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, WM8994_AIF1DAC2L_ENA); break; + case SND_SOC_DAPM_POST_PMU: + for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) + snd_soc_write(codec, wm8994_vu_bits[i].reg, + snd_soc_read(codec, + wm8994_vu_bits[i].reg)); + break; + case SND_SOC_DAPM_PRE_PMD: case SND_SOC_DAPM_POST_PMD: snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, @@ -1089,6 +1110,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; + int i; int dac; int adc; int val; @@ -1139,12 +1161,19 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, WM8994_AIF2DACR_ENA); break; + case SND_SOC_DAPM_POST_PMU: + for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) + snd_soc_write(codec, wm8994_vu_bits[i].reg, + snd_soc_read(codec, + wm8994_vu_bits[i].reg)); + break; + case SND_SOC_DAPM_PRE_PMD: case SND_SOC_DAPM_POST_PMD: snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, WM8994_AIF2DACL_ENA | WM8994_AIF2DACR_ENA, 0); - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, WM8994_AIF2ADCL_ENA | WM8994_AIF2ADCR_ENA, 0); @@ -1207,17 +1236,19 @@ static int late_enable_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: if (wm8994->aif1clk_enable) { - aif1clk_ev(w, kcontrol, event); + aif1clk_ev(w, kcontrol, SND_SOC_DAPM_PRE_PMU); snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, WM8994_AIF1CLK_ENA_MASK, WM8994_AIF1CLK_ENA); + aif1clk_ev(w, kcontrol, SND_SOC_DAPM_POST_PMU); wm8994->aif1clk_enable = 0; } if (wm8994->aif2clk_enable) { - aif2clk_ev(w, kcontrol, event); + aif2clk_ev(w, kcontrol, SND_SOC_DAPM_PRE_PMU); snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, WM8994_AIF2CLK_ENA_MASK, WM8994_AIF2CLK_ENA); + aif2clk_ev(w, kcontrol, SND_SOC_DAPM_POST_PMU); wm8994->aif2clk_enable = 0; } break; @@ -1238,15 +1269,17 @@ static int late_disable_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMD: if (wm8994->aif1clk_disable) { + aif1clk_ev(w, kcontrol, SND_SOC_DAPM_PRE_PMD); snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, WM8994_AIF1CLK_ENA_MASK, 0); - aif1clk_ev(w, kcontrol, event); + aif1clk_ev(w, kcontrol, SND_SOC_DAPM_POST_PMD); wm8994->aif1clk_disable = 0; } if (wm8994->aif2clk_disable) { + aif2clk_ev(w, kcontrol, SND_SOC_DAPM_PRE_PMD); snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, WM8994_AIF2CLK_ENA_MASK, 0); - aif2clk_ev(w, kcontrol, event); + aif2clk_ev(w, kcontrol, SND_SOC_DAPM_POST_PMD); wm8994->aif2clk_disable = 0; } break; @@ -1280,45 +1313,6 @@ static int dac_ev(struct snd_soc_dapm_widget *w, return 0; } -static const char *hp_mux_text[] = { - "Mixer", - "DAC", -}; - -#define WM8994_HP_ENUM(xname, xenum) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_enum_double, \ - .get = snd_soc_dapm_get_enum_double, \ - .put = wm8994_put_hp_enum, \ - .private_value = (unsigned long)&xenum } - -static int wm8994_put_hp_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *w = wlist->widgets[0]; - struct snd_soc_codec *codec = w->codec; - int ret; - - ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol); - - wm8994_update_class_w(codec); - - return ret; -} - -static const struct soc_enum hpl_enum = - SOC_ENUM_SINGLE(WM8994_OUTPUT_MIXER_1, 8, 2, hp_mux_text); - -static const struct snd_kcontrol_new hpl_mux = - WM8994_HP_ENUM("Left Headphone Mux", hpl_enum); - -static const struct soc_enum hpr_enum = - SOC_ENUM_SINGLE(WM8994_OUTPUT_MIXER_2, 8, 2, hp_mux_text); - -static const struct snd_kcontrol_new hpr_mux = - WM8994_HP_ENUM("Right Headphone Mux", hpr_enum); - static const char *adc_mux_text[] = { "ADC", "DMIC", @@ -1430,7 +1424,7 @@ static int wm8994_put_class_w(struct snd_kcontrol *kcontrol, ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); - wm8994_update_class_w(codec); + wm_hubs_update_class_w(codec); return ret; } @@ -1524,7 +1518,7 @@ static const struct snd_kcontrol_new wm8958_aif3adc_mux = SOC_DAPM_ENUM("AIF3ADC Mux", wm8958_aif3adc_enum); static const char *mono_pcm_out_text[] = { - "None", "AIF2ADCL", "AIF2ADCR", + "None", "AIF2ADCL", "AIF2ADCR", }; static const struct soc_enum mono_pcm_out_enum = @@ -1573,9 +1567,9 @@ SND_SOC_DAPM_MIXER_E("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, SND_SOC_DAPM_MIXER_E("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer), late_enable_ev, SND_SOC_DAPM_PRE_PMU), -SND_SOC_DAPM_MUX_E("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux, +SND_SOC_DAPM_MUX_E("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpl_mux, late_enable_ev, SND_SOC_DAPM_PRE_PMU), -SND_SOC_DAPM_MUX_E("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux, +SND_SOC_DAPM_MUX_E("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpr_mux, late_enable_ev, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev) @@ -1583,16 +1577,18 @@ SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev) static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = { SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, aif1clk_ev, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, aif2clk_ev, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), SND_SOC_DAPM_MIXER("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), -SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), -SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpl_mux), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpr_mux), }; static const struct snd_soc_dapm_widget wm8994_dac_revd_widgets[] = { @@ -1732,6 +1728,7 @@ SND_SOC_DAPM_MUX("AIF3ADC Mux", SND_SOC_NOPM, 0, 0, &wm8994_aif3adc_mux), }; static const struct snd_soc_dapm_widget wm8958_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("AIF3", WM8994_POWER_MANAGEMENT_6, 5, 1, NULL, 0), SND_SOC_DAPM_MUX("Mono PCM Out Mux", SND_SOC_NOPM, 0, 0, &mono_pcm_out_mux), SND_SOC_DAPM_MUX("AIF2DACL Mux", SND_SOC_NOPM, 0, 0, &aif2dacl_src_mux), SND_SOC_DAPM_MUX("AIF2DACR Mux", SND_SOC_NOPM, 0, 0, &aif2dacr_src_mux), @@ -1972,6 +1969,9 @@ static const struct snd_soc_dapm_route wm8958_intercon[] = { { "AIF2DACR Mux", "AIF2", "AIF2DAC Mux" }, { "AIF2DACR Mux", "AIF3", "AIF3DACDAT" }, + { "AIF3DACDAT", NULL, "AIF3" }, + { "AIF3ADCDAT", NULL, "AIF3" }, + { "Mono PCM Out Mux", "AIF2ADCL", "AIF2ADCL" }, { "Mono PCM Out Mux", "AIF2ADCR", "AIF2ADCR" }, @@ -2068,24 +2068,20 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, struct wm8994 *control = wm8994->wm8994; int reg_offset, ret; struct fll_div fll; - u16 reg, aif1, aif2; + u16 reg, clk1, aif_reg, aif_src; unsigned long timeout; bool was_enabled; - aif1 = snd_soc_read(codec, WM8994_AIF1_CLOCKING_1) - & WM8994_AIF1CLK_ENA; - - aif2 = snd_soc_read(codec, WM8994_AIF2_CLOCKING_1) - & WM8994_AIF2CLK_ENA; - switch (id) { case WM8994_FLL1: reg_offset = 0; id = 0; + aif_src = 0x10; break; case WM8994_FLL2: reg_offset = 0x20; id = 1; + aif_src = 0x18; break; default: return -EINVAL; @@ -2127,16 +2123,33 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, if (ret < 0) return ret; - /* Gate the AIF clocks while we reclock */ - snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, - WM8994_AIF1CLK_ENA, 0); - snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, - WM8994_AIF2CLK_ENA, 0); + /* Make sure that we're not providing SYSCLK right now */ + clk1 = snd_soc_read(codec, WM8994_CLOCKING_1); + if (clk1 & WM8994_SYSCLK_SRC) + aif_reg = WM8994_AIF2_CLOCKING_1; + else + aif_reg = WM8994_AIF1_CLOCKING_1; + reg = snd_soc_read(codec, aif_reg); + + if ((reg & WM8994_AIF1CLK_ENA) && + (reg & WM8994_AIF1CLK_SRC_MASK) == aif_src) { + dev_err(codec->dev, "FLL%d is currently providing SYSCLK\n", + id + 1); + return -EBUSY; + } /* We always need to disable the FLL while reconfiguring */ snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset, WM8994_FLL1_ENA, 0); + if (wm8994->fll_byp && src == WM8994_FLL_SRC_BCLK && + freq_in == freq_out && freq_out) { + dev_dbg(codec->dev, "Bypassing FLL%d\n", id + 1); + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, + WM8958_FLL1_BYP, WM8958_FLL1_BYP); + goto out; + } + reg = (fll.outdiv << WM8994_FLL1_OUTDIV_SHIFT) | (fll.fll_fratio << WM8994_FLL1_FRATIO_SHIFT); snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_2 + reg_offset, @@ -2151,6 +2164,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, fll.n << WM8994_FLL1_N_SHIFT); snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, + WM8958_FLL1_BYP | WM8994_FLL1_REFCLK_DIV_MASK | WM8994_FLL1_REFCLK_SRC_MASK, (fll.clk_ref_div << WM8994_FLL1_REFCLK_DIV_SHIFT) | @@ -2213,16 +2227,11 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, } } +out: wm8994->fll[id].in = freq_in; wm8994->fll[id].out = freq_out; wm8994->fll[id].src = src; - /* Enable any gated AIF clocks */ - snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, - WM8994_AIF1CLK_ENA, aif1); - snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, - WM8994_AIF2CLK_ENA, aif2); - configure_clock(codec); return 0; @@ -2290,7 +2299,7 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, case WM8994_SYSCLK_OPCLK: /* Special case - a division (times 10) is given and - * no effect on main clocking. + * no effect on main clocking. */ if (freq) { for (i = 0; i < ARRAY_SIZE(opclk_divs); i++) @@ -2792,33 +2801,6 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream, return snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_WL_MASK, aif1); } -static void wm8994_aif_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - int rate_reg = 0; - - switch (dai->id) { - case 1: - rate_reg = WM8994_AIF1_RATE; - break; - case 2: - rate_reg = WM8994_AIF2_RATE; - break; - default: - break; - } - - /* If the DAI is idle then configure the divider tree for the - * lowest output rate to save a little power if the clock is - * still active (eg, because it is system clock). - */ - if (rate_reg && !dai->playback_active && !dai->capture_active) - snd_soc_update_bits(codec, rate_reg, - WM8994_AIF1_SR_MASK | - WM8994_AIF1CLK_RATE_MASK, 0x9); -} - static int wm8994_aif_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; @@ -2860,10 +2842,6 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate) reg = WM8994_AIF2_MASTER_SLAVE; mask = WM8994_AIF2_TRI; break; - case 3: - reg = WM8994_POWER_MANAGEMENT_6; - mask = WM8994_AIF3_TRI; - break; default: return -EINVAL; } @@ -2900,7 +2878,6 @@ static const struct snd_soc_dai_ops wm8994_aif1_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, - .shutdown = wm8994_aif_shutdown, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, .set_tristate = wm8994_set_tristate, @@ -2910,7 +2887,6 @@ static const struct snd_soc_dai_ops wm8994_aif2_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, - .shutdown = wm8994_aif_shutdown, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, .set_tristate = wm8994_set_tristate, @@ -2918,7 +2894,6 @@ static const struct snd_soc_dai_ops wm8994_aif2_dai_ops = { static const struct snd_soc_dai_ops wm8994_aif3_dai_ops = { .hw_params = wm8994_aif3_hw_params, - .set_tristate = wm8994_set_tristate, }; static struct snd_soc_dai_driver wm8994_dai[] = { @@ -2992,23 +2967,8 @@ static struct snd_soc_dai_driver wm8994_dai[] = { static int wm8994_codec_suspend(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = wm8994->wm8994; int i, ret; - switch (control->type) { - case WM8994: - snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0); - break; - case WM1811: - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM1811_JACKDET_MODE_MASK, 0); - /* Fall through */ - case WM8958: - snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, - WM8958_MICD_ENA, 0); - break; - } - for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i], sizeof(struct wm8994_fll_config)); @@ -3058,28 +3018,6 @@ static int wm8994_codec_resume(struct snd_soc_codec *codec) i + 1, ret); } - switch (control->type) { - case WM8994: - if (wm8994->micdet[0].jack || wm8994->micdet[1].jack) - snd_soc_update_bits(codec, WM8994_MICBIAS, - WM8994_MICD_ENA, WM8994_MICD_ENA); - break; - case WM1811: - if (wm8994->jackdet && wm8994->jack_cb) { - /* Restart from idle */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM1811_JACKDET_MODE_MASK, - WM1811_JACKDET_MODE_JACK); - break; - } - break; - case WM8958: - if (wm8994->jack_cb) - snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, - WM8958_MICD_ENA, WM8958_MICD_ENA); - break; - } - return 0; } #else @@ -3126,14 +3064,14 @@ static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994) /* Expand the array... */ t = krealloc(wm8994->retune_mobile_texts, - sizeof(char *) * + sizeof(char *) * (wm8994->num_retune_mobile_texts + 1), GFP_KERNEL); if (t == NULL) continue; /* ...store the new entry... */ - t[wm8994->num_retune_mobile_texts] = + t[wm8994->num_retune_mobile_texts] = pdata->retune_mobile_cfgs[i].name; /* ...and remember the new version. */ @@ -3304,25 +3242,25 @@ int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, } EXPORT_SYMBOL_GPL(wm8994_mic_detect); -static irqreturn_t wm8994_mic_irq(int irq, void *data) +static void wm8994_mic_work(struct work_struct *work) { - struct wm8994_priv *priv = data; - struct snd_soc_codec *codec = priv->codec; - int reg; + struct wm8994_priv *priv = container_of(work, + struct wm8994_priv, + mic_work.work); + struct regmap *regmap = priv->wm8994->regmap; + struct device *dev = priv->wm8994->dev; + unsigned int reg; + int ret; int report; -#ifndef CONFIG_SND_SOC_WM8994_MODULE - trace_snd_soc_jack_irq(dev_name(codec->dev)); -#endif - - reg = snd_soc_read(codec, WM8994_INTERRUPT_RAW_STATUS_2); - if (reg < 0) { - dev_err(codec->dev, "Failed to read microphone status: %d\n", - reg); - return IRQ_HANDLED; + ret = regmap_read(regmap, WM8994_INTERRUPT_RAW_STATUS_2, ®); + if (ret < 0) { + dev_err(dev, "Failed to read microphone status: %d\n", + ret); + return; } - dev_dbg(codec->dev, "Microphone status: %x\n", reg); + dev_dbg(dev, "Microphone status: %x\n", reg); report = 0; if (reg & WM8994_MIC1_DET_STS) { @@ -3361,6 +3299,20 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data) snd_soc_jack_report(priv->micdet[1].jack, report, SND_JACK_HEADSET | SND_JACK_BTN_0); +} + +static irqreturn_t wm8994_mic_irq(int irq, void *data) +{ + struct wm8994_priv *priv = data; + struct snd_soc_codec *codec = priv->codec; + +#ifndef CONFIG_SND_SOC_WM8994_MODULE + trace_snd_soc_jack_irq(dev_name(codec->dev)); +#endif + + pm_wakeup_event(codec->dev, 300); + + schedule_delayed_work(&priv->mic_work, msecs_to_jiffies(250)); return IRQ_HANDLED; } @@ -3415,9 +3367,6 @@ static void wm8958_default_micdet(u16 status, void *data) wm8958_micd_set_rate(codec); - snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADPHONE, - SND_JACK_HEADSET); - /* If we have jackdet that will detect removal */ if (wm8994->jackdet) { mutex_lock(&wm8994->accdet_lock); @@ -3430,14 +3379,13 @@ static void wm8958_default_micdet(u16 status, void *data) mutex_unlock(&wm8994->accdet_lock); - if (wm8994->pdata->jd_ext_cap) { - mutex_lock(&codec->mutex); + if (wm8994->pdata->jd_ext_cap) snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2"); - snd_soc_dapm_sync(&codec->dapm); - mutex_unlock(&codec->mutex); - } } + + snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADPHONE, + SND_JACK_HEADSET); } /* Report short circuit as a button */ @@ -3489,6 +3437,8 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) if (present) { dev_dbg(codec->dev, "Jack detected\n"); + wm8958_micd_set_rate(codec); + snd_soc_update_bits(codec, WM8958_MICBIAS2, WM8958_MICB2_DISCH, 0); @@ -3526,16 +3476,11 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) /* If required for an external cap force MICBIAS on */ if (wm8994->pdata->jd_ext_cap) { - mutex_lock(&codec->mutex); - if (present) snd_soc_dapm_force_enable_pin(&codec->dapm, "MICBIAS2"); else snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2"); - - snd_soc_dapm_sync(&codec->dapm); - mutex_unlock(&codec->mutex); } if (present) @@ -3740,15 +3685,13 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->codec = codec; mutex_init(&wm8994->accdet_lock); + INIT_DELAYED_WORK(&wm8994->mic_work, wm8994_mic_work); for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) init_completion(&wm8994->fll_locked[i]); if (wm8994->pdata && wm8994->pdata->micdet_irq) wm8994->micdet_irq = wm8994->pdata->micdet_irq; - else if (wm8994->pdata && wm8994->pdata->irq_base) - wm8994->micdet_irq = wm8994->pdata->irq_base + - WM8994_IRQ_MIC1_DET; pm_runtime_enable(codec->dev); pm_runtime_idle(codec->dev); @@ -3783,13 +3726,22 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8958: wm8994->hubs.dcs_readback_mode = 1; wm8994->hubs.hp_startup_mode = 1; + + switch (wm8994->revision) { + case 0: + break; + default: + wm8994->fll_byp = true; + break; + } break; case WM1811: wm8994->hubs.dcs_readback_mode = 2; wm8994->hubs.no_series_update = 1; wm8994->hubs.hp_startup_mode = 1; - wm8994->hubs.no_cache_class_w = true; + wm8994->hubs.no_cache_dac_hp_direct = true; + wm8994->fll_byp = true; switch (wm8994->revision) { case 0: @@ -3878,6 +3830,10 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) dev_warn(codec->dev, "Failed to request Mic detect IRQ: %d\n", ret); + } else { + wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_MIC1_DET, + wm8958_mic_irq, "Mic detect", + wm8994); } } @@ -3939,39 +3895,11 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) pm_runtime_put(codec->dev); - /* Latch volume updates (right only; we always do left then right). */ - snd_soc_update_bits(codec, WM8994_AIF1_DAC1_LEFT_VOLUME, - WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU); - snd_soc_update_bits(codec, WM8994_AIF1_DAC1_RIGHT_VOLUME, - WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU); - snd_soc_update_bits(codec, WM8994_AIF1_DAC2_LEFT_VOLUME, - WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU); - snd_soc_update_bits(codec, WM8994_AIF1_DAC2_RIGHT_VOLUME, - WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU); - snd_soc_update_bits(codec, WM8994_AIF2_DAC_LEFT_VOLUME, - WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU); - snd_soc_update_bits(codec, WM8994_AIF2_DAC_RIGHT_VOLUME, - WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU); - snd_soc_update_bits(codec, WM8994_AIF1_ADC1_LEFT_VOLUME, - WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU); - snd_soc_update_bits(codec, WM8994_AIF1_ADC1_RIGHT_VOLUME, - WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU); - snd_soc_update_bits(codec, WM8994_AIF1_ADC2_LEFT_VOLUME, - WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU); - snd_soc_update_bits(codec, WM8994_AIF1_ADC2_RIGHT_VOLUME, - WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU); - snd_soc_update_bits(codec, WM8994_AIF2_ADC_LEFT_VOLUME, - WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU); - snd_soc_update_bits(codec, WM8994_AIF2_ADC_RIGHT_VOLUME, - WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU); - snd_soc_update_bits(codec, WM8994_DAC1_LEFT_VOLUME, - WM8994_DAC1_VU, WM8994_DAC1_VU); - snd_soc_update_bits(codec, WM8994_DAC1_RIGHT_VOLUME, - WM8994_DAC1_VU, WM8994_DAC1_VU); - snd_soc_update_bits(codec, WM8994_DAC2_LEFT_VOLUME, - WM8994_DAC2_VU, WM8994_DAC2_VU); - snd_soc_update_bits(codec, WM8994_DAC2_RIGHT_VOLUME, - WM8994_DAC2_VU, WM8994_DAC2_VU); + /* Latch volume update bits */ + for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) + snd_soc_update_bits(codec, wm8994_vu_bits[i].reg, + wm8994_vu_bits[i].mask, + wm8994_vu_bits[i].mask); /* Set the low bit of the 3D stereo depth so TLV matches */ snd_soc_update_bits(codec, WM8994_AIF1_DAC1_FILTERS_2, @@ -4010,7 +3938,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; } - wm8994_update_class_w(codec); + wm8994->hubs.check_class_w_digital = wm8994_check_class_w_digital; + wm_hubs_update_class_w(codec); wm8994_handle_pdata(wm8994); @@ -4075,7 +4004,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm8994_dac_widgets)); break; } - wm_hubs_add_analogue_routes(codec, 0, 0); snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); @@ -4140,7 +4068,7 @@ err_irq: return ret; } -static int wm8994_codec_remove(struct snd_soc_codec *codec) +static int wm8994_codec_remove(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; @@ -4181,14 +4109,10 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) free_irq(wm8994->micdet_irq, wm8994); break; } - if (wm8994->mbc) - release_firmware(wm8994->mbc); - if (wm8994->mbc_vss) - release_firmware(wm8994->mbc_vss); - if (wm8994->enh_eq) - release_firmware(wm8994->enh_eq); + release_firmware(wm8994->mbc); + release_firmware(wm8994->mbc_vss); + release_firmware(wm8994->enh_eq); kfree(wm8994->retune_mobile_texts); - return 0; } diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index c724112998d8..d77e06f0a675 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -12,6 +12,7 @@ #include <sound/soc.h> #include <linux/firmware.h> #include <linux/completion.h> +#include <linux/workqueue.h> #include "wm_hubs.h" @@ -79,6 +80,7 @@ struct wm8994_priv { struct wm8994_fll_config fll[2], fll_suspend[2]; struct completion fll_locked[2]; bool fll_locked_irq; + bool fll_byp; int vmid_refcount; int active_refcount; @@ -126,6 +128,7 @@ struct wm8994_priv { struct mutex accdet_lock; struct wm8994_micdet micdet[2]; + struct delayed_work mic_work; bool mic_detecting; bool jack_mic; int btn_mask; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 1fd635494045..00f183dfa454 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1,7 +1,7 @@ /* * wm8996.c - WM8996 audio codec interface * - * Copyright 2011 Wolfson Microelectronics PLC. + * Copyright 2011-2 Wolfson Microelectronics PLC. * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * * This program is free software; you can redistribute it and/or modify it @@ -296,184 +296,6 @@ static struct reg_default wm8996_reg[] = { { WM8996_RIGHT_PDM_SPEAKER, 0x1 }, { WM8996_PDM_SPEAKER_MUTE_SEQUENCE, 0x69 }, { WM8996_PDM_SPEAKER_VOLUME, 0x66 }, - { WM8996_WRITE_SEQUENCER_0, 0x1 }, - { WM8996_WRITE_SEQUENCER_1, 0x1 }, - { WM8996_WRITE_SEQUENCER_3, 0x6 }, - { WM8996_WRITE_SEQUENCER_4, 0x40 }, - { WM8996_WRITE_SEQUENCER_5, 0x1 }, - { WM8996_WRITE_SEQUENCER_6, 0xf }, - { WM8996_WRITE_SEQUENCER_7, 0x6 }, - { WM8996_WRITE_SEQUENCER_8, 0x1 }, - { WM8996_WRITE_SEQUENCER_9, 0x3 }, - { WM8996_WRITE_SEQUENCER_10, 0x104 }, - { WM8996_WRITE_SEQUENCER_12, 0x60 }, - { WM8996_WRITE_SEQUENCER_13, 0x11 }, - { WM8996_WRITE_SEQUENCER_14, 0x401 }, - { WM8996_WRITE_SEQUENCER_16, 0x50 }, - { WM8996_WRITE_SEQUENCER_17, 0x3 }, - { WM8996_WRITE_SEQUENCER_18, 0x100 }, - { WM8996_WRITE_SEQUENCER_20, 0x51 }, - { WM8996_WRITE_SEQUENCER_21, 0x3 }, - { WM8996_WRITE_SEQUENCER_22, 0x104 }, - { WM8996_WRITE_SEQUENCER_23, 0xa }, - { WM8996_WRITE_SEQUENCER_24, 0x60 }, - { WM8996_WRITE_SEQUENCER_25, 0x3b }, - { WM8996_WRITE_SEQUENCER_26, 0x502 }, - { WM8996_WRITE_SEQUENCER_27, 0x100 }, - { WM8996_WRITE_SEQUENCER_28, 0x2fff }, - { WM8996_WRITE_SEQUENCER_32, 0x2fff }, - { WM8996_WRITE_SEQUENCER_36, 0x2fff }, - { WM8996_WRITE_SEQUENCER_40, 0x2fff }, - { WM8996_WRITE_SEQUENCER_44, 0x2fff }, - { WM8996_WRITE_SEQUENCER_48, 0x2fff }, - { WM8996_WRITE_SEQUENCER_52, 0x2fff }, - { WM8996_WRITE_SEQUENCER_56, 0x2fff }, - { WM8996_WRITE_SEQUENCER_60, 0x2fff }, - { WM8996_WRITE_SEQUENCER_64, 0x1 }, - { WM8996_WRITE_SEQUENCER_65, 0x1 }, - { WM8996_WRITE_SEQUENCER_67, 0x6 }, - { WM8996_WRITE_SEQUENCER_68, 0x40 }, - { WM8996_WRITE_SEQUENCER_69, 0x1 }, - { WM8996_WRITE_SEQUENCER_70, 0xf }, - { WM8996_WRITE_SEQUENCER_71, 0x6 }, - { WM8996_WRITE_SEQUENCER_72, 0x1 }, - { WM8996_WRITE_SEQUENCER_73, 0x3 }, - { WM8996_WRITE_SEQUENCER_74, 0x104 }, - { WM8996_WRITE_SEQUENCER_76, 0x60 }, - { WM8996_WRITE_SEQUENCER_77, 0x11 }, - { WM8996_WRITE_SEQUENCER_78, 0x401 }, - { WM8996_WRITE_SEQUENCER_80, 0x50 }, - { WM8996_WRITE_SEQUENCER_81, 0x3 }, - { WM8996_WRITE_SEQUENCER_82, 0x100 }, - { WM8996_WRITE_SEQUENCER_84, 0x60 }, - { WM8996_WRITE_SEQUENCER_85, 0x3b }, - { WM8996_WRITE_SEQUENCER_86, 0x502 }, - { WM8996_WRITE_SEQUENCER_87, 0x100 }, - { WM8996_WRITE_SEQUENCER_88, 0x2fff }, - { WM8996_WRITE_SEQUENCER_92, 0x2fff }, - { WM8996_WRITE_SEQUENCER_96, 0x2fff }, - { WM8996_WRITE_SEQUENCER_100, 0x2fff }, - { WM8996_WRITE_SEQUENCER_104, 0x2fff }, - { WM8996_WRITE_SEQUENCER_108, 0x2fff }, - { WM8996_WRITE_SEQUENCER_112, 0x2fff }, - { WM8996_WRITE_SEQUENCER_116, 0x2fff }, - { WM8996_WRITE_SEQUENCER_120, 0x2fff }, - { WM8996_WRITE_SEQUENCER_124, 0x2fff }, - { WM8996_WRITE_SEQUENCER_128, 0x1 }, - { WM8996_WRITE_SEQUENCER_129, 0x1 }, - { WM8996_WRITE_SEQUENCER_131, 0x6 }, - { WM8996_WRITE_SEQUENCER_132, 0x40 }, - { WM8996_WRITE_SEQUENCER_133, 0x1 }, - { WM8996_WRITE_SEQUENCER_134, 0xf }, - { WM8996_WRITE_SEQUENCER_135, 0x6 }, - { WM8996_WRITE_SEQUENCER_136, 0x1 }, - { WM8996_WRITE_SEQUENCER_137, 0x3 }, - { WM8996_WRITE_SEQUENCER_138, 0x106 }, - { WM8996_WRITE_SEQUENCER_140, 0x61 }, - { WM8996_WRITE_SEQUENCER_141, 0x11 }, - { WM8996_WRITE_SEQUENCER_142, 0x401 }, - { WM8996_WRITE_SEQUENCER_144, 0x50 }, - { WM8996_WRITE_SEQUENCER_145, 0x3 }, - { WM8996_WRITE_SEQUENCER_146, 0x102 }, - { WM8996_WRITE_SEQUENCER_148, 0x51 }, - { WM8996_WRITE_SEQUENCER_149, 0x3 }, - { WM8996_WRITE_SEQUENCER_150, 0x106 }, - { WM8996_WRITE_SEQUENCER_151, 0xa }, - { WM8996_WRITE_SEQUENCER_152, 0x61 }, - { WM8996_WRITE_SEQUENCER_153, 0x3b }, - { WM8996_WRITE_SEQUENCER_154, 0x502 }, - { WM8996_WRITE_SEQUENCER_155, 0x100 }, - { WM8996_WRITE_SEQUENCER_156, 0x2fff }, - { WM8996_WRITE_SEQUENCER_160, 0x2fff }, - { WM8996_WRITE_SEQUENCER_164, 0x2fff }, - { WM8996_WRITE_SEQUENCER_168, 0x2fff }, - { WM8996_WRITE_SEQUENCER_172, 0x2fff }, - { WM8996_WRITE_SEQUENCER_176, 0x2fff }, - { WM8996_WRITE_SEQUENCER_180, 0x2fff }, - { WM8996_WRITE_SEQUENCER_184, 0x2fff }, - { WM8996_WRITE_SEQUENCER_188, 0x2fff }, - { WM8996_WRITE_SEQUENCER_192, 0x1 }, - { WM8996_WRITE_SEQUENCER_193, 0x1 }, - { WM8996_WRITE_SEQUENCER_195, 0x6 }, - { WM8996_WRITE_SEQUENCER_196, 0x40 }, - { WM8996_WRITE_SEQUENCER_197, 0x1 }, - { WM8996_WRITE_SEQUENCER_198, 0xf }, - { WM8996_WRITE_SEQUENCER_199, 0x6 }, - { WM8996_WRITE_SEQUENCER_200, 0x1 }, - { WM8996_WRITE_SEQUENCER_201, 0x3 }, - { WM8996_WRITE_SEQUENCER_202, 0x106 }, - { WM8996_WRITE_SEQUENCER_204, 0x61 }, - { WM8996_WRITE_SEQUENCER_205, 0x11 }, - { WM8996_WRITE_SEQUENCER_206, 0x401 }, - { WM8996_WRITE_SEQUENCER_208, 0x50 }, - { WM8996_WRITE_SEQUENCER_209, 0x3 }, - { WM8996_WRITE_SEQUENCER_210, 0x102 }, - { WM8996_WRITE_SEQUENCER_212, 0x61 }, - { WM8996_WRITE_SEQUENCER_213, 0x3b }, - { WM8996_WRITE_SEQUENCER_214, 0x502 }, - { WM8996_WRITE_SEQUENCER_215, 0x100 }, - { WM8996_WRITE_SEQUENCER_216, 0x2fff }, - { WM8996_WRITE_SEQUENCER_220, 0x2fff }, - { WM8996_WRITE_SEQUENCER_224, 0x2fff }, - { WM8996_WRITE_SEQUENCER_228, 0x2fff }, - { WM8996_WRITE_SEQUENCER_232, 0x2fff }, - { WM8996_WRITE_SEQUENCER_236, 0x2fff }, - { WM8996_WRITE_SEQUENCER_240, 0x2fff }, - { WM8996_WRITE_SEQUENCER_244, 0x2fff }, - { WM8996_WRITE_SEQUENCER_248, 0x2fff }, - { WM8996_WRITE_SEQUENCER_252, 0x2fff }, - { WM8996_WRITE_SEQUENCER_256, 0x60 }, - { WM8996_WRITE_SEQUENCER_258, 0x601 }, - { WM8996_WRITE_SEQUENCER_260, 0x50 }, - { WM8996_WRITE_SEQUENCER_262, 0x100 }, - { WM8996_WRITE_SEQUENCER_264, 0x1 }, - { WM8996_WRITE_SEQUENCER_266, 0x104 }, - { WM8996_WRITE_SEQUENCER_267, 0x100 }, - { WM8996_WRITE_SEQUENCER_268, 0x2fff }, - { WM8996_WRITE_SEQUENCER_272, 0x2fff }, - { WM8996_WRITE_SEQUENCER_276, 0x2fff }, - { WM8996_WRITE_SEQUENCER_280, 0x2fff }, - { WM8996_WRITE_SEQUENCER_284, 0x2fff }, - { WM8996_WRITE_SEQUENCER_288, 0x2fff }, - { WM8996_WRITE_SEQUENCER_292, 0x2fff }, - { WM8996_WRITE_SEQUENCER_296, 0x2fff }, - { WM8996_WRITE_SEQUENCER_300, 0x2fff }, - { WM8996_WRITE_SEQUENCER_304, 0x2fff }, - { WM8996_WRITE_SEQUENCER_308, 0x2fff }, - { WM8996_WRITE_SEQUENCER_312, 0x2fff }, - { WM8996_WRITE_SEQUENCER_316, 0x2fff }, - { WM8996_WRITE_SEQUENCER_320, 0x61 }, - { WM8996_WRITE_SEQUENCER_322, 0x601 }, - { WM8996_WRITE_SEQUENCER_324, 0x50 }, - { WM8996_WRITE_SEQUENCER_326, 0x102 }, - { WM8996_WRITE_SEQUENCER_328, 0x1 }, - { WM8996_WRITE_SEQUENCER_330, 0x106 }, - { WM8996_WRITE_SEQUENCER_331, 0x100 }, - { WM8996_WRITE_SEQUENCER_332, 0x2fff }, - { WM8996_WRITE_SEQUENCER_336, 0x2fff }, - { WM8996_WRITE_SEQUENCER_340, 0x2fff }, - { WM8996_WRITE_SEQUENCER_344, 0x2fff }, - { WM8996_WRITE_SEQUENCER_348, 0x2fff }, - { WM8996_WRITE_SEQUENCER_352, 0x2fff }, - { WM8996_WRITE_SEQUENCER_356, 0x2fff }, - { WM8996_WRITE_SEQUENCER_360, 0x2fff }, - { WM8996_WRITE_SEQUENCER_364, 0x2fff }, - { WM8996_WRITE_SEQUENCER_368, 0x2fff }, - { WM8996_WRITE_SEQUENCER_372, 0x2fff }, - { WM8996_WRITE_SEQUENCER_376, 0x2fff }, - { WM8996_WRITE_SEQUENCER_380, 0x2fff }, - { WM8996_WRITE_SEQUENCER_384, 0x60 }, - { WM8996_WRITE_SEQUENCER_386, 0x601 }, - { WM8996_WRITE_SEQUENCER_388, 0x61 }, - { WM8996_WRITE_SEQUENCER_390, 0x601 }, - { WM8996_WRITE_SEQUENCER_392, 0x50 }, - { WM8996_WRITE_SEQUENCER_394, 0x300 }, - { WM8996_WRITE_SEQUENCER_396, 0x1 }, - { WM8996_WRITE_SEQUENCER_398, 0x304 }, - { WM8996_WRITE_SEQUENCER_400, 0x40 }, - { WM8996_WRITE_SEQUENCER_402, 0xf }, - { WM8996_WRITE_SEQUENCER_404, 0x1 }, - { WM8996_WRITE_SEQUENCER_407, 0x100 }, }; static const DECLARE_TLV_DB_SCALE(inpga_tlv, 0, 100, 0); @@ -1706,18 +1528,6 @@ static bool wm8996_volatile_register(struct device *dev, unsigned int reg) } } -static int wm8996_reset(struct wm8996_priv *wm8996) -{ - if (wm8996->pdata.ldo_ena > 0) { - gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); - gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 1); - return 0; - } else { - return regmap_write(wm8996->regmap, WM8996_SOFTWARE_RESET, - 0x8915); - } -} - static const int bclk_divs[] = { 1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96 }; @@ -1770,7 +1580,13 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + break; case SND_SOC_BIAS_PREPARE: + /* Put the MICBIASes into regulating mode */ + snd_soc_update_bits(codec, WM8996_MICBIAS_1, + WM8996_MICB1_MODE, 0); + snd_soc_update_bits(codec, WM8996_MICBIAS_2, + WM8996_MICB2_MODE, 0); break; case SND_SOC_BIAS_STANDBY: @@ -1793,12 +1609,20 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, regcache_cache_only(codec->control_data, false); regcache_sync(codec->control_data); } + + /* Bypass the MICBIASes for lowest power */ + snd_soc_update_bits(codec, WM8996_MICBIAS_1, + WM8996_MICB1_MODE, WM8996_MICB1_MODE); + snd_soc_update_bits(codec, WM8996_MICBIAS_2, + WM8996_MICB2_MODE, WM8996_MICB2_MODE); break; case SND_SOC_BIAS_OFF: regcache_cache_only(codec->control_data, true); - if (wm8996->pdata.ldo_ena >= 0) + if (wm8996->pdata.ldo_ena >= 0) { gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); + regcache_cache_only(codec->control_data, true); + } regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); break; @@ -2795,7 +2619,7 @@ static int wm8996_probe(struct snd_soc_codec *codec) int ret; struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); struct i2c_client *i2c = to_i2c_client(codec->dev); - int i, irq_flags; + int irq_flags; wm8996->codec = codec; @@ -2810,179 +2634,12 @@ static int wm8996_probe(struct snd_soc_codec *codec) goto err; } - wm8996->disable_nb[0].notifier_call = wm8996_regulator_event_0; - wm8996->disable_nb[1].notifier_call = wm8996_regulator_event_1; - wm8996->disable_nb[2].notifier_call = wm8996_regulator_event_2; - - /* This should really be moved into the regulator core */ - for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) { - ret = regulator_register_notifier(wm8996->supplies[i].consumer, - &wm8996->disable_nb[i]); - if (ret != 0) { - dev_err(codec->dev, - "Failed to register regulator notifier: %d\n", - ret); - } - } - - regcache_cache_only(codec->control_data, true); - - /* Apply platform data settings */ - snd_soc_update_bits(codec, WM8996_LINE_INPUT_CONTROL, - WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK, - wm8996->pdata.inl_mode << WM8996_INL_MODE_SHIFT | - wm8996->pdata.inr_mode); - - for (i = 0; i < ARRAY_SIZE(wm8996->pdata.gpio_default); i++) { - if (!wm8996->pdata.gpio_default[i]) - continue; - - snd_soc_write(codec, WM8996_GPIO_1 + i, - wm8996->pdata.gpio_default[i] & 0xffff); - } - - if (wm8996->pdata.spkmute_seq) - snd_soc_update_bits(codec, WM8996_PDM_SPEAKER_MUTE_SEQUENCE, - WM8996_SPK_MUTE_ENDIAN | - WM8996_SPK_MUTE_SEQ1_MASK, - wm8996->pdata.spkmute_seq); - - snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_2, - WM8996_MICD_BIAS_SRC | WM8996_HPOUT1FB_SRC | - WM8996_MICD_SRC, wm8996->pdata.micdet_def); - - /* Latch volume update bits */ - snd_soc_update_bits(codec, WM8996_LEFT_LINE_INPUT_VOLUME, - WM8996_IN1_VU, WM8996_IN1_VU); - snd_soc_update_bits(codec, WM8996_RIGHT_LINE_INPUT_VOLUME, - WM8996_IN1_VU, WM8996_IN1_VU); - - snd_soc_update_bits(codec, WM8996_DAC1_LEFT_VOLUME, - WM8996_DAC1_VU, WM8996_DAC1_VU); - snd_soc_update_bits(codec, WM8996_DAC1_RIGHT_VOLUME, - WM8996_DAC1_VU, WM8996_DAC1_VU); - snd_soc_update_bits(codec, WM8996_DAC2_LEFT_VOLUME, - WM8996_DAC2_VU, WM8996_DAC2_VU); - snd_soc_update_bits(codec, WM8996_DAC2_RIGHT_VOLUME, - WM8996_DAC2_VU, WM8996_DAC2_VU); - - snd_soc_update_bits(codec, WM8996_OUTPUT1_LEFT_VOLUME, - WM8996_DAC1_VU, WM8996_DAC1_VU); - snd_soc_update_bits(codec, WM8996_OUTPUT1_RIGHT_VOLUME, - WM8996_DAC1_VU, WM8996_DAC1_VU); - snd_soc_update_bits(codec, WM8996_OUTPUT2_LEFT_VOLUME, - WM8996_DAC2_VU, WM8996_DAC2_VU); - snd_soc_update_bits(codec, WM8996_OUTPUT2_RIGHT_VOLUME, - WM8996_DAC2_VU, WM8996_DAC2_VU); - - snd_soc_update_bits(codec, WM8996_DSP1_TX_LEFT_VOLUME, - WM8996_DSP1TX_VU, WM8996_DSP1TX_VU); - snd_soc_update_bits(codec, WM8996_DSP1_TX_RIGHT_VOLUME, - WM8996_DSP1TX_VU, WM8996_DSP1TX_VU); - snd_soc_update_bits(codec, WM8996_DSP2_TX_LEFT_VOLUME, - WM8996_DSP2TX_VU, WM8996_DSP2TX_VU); - snd_soc_update_bits(codec, WM8996_DSP2_TX_RIGHT_VOLUME, - WM8996_DSP2TX_VU, WM8996_DSP2TX_VU); - - snd_soc_update_bits(codec, WM8996_DSP1_RX_LEFT_VOLUME, - WM8996_DSP1RX_VU, WM8996_DSP1RX_VU); - snd_soc_update_bits(codec, WM8996_DSP1_RX_RIGHT_VOLUME, - WM8996_DSP1RX_VU, WM8996_DSP1RX_VU); - snd_soc_update_bits(codec, WM8996_DSP2_RX_LEFT_VOLUME, - WM8996_DSP2RX_VU, WM8996_DSP2RX_VU); - snd_soc_update_bits(codec, WM8996_DSP2_RX_RIGHT_VOLUME, - WM8996_DSP2RX_VU, WM8996_DSP2RX_VU); - - /* No support currently for the underclocked TDM modes and - * pick a default TDM layout with each channel pair working with - * slots 0 and 1. */ - snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_0_CONFIGURATION, - WM8996_AIF1RX_CHAN0_SLOTS_MASK | - WM8996_AIF1RX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1RX_CHAN0_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_1_CONFIGURATION, - WM8996_AIF1RX_CHAN1_SLOTS_MASK | - WM8996_AIF1RX_CHAN1_START_SLOT_MASK, - 1 << WM8996_AIF1RX_CHAN1_SLOTS_SHIFT | 1); - snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_2_CONFIGURATION, - WM8996_AIF1RX_CHAN2_SLOTS_MASK | - WM8996_AIF1RX_CHAN2_START_SLOT_MASK, - 1 << WM8996_AIF1RX_CHAN2_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_3_CONFIGURATION, - WM8996_AIF1RX_CHAN3_SLOTS_MASK | - WM8996_AIF1RX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1RX_CHAN3_SLOTS_SHIFT | 1); - snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_4_CONFIGURATION, - WM8996_AIF1RX_CHAN4_SLOTS_MASK | - WM8996_AIF1RX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1RX_CHAN4_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_5_CONFIGURATION, - WM8996_AIF1RX_CHAN5_SLOTS_MASK | - WM8996_AIF1RX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1RX_CHAN5_SLOTS_SHIFT | 1); - - snd_soc_update_bits(codec, WM8996_AIF2RX_CHANNEL_0_CONFIGURATION, - WM8996_AIF2RX_CHAN0_SLOTS_MASK | - WM8996_AIF2RX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF2RX_CHAN0_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF2RX_CHANNEL_1_CONFIGURATION, - WM8996_AIF2RX_CHAN1_SLOTS_MASK | - WM8996_AIF2RX_CHAN1_START_SLOT_MASK, - 1 << WM8996_AIF2RX_CHAN1_SLOTS_SHIFT | 1); - - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_0_CONFIGURATION, - WM8996_AIF1TX_CHAN0_SLOTS_MASK | - WM8996_AIF1TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN0_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_1_CONFIGURATION, - WM8996_AIF1TX_CHAN1_SLOTS_MASK | - WM8996_AIF1TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1); - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_2_CONFIGURATION, - WM8996_AIF1TX_CHAN2_SLOTS_MASK | - WM8996_AIF1TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN2_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_3_CONFIGURATION, - WM8996_AIF1TX_CHAN3_SLOTS_MASK | - WM8996_AIF1TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN3_SLOTS_SHIFT | 1); - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_4_CONFIGURATION, - WM8996_AIF1TX_CHAN4_SLOTS_MASK | - WM8996_AIF1TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN4_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_5_CONFIGURATION, - WM8996_AIF1TX_CHAN5_SLOTS_MASK | - WM8996_AIF1TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN5_SLOTS_SHIFT | 1); - - snd_soc_update_bits(codec, WM8996_AIF2TX_CHANNEL_0_CONFIGURATION, - WM8996_AIF2TX_CHAN0_SLOTS_MASK | - WM8996_AIF2TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF2TX_CHAN0_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_1_CONFIGURATION, - WM8996_AIF2TX_CHAN1_SLOTS_MASK | - WM8996_AIF2TX_CHAN1_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1); - if (wm8996->pdata.num_retune_mobile_cfgs) wm8996_retune_mobile_pdata(codec); else snd_soc_add_codec_controls(codec, wm8996_eq_controls, ARRAY_SIZE(wm8996_eq_controls)); - /* If the TX LRCLK pins are not in LRCLK mode configure the - * AIFs to source their clocks from the RX LRCLKs. - */ - if ((snd_soc_read(codec, WM8996_GPIO_1))) - snd_soc_update_bits(codec, WM8996_AIF1_TX_LRCLK_2, - WM8996_AIF1TX_LRCLK_MODE, - WM8996_AIF1TX_LRCLK_MODE); - - if ((snd_soc_read(codec, WM8996_GPIO_2))) - snd_soc_update_bits(codec, WM8996_AIF2_TX_LRCLK_2, - WM8996_AIF2TX_LRCLK_MODE, - WM8996_AIF2TX_LRCLK_MODE); - if (i2c->irq) { if (wm8996->pdata.irq_flags) irq_flags = wm8996->pdata.irq_flags; @@ -3026,9 +2683,7 @@ err: static int wm8996_remove(struct snd_soc_codec *codec) { - struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); struct i2c_client *i2c = to_i2c_client(codec->dev); - int i; snd_soc_update_bits(codec, WM8996_INTERRUPT_CONTROL, WM8996_IM_IRQ, WM8996_IM_IRQ); @@ -3036,11 +2691,6 @@ static int wm8996_remove(struct snd_soc_codec *codec) if (i2c->irq) free_irq(i2c->irq, codec); - for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) - regulator_unregister_notifier(wm8996->supplies[i].consumer, - &wm8996->disable_nb[i]); - regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); - return 0; } @@ -3154,6 +2804,21 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, goto err_gpio; } + wm8996->disable_nb[0].notifier_call = wm8996_regulator_event_0; + wm8996->disable_nb[1].notifier_call = wm8996_regulator_event_1; + wm8996->disable_nb[2].notifier_call = wm8996_regulator_event_2; + + /* This should really be moved into the regulator core */ + for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) { + ret = regulator_register_notifier(wm8996->supplies[i].consumer, + &wm8996->disable_nb[i]); + if (ret != 0) { + dev_err(&i2c->dev, + "Failed to register regulator notifier: %d\n", + ret); + } + } + ret = regulator_bulk_enable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); if (ret != 0) { @@ -3166,7 +2831,7 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, msleep(5); } - wm8996->regmap = regmap_init_i2c(i2c, &wm8996_regmap); + wm8996->regmap = devm_regmap_init_i2c(i2c, &wm8996_regmap); if (IS_ERR(wm8996->regmap)) { ret = PTR_ERR(wm8996->regmap); dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); @@ -3194,14 +2859,199 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, dev_info(&i2c->dev, "revision %c\n", (reg & WM8996_CHIP_REV_MASK) + 'A'); + if (wm8996->pdata.ldo_ena > 0) { + gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); + regcache_cache_only(wm8996->regmap, true); + } else { + ret = regmap_write(wm8996->regmap, WM8996_SOFTWARE_RESET, + 0x8915); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret); + goto err_regmap; + } + } + regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); - ret = wm8996_reset(wm8996); - if (ret < 0) { - dev_err(&i2c->dev, "Failed to issue reset\n"); + /* Apply platform data settings */ + regmap_update_bits(wm8996->regmap, WM8996_LINE_INPUT_CONTROL, + WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK, + wm8996->pdata.inl_mode << WM8996_INL_MODE_SHIFT | + wm8996->pdata.inr_mode); + + for (i = 0; i < ARRAY_SIZE(wm8996->pdata.gpio_default); i++) { + if (!wm8996->pdata.gpio_default[i]) + continue; + + regmap_write(wm8996->regmap, WM8996_GPIO_1 + i, + wm8996->pdata.gpio_default[i] & 0xffff); + } + + if (wm8996->pdata.spkmute_seq) + regmap_update_bits(wm8996->regmap, + WM8996_PDM_SPEAKER_MUTE_SEQUENCE, + WM8996_SPK_MUTE_ENDIAN | + WM8996_SPK_MUTE_SEQ1_MASK, + wm8996->pdata.spkmute_seq); + + regmap_update_bits(wm8996->regmap, WM8996_ACCESSORY_DETECT_MODE_2, + WM8996_MICD_BIAS_SRC | WM8996_HPOUT1FB_SRC | + WM8996_MICD_SRC, wm8996->pdata.micdet_def); + + /* Latch volume update bits */ + regmap_update_bits(wm8996->regmap, WM8996_LEFT_LINE_INPUT_VOLUME, + WM8996_IN1_VU, WM8996_IN1_VU); + regmap_update_bits(wm8996->regmap, WM8996_RIGHT_LINE_INPUT_VOLUME, + WM8996_IN1_VU, WM8996_IN1_VU); + + regmap_update_bits(wm8996->regmap, WM8996_DAC1_LEFT_VOLUME, + WM8996_DAC1_VU, WM8996_DAC1_VU); + regmap_update_bits(wm8996->regmap, WM8996_DAC1_RIGHT_VOLUME, + WM8996_DAC1_VU, WM8996_DAC1_VU); + regmap_update_bits(wm8996->regmap, WM8996_DAC2_LEFT_VOLUME, + WM8996_DAC2_VU, WM8996_DAC2_VU); + regmap_update_bits(wm8996->regmap, WM8996_DAC2_RIGHT_VOLUME, + WM8996_DAC2_VU, WM8996_DAC2_VU); + + regmap_update_bits(wm8996->regmap, WM8996_OUTPUT1_LEFT_VOLUME, + WM8996_DAC1_VU, WM8996_DAC1_VU); + regmap_update_bits(wm8996->regmap, WM8996_OUTPUT1_RIGHT_VOLUME, + WM8996_DAC1_VU, WM8996_DAC1_VU); + regmap_update_bits(wm8996->regmap, WM8996_OUTPUT2_LEFT_VOLUME, + WM8996_DAC2_VU, WM8996_DAC2_VU); + regmap_update_bits(wm8996->regmap, WM8996_OUTPUT2_RIGHT_VOLUME, + WM8996_DAC2_VU, WM8996_DAC2_VU); + + regmap_update_bits(wm8996->regmap, WM8996_DSP1_TX_LEFT_VOLUME, + WM8996_DSP1TX_VU, WM8996_DSP1TX_VU); + regmap_update_bits(wm8996->regmap, WM8996_DSP1_TX_RIGHT_VOLUME, + WM8996_DSP1TX_VU, WM8996_DSP1TX_VU); + regmap_update_bits(wm8996->regmap, WM8996_DSP2_TX_LEFT_VOLUME, + WM8996_DSP2TX_VU, WM8996_DSP2TX_VU); + regmap_update_bits(wm8996->regmap, WM8996_DSP2_TX_RIGHT_VOLUME, + WM8996_DSP2TX_VU, WM8996_DSP2TX_VU); + + regmap_update_bits(wm8996->regmap, WM8996_DSP1_RX_LEFT_VOLUME, + WM8996_DSP1RX_VU, WM8996_DSP1RX_VU); + regmap_update_bits(wm8996->regmap, WM8996_DSP1_RX_RIGHT_VOLUME, + WM8996_DSP1RX_VU, WM8996_DSP1RX_VU); + regmap_update_bits(wm8996->regmap, WM8996_DSP2_RX_LEFT_VOLUME, + WM8996_DSP2RX_VU, WM8996_DSP2RX_VU); + regmap_update_bits(wm8996->regmap, WM8996_DSP2_RX_RIGHT_VOLUME, + WM8996_DSP2RX_VU, WM8996_DSP2RX_VU); + + /* No support currently for the underclocked TDM modes and + * pick a default TDM layout with each channel pair working with + * slots 0 and 1. */ + regmap_update_bits(wm8996->regmap, + WM8996_AIF1RX_CHANNEL_0_CONFIGURATION, + WM8996_AIF1RX_CHAN0_SLOTS_MASK | + WM8996_AIF1RX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1RX_CHAN0_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1RX_CHANNEL_1_CONFIGURATION, + WM8996_AIF1RX_CHAN1_SLOTS_MASK | + WM8996_AIF1RX_CHAN1_START_SLOT_MASK, + 1 << WM8996_AIF1RX_CHAN1_SLOTS_SHIFT | 1); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1RX_CHANNEL_2_CONFIGURATION, + WM8996_AIF1RX_CHAN2_SLOTS_MASK | + WM8996_AIF1RX_CHAN2_START_SLOT_MASK, + 1 << WM8996_AIF1RX_CHAN2_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1RX_CHANNEL_3_CONFIGURATION, + WM8996_AIF1RX_CHAN3_SLOTS_MASK | + WM8996_AIF1RX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1RX_CHAN3_SLOTS_SHIFT | 1); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1RX_CHANNEL_4_CONFIGURATION, + WM8996_AIF1RX_CHAN4_SLOTS_MASK | + WM8996_AIF1RX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1RX_CHAN4_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1RX_CHANNEL_5_CONFIGURATION, + WM8996_AIF1RX_CHAN5_SLOTS_MASK | + WM8996_AIF1RX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1RX_CHAN5_SLOTS_SHIFT | 1); + + regmap_update_bits(wm8996->regmap, + WM8996_AIF2RX_CHANNEL_0_CONFIGURATION, + WM8996_AIF2RX_CHAN0_SLOTS_MASK | + WM8996_AIF2RX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF2RX_CHAN0_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF2RX_CHANNEL_1_CONFIGURATION, + WM8996_AIF2RX_CHAN1_SLOTS_MASK | + WM8996_AIF2RX_CHAN1_START_SLOT_MASK, + 1 << WM8996_AIF2RX_CHAN1_SLOTS_SHIFT | 1); + + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_0_CONFIGURATION, + WM8996_AIF1TX_CHAN0_SLOTS_MASK | + WM8996_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN0_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_1_CONFIGURATION, + WM8996_AIF1TX_CHAN1_SLOTS_MASK | + WM8996_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_2_CONFIGURATION, + WM8996_AIF1TX_CHAN2_SLOTS_MASK | + WM8996_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN2_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_3_CONFIGURATION, + WM8996_AIF1TX_CHAN3_SLOTS_MASK | + WM8996_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN3_SLOTS_SHIFT | 1); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_4_CONFIGURATION, + WM8996_AIF1TX_CHAN4_SLOTS_MASK | + WM8996_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN4_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_5_CONFIGURATION, + WM8996_AIF1TX_CHAN5_SLOTS_MASK | + WM8996_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN5_SLOTS_SHIFT | 1); + + regmap_update_bits(wm8996->regmap, + WM8996_AIF2TX_CHANNEL_0_CONFIGURATION, + WM8996_AIF2TX_CHAN0_SLOTS_MASK | + WM8996_AIF2TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF2TX_CHAN0_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_1_CONFIGURATION, + WM8996_AIF2TX_CHAN1_SLOTS_MASK | + WM8996_AIF2TX_CHAN1_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1); + + /* If the TX LRCLK pins are not in LRCLK mode configure the + * AIFs to source their clocks from the RX LRCLKs. + */ + ret = regmap_read(wm8996->regmap, WM8996_GPIO_1, ®); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to read GPIO1: %d\n", ret); + goto err_regmap; + } + + if (reg & WM8996_GP1_FN_MASK) + regmap_update_bits(wm8996->regmap, WM8996_AIF1_TX_LRCLK_2, + WM8996_AIF1TX_LRCLK_MODE, + WM8996_AIF1TX_LRCLK_MODE); + + ret = regmap_read(wm8996->regmap, WM8996_GPIO_2, ®); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to read GPIO2: %d\n", ret); goto err_regmap; } + if (reg & WM8996_GP2_FN_MASK) + regmap_update_bits(wm8996->regmap, WM8996_AIF2_TX_LRCLK_2, + WM8996_AIF2TX_LRCLK_MODE, + WM8996_AIF2TX_LRCLK_MODE); + wm8996_init_gpio(wm8996); ret = snd_soc_register_codec(&i2c->dev, @@ -3215,7 +3065,6 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, err_gpiolib: wm8996_free_gpio(wm8996); err_regmap: - regmap_exit(wm8996->regmap); err_enable: if (wm8996->pdata.ldo_ena > 0) gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); @@ -3231,14 +3080,18 @@ err: static __devexit int wm8996_i2c_remove(struct i2c_client *client) { struct wm8996_priv *wm8996 = i2c_get_clientdata(client); + int i; snd_soc_unregister_codec(&client->dev); wm8996_free_gpio(wm8996); - regmap_exit(wm8996->regmap); if (wm8996->pdata.ldo_ena > 0) { gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); gpio_free(wm8996->pdata.ldo_ena); } + for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) + regulator_unregister_notifier(wm8996->supplies[i].consumer, + &wm8996->disable_nb[i]); + return 0; } diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 076c126ed9b1..2de74e1ea225 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -3,7 +3,7 @@ * * Author: Mark Brown * - * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2009-12 Wolfson Microelectronics plc * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -774,7 +774,7 @@ static const struct snd_soc_dapm_widget wm9081_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN1"), SND_SOC_DAPM_INPUT("IN2"), -SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM9081_POWER_MANAGEMENT, 0, 0), +SND_SOC_DAPM_DAC("DAC", NULL, WM9081_POWER_MANAGEMENT, 0, 0), SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0, mixer, ARRAY_SIZE(mixer)), @@ -799,6 +799,7 @@ SND_SOC_DAPM_SUPPLY("TSENSE", WM9081_POWER_MANAGEMENT, 7, 0, NULL, 0), static const struct snd_soc_dapm_route wm9081_audio_paths[] = { { "DAC", NULL, "CLK_SYS" }, { "DAC", NULL, "CLK_DSP" }, + { "DAC", NULL, "AIF" }, { "Mixer", "IN1 Switch", "IN1" }, { "Mixer", "IN2 Switch", "IN2" }, @@ -1252,7 +1253,7 @@ static const struct snd_soc_dai_ops wm9081_dai_ops = { static struct snd_soc_dai_driver wm9081_dai = { .name = "wm9081-hifi", .playback = { - .stream_name = "HiFi Playback", + .stream_name = "AIF", .channels_min = 1, .channels_max = 2, .rates = WM9081_RATES, diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 4b263b6edf13..2c2346fdd637 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -1,7 +1,7 @@ /* * ALSA SoC WM9090 driver * - * Copyright 2009, 2010 Wolfson Microelectronics + * Copyright 2009-12 Wolfson Microelectronics * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index cacc6a86b46f..e8e782a0c78d 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -236,9 +236,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int reg; u16 vra; @@ -250,7 +248,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, else reg = AC97_PCM_LR_ADC_RATE; - return ac97_write(codec, reg, runtime->rate); + return ac97_write(codec, reg, substream->runtime->rate); } #define WM9705_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index b342ae50bcd6..099e6ec32125 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -1,7 +1,7 @@ /* * wm9712.c -- ALSA Soc WM9712 codec support * - * Copyright 2006 Wolfson Microelectronics PLC. + * Copyright 2006-12 Wolfson Microelectronics PLC. * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it @@ -467,11 +467,10 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; int reg; u16 vra; + struct snd_pcm_runtime *runtime = substream->runtime; vra = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); @@ -487,10 +486,9 @@ static int ac97_prepare(struct snd_pcm_substream *substream, static int ac97_aux_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 vra, xsle; + struct snd_pcm_runtime *runtime = substream->runtime; vra = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 2d22cc70d536..3eb19fb71d17 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1,7 +1,7 @@ /* * wm9713.c -- ALSA Soc WM9713 codec support * - * Copyright 2006 Wolfson Microelectronics PLC. + * Copyright 2006-10 Wolfson Microelectronics PLC. * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 6c028c470601..61baa48823cb 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -1,7 +1,7 @@ /* * wm_hubs.c -- WM8993/4 common code * - * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2009-12 Wolfson Microelectronics plc * * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> * @@ -109,12 +109,103 @@ irqreturn_t wm_hubs_dcs_done(int irq, void *data) } EXPORT_SYMBOL_GPL(wm_hubs_dcs_done); +static bool wm_hubs_dac_hp_direct(struct snd_soc_codec *codec) +{ + int reg; + + /* If we're going via the mixer we'll need to do additional checks */ + reg = snd_soc_read(codec, WM8993_OUTPUT_MIXER1); + if (!(reg & WM8993_DACL_TO_HPOUT1L)) { + if (reg & ~WM8993_DACL_TO_MIXOUTL) { + dev_vdbg(codec->dev, "Analogue paths connected: %x\n", + reg & ~WM8993_DACL_TO_HPOUT1L); + return false; + } else { + dev_vdbg(codec->dev, "HPL connected to mixer\n"); + } + } else { + dev_vdbg(codec->dev, "HPL connected to DAC\n"); + } + + reg = snd_soc_read(codec, WM8993_OUTPUT_MIXER2); + if (!(reg & WM8993_DACR_TO_HPOUT1R)) { + if (reg & ~WM8993_DACR_TO_MIXOUTR) { + dev_vdbg(codec->dev, "Analogue paths connected: %x\n", + reg & ~WM8993_DACR_TO_HPOUT1R); + return false; + } else { + dev_vdbg(codec->dev, "HPR connected to mixer\n"); + } + } else { + dev_vdbg(codec->dev, "HPR connected to DAC\n"); + } + + return true; +} + +struct wm_hubs_dcs_cache { + struct list_head list; + unsigned int left; + unsigned int right; + u16 dcs_cfg; +}; + +static bool wm_hubs_dcs_cache_get(struct snd_soc_codec *codec, + struct wm_hubs_dcs_cache **entry) +{ + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + struct wm_hubs_dcs_cache *cache; + unsigned int left, right; + + left = snd_soc_read(codec, WM8993_LEFT_OUTPUT_VOLUME); + left &= WM8993_HPOUT1L_VOL_MASK; + + right = snd_soc_read(codec, WM8993_RIGHT_OUTPUT_VOLUME); + right &= WM8993_HPOUT1R_VOL_MASK; + + list_for_each_entry(cache, &hubs->dcs_cache, list) { + if (cache->left != left || cache->right != right) + continue; + + *entry = cache; + return true; + } + + return false; +} + +static void wm_hubs_dcs_cache_set(struct snd_soc_codec *codec, u16 dcs_cfg) +{ + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + struct wm_hubs_dcs_cache *cache; + + if (hubs->no_cache_dac_hp_direct) + return; + + cache = devm_kzalloc(codec->dev, sizeof(*cache), GFP_KERNEL); + if (!cache) { + dev_err(codec->dev, "Failed to allocate DCS cache entry\n"); + return; + } + + cache->left = snd_soc_read(codec, WM8993_LEFT_OUTPUT_VOLUME); + cache->left &= WM8993_HPOUT1L_VOL_MASK; + + cache->right = snd_soc_read(codec, WM8993_RIGHT_OUTPUT_VOLUME); + cache->right &= WM8993_HPOUT1R_VOL_MASK; + + cache->dcs_cfg = dcs_cfg; + + list_add_tail(&cache->list, &hubs->dcs_cache); +} + /* * Startup calibration of the DC servo */ static void calibrate_dc_servo(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + struct wm_hubs_dcs_cache *cache; s8 offset; u16 reg, reg_l, reg_r, dcs_cfg, dcs_reg; @@ -129,10 +220,11 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) /* If we're using a digital only path and have a previously * callibrated DC servo offset stored then use that. */ - if (hubs->class_w && hubs->class_w_dcs) { - dev_dbg(codec->dev, "Using cached DC servo offset %x\n", - hubs->class_w_dcs); - snd_soc_write(codec, dcs_reg, hubs->class_w_dcs); + if (wm_hubs_dac_hp_direct(codec) && + wm_hubs_dcs_cache_get(codec, &cache)) { + dev_dbg(codec->dev, "Using cached DCS offset %x for %d,%d\n", + cache->dcs_cfg, cache->left, cache->right); + snd_soc_write(codec, dcs_reg, cache->dcs_cfg); wait_for_dc_servo(codec, WM8993_DCS_TRIG_DAC_WR_0 | WM8993_DCS_TRIG_DAC_WR_1); @@ -207,8 +299,8 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) /* Save the callibrated offset if we're in class W mode and * therefore don't have any analogue signal mixed in. */ - if (hubs->class_w && !hubs->no_cache_class_w) - hubs->class_w_dcs = dcs_cfg; + if (wm_hubs_dac_hp_direct(codec)) + wm_hubs_dcs_cache_set(codec, dcs_cfg); } /* @@ -223,9 +315,6 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol, ret = snd_soc_put_volsw(kcontrol, ucontrol); - /* Updating the analogue gains invalidates the DC servo cache */ - hubs->class_w_dcs = 0; - /* If we're applying an offset correction then updating the * callibration would be likely to introduce further offsets. */ if (hubs->dcs_codes_l || hubs->dcs_codes_r || hubs->no_series_update) @@ -530,6 +619,86 @@ static int lineout_event(struct snd_soc_dapm_widget *w, return 0; } +void wm_hubs_update_class_w(struct snd_soc_codec *codec) +{ + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + int enable = WM8993_CP_DYN_V | WM8993_CP_DYN_FREQ; + + if (!wm_hubs_dac_hp_direct(codec)) + enable = false; + + if (hubs->check_class_w_digital && !hubs->check_class_w_digital(codec)) + enable = false; + + dev_vdbg(codec->dev, "Class W %s\n", enable ? "enabled" : "disabled"); + + snd_soc_update_bits(codec, WM8993_CLASS_W_0, + WM8993_CP_DYN_V | WM8993_CP_DYN_FREQ, enable); +} +EXPORT_SYMBOL_GPL(wm_hubs_update_class_w); + +#define WM_HUBS_SINGLE_W(xname, reg, shift, max, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_dapm_get_volsw, .put = class_w_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + +static int class_w_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = widget->codec; + int ret; + + ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); + + wm_hubs_update_class_w(codec); + + return ret; +} + +#define WM_HUBS_ENUM_W(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_enum_double, \ + .get = snd_soc_dapm_get_enum_double, \ + .put = class_w_put_double, \ + .private_value = (unsigned long)&xenum } + +static int class_w_put_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = widget->codec; + int ret; + + ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + + wm_hubs_update_class_w(codec); + + return ret; +} + +static const char *hp_mux_text[] = { + "Mixer", + "DAC", +}; + +static const struct soc_enum hpl_enum = + SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER1, 8, 2, hp_mux_text); + +const struct snd_kcontrol_new wm_hubs_hpl_mux = + WM_HUBS_ENUM_W("Left Headphone Mux", hpl_enum); +EXPORT_SYMBOL_GPL(wm_hubs_hpl_mux); + +static const struct soc_enum hpr_enum = + SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER2, 8, 2, hp_mux_text); + +const struct snd_kcontrol_new wm_hubs_hpr_mux = + WM_HUBS_ENUM_W("Right Headphone Mux", hpr_enum); +EXPORT_SYMBOL_GPL(wm_hubs_hpr_mux); + static const struct snd_kcontrol_new in1l_pga[] = { SOC_DAPM_SINGLE("IN1LP Switch", WM8993_INPUT_MIXER2, 5, 1, 0), SOC_DAPM_SINGLE("IN1LN Switch", WM8993_INPUT_MIXER2, 4, 1, 0), @@ -561,25 +730,25 @@ SOC_DAPM_SINGLE("IN1R Switch", WM8993_INPUT_MIXER4, 5, 1, 0), }; static const struct snd_kcontrol_new left_output_mixer[] = { -SOC_DAPM_SINGLE("Right Input Switch", WM8993_OUTPUT_MIXER1, 7, 1, 0), -SOC_DAPM_SINGLE("Left Input Switch", WM8993_OUTPUT_MIXER1, 6, 1, 0), -SOC_DAPM_SINGLE("IN2RN Switch", WM8993_OUTPUT_MIXER1, 5, 1, 0), -SOC_DAPM_SINGLE("IN2LN Switch", WM8993_OUTPUT_MIXER1, 4, 1, 0), -SOC_DAPM_SINGLE("IN2LP Switch", WM8993_OUTPUT_MIXER1, 1, 1, 0), -SOC_DAPM_SINGLE("IN1R Switch", WM8993_OUTPUT_MIXER1, 3, 1, 0), -SOC_DAPM_SINGLE("IN1L Switch", WM8993_OUTPUT_MIXER1, 2, 1, 0), -SOC_DAPM_SINGLE("DAC Switch", WM8993_OUTPUT_MIXER1, 0, 1, 0), +WM_HUBS_SINGLE_W("Right Input Switch", WM8993_OUTPUT_MIXER1, 7, 1, 0), +WM_HUBS_SINGLE_W("Left Input Switch", WM8993_OUTPUT_MIXER1, 6, 1, 0), +WM_HUBS_SINGLE_W("IN2RN Switch", WM8993_OUTPUT_MIXER1, 5, 1, 0), +WM_HUBS_SINGLE_W("IN2LN Switch", WM8993_OUTPUT_MIXER1, 4, 1, 0), +WM_HUBS_SINGLE_W("IN2LP Switch", WM8993_OUTPUT_MIXER1, 1, 1, 0), +WM_HUBS_SINGLE_W("IN1R Switch", WM8993_OUTPUT_MIXER1, 3, 1, 0), +WM_HUBS_SINGLE_W("IN1L Switch", WM8993_OUTPUT_MIXER1, 2, 1, 0), +WM_HUBS_SINGLE_W("DAC Switch", WM8993_OUTPUT_MIXER1, 0, 1, 0), }; static const struct snd_kcontrol_new right_output_mixer[] = { -SOC_DAPM_SINGLE("Left Input Switch", WM8993_OUTPUT_MIXER2, 7, 1, 0), -SOC_DAPM_SINGLE("Right Input Switch", WM8993_OUTPUT_MIXER2, 6, 1, 0), -SOC_DAPM_SINGLE("IN2LN Switch", WM8993_OUTPUT_MIXER2, 5, 1, 0), -SOC_DAPM_SINGLE("IN2RN Switch", WM8993_OUTPUT_MIXER2, 4, 1, 0), -SOC_DAPM_SINGLE("IN1L Switch", WM8993_OUTPUT_MIXER2, 3, 1, 0), -SOC_DAPM_SINGLE("IN1R Switch", WM8993_OUTPUT_MIXER2, 2, 1, 0), -SOC_DAPM_SINGLE("IN2RP Switch", WM8993_OUTPUT_MIXER2, 1, 1, 0), -SOC_DAPM_SINGLE("DAC Switch", WM8993_OUTPUT_MIXER2, 0, 1, 0), +WM_HUBS_SINGLE_W("Left Input Switch", WM8993_OUTPUT_MIXER2, 7, 1, 0), +WM_HUBS_SINGLE_W("Right Input Switch", WM8993_OUTPUT_MIXER2, 6, 1, 0), +WM_HUBS_SINGLE_W("IN2LN Switch", WM8993_OUTPUT_MIXER2, 5, 1, 0), +WM_HUBS_SINGLE_W("IN2RN Switch", WM8993_OUTPUT_MIXER2, 4, 1, 0), +WM_HUBS_SINGLE_W("IN1L Switch", WM8993_OUTPUT_MIXER2, 3, 1, 0), +WM_HUBS_SINGLE_W("IN1R Switch", WM8993_OUTPUT_MIXER2, 2, 1, 0), +WM_HUBS_SINGLE_W("IN2RP Switch", WM8993_OUTPUT_MIXER2, 1, 1, 0), +WM_HUBS_SINGLE_W("DAC Switch", WM8993_OUTPUT_MIXER2, 0, 1, 0), }; static const struct snd_kcontrol_new earpiece_mixer[] = { @@ -943,6 +1112,7 @@ int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec, struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; + INIT_LIST_HEAD(&hubs->dcs_cache); init_completion(&hubs->dcs_done); snd_soc_dapm_add_routes(dapm, analogue_routes, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 5705276f4943..da2dc899ce6d 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -16,6 +16,8 @@ #include <linux/completion.h> #include <linux/interrupt.h> +#include <linux/list.h> +#include <sound/control.h> struct snd_soc_codec; @@ -30,9 +32,9 @@ struct wm_hubs_data { int series_startup; int no_series_update; - bool no_cache_class_w; - bool class_w; - u16 class_w_dcs; + bool no_cache_dac_hp_direct; + struct list_head dcs_cache; + bool (*check_class_w_digital)(struct snd_soc_codec *); bool lineout1_se; bool lineout1n_ena; @@ -58,5 +60,9 @@ extern irqreturn_t wm_hubs_dcs_done(int irq, void *data); extern void wm_hubs_vmid_ena(struct snd_soc_codec *codec); extern void wm_hubs_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level); +extern void wm_hubs_update_class_w(struct snd_soc_codec *codec); + +extern const struct snd_kcontrol_new wm_hubs_hpl_mux; +extern const struct snd_kcontrol_new wm_hubs_hpr_mux; #endif diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig new file mode 100644 index 000000000000..e334900cf0b8 --- /dev/null +++ b/sound/soc/dwc/Kconfig @@ -0,0 +1,9 @@ +config SND_DESIGNWARE_I2S + tristate "Synopsys I2S Device Driver" + depends on CLKDEV_LOOKUP + help + Say Y or M if you want to add support for I2S driver for + Synopsys desigwnware I2S device. The device supports upto + maximum of 8 channels each for play and record. + + diff --git a/sound/soc/dwc/Makefile b/sound/soc/dwc/Makefile new file mode 100644 index 000000000000..319371f690f4 --- /dev/null +++ b/sound/soc/dwc/Makefile @@ -0,0 +1,3 @@ +# SYNOPSYS Platform Support +obj-$(CONFIG_SND_DESIGNWARE_I2S) += designware_i2s.o + diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c new file mode 100644 index 000000000000..1aa51300c564 --- /dev/null +++ b/sound/soc/dwc/designware_i2s.c @@ -0,0 +1,455 @@ +/* + * ALSA SoC Synopsys I2S Audio Layer + * + * sound/soc/spear/designware_i2s.c + * + * Copyright (C) 2010 ST Microelectronics + * Rajeev Kumar <rajeev-dlh.kumar@st.com> + * + * This file is licensed under the terms of the GNU General Public + * License version 2. This program is licensed "as is" without any + * warranty of any kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <sound/designware_i2s.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +/* common register for all channel */ +#define IER 0x000 +#define IRER 0x004 +#define ITER 0x008 +#define CER 0x00C +#define CCR 0x010 +#define RXFFR 0x014 +#define TXFFR 0x018 + +/* I2STxRxRegisters for all channels */ +#define LRBR_LTHR(x) (0x40 * x + 0x020) +#define RRBR_RTHR(x) (0x40 * x + 0x024) +#define RER(x) (0x40 * x + 0x028) +#define TER(x) (0x40 * x + 0x02C) +#define RCR(x) (0x40 * x + 0x030) +#define TCR(x) (0x40 * x + 0x034) +#define ISR(x) (0x40 * x + 0x038) +#define IMR(x) (0x40 * x + 0x03C) +#define ROR(x) (0x40 * x + 0x040) +#define TOR(x) (0x40 * x + 0x044) +#define RFCR(x) (0x40 * x + 0x048) +#define TFCR(x) (0x40 * x + 0x04C) +#define RFF(x) (0x40 * x + 0x050) +#define TFF(x) (0x40 * x + 0x054) + +/* I2SCOMPRegisters */ +#define I2S_COMP_PARAM_2 0x01F0 +#define I2S_COMP_PARAM_1 0x01F4 +#define I2S_COMP_VERSION 0x01F8 +#define I2S_COMP_TYPE 0x01FC + +#define MAX_CHANNEL_NUM 8 +#define MIN_CHANNEL_NUM 2 + +struct dw_i2s_dev { + void __iomem *i2s_base; + struct clk *clk; + int active; + unsigned int capability; + struct device *dev; + + /* data related to DMA transfers b/w i2s and DMAC */ + struct i2s_dma_data play_dma_data; + struct i2s_dma_data capture_dma_data; + struct i2s_clk_config_data config; + int (*i2s_clk_cfg)(struct i2s_clk_config_data *config); +}; + +static inline void i2s_write_reg(void __iomem *io_base, int reg, u32 val) +{ + writel(val, io_base + reg); +} + +static inline u32 i2s_read_reg(void __iomem *io_base, int reg) +{ + return readl(io_base + reg); +} + +static inline void i2s_disable_channels(struct dw_i2s_dev *dev, u32 stream) +{ + u32 i = 0; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < 4; i++) + i2s_write_reg(dev->i2s_base, TER(i), 0); + } else { + for (i = 0; i < 4; i++) + i2s_write_reg(dev->i2s_base, RER(i), 0); + } +} + +static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) +{ + u32 i = 0; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < 4; i++) + i2s_write_reg(dev->i2s_base, TOR(i), 0); + } else { + for (i = 0; i < 4; i++) + i2s_write_reg(dev->i2s_base, ROR(i), 0); + } +} + +static void i2s_start(struct dw_i2s_dev *dev, + struct snd_pcm_substream *substream) +{ + + i2s_write_reg(dev->i2s_base, IER, 1); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + i2s_write_reg(dev->i2s_base, ITER, 1); + else + i2s_write_reg(dev->i2s_base, IRER, 1); + + i2s_write_reg(dev->i2s_base, CER, 1); +} + +static void i2s_stop(struct dw_i2s_dev *dev, + struct snd_pcm_substream *substream) +{ + u32 i = 0, irq; + + i2s_clear_irqs(dev, substream->stream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + i2s_write_reg(dev->i2s_base, ITER, 0); + + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq | 0x30); + } + } else { + i2s_write_reg(dev->i2s_base, IRER, 0); + + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq | 0x03); + } + } + + if (!dev->active) { + i2s_write_reg(dev->i2s_base, CER, 0); + i2s_write_reg(dev->i2s_base, IER, 0); + } +} + +static int dw_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); + struct i2s_dma_data *dma_data = NULL; + + if (!(dev->capability & DWC_I2S_RECORD) && + (substream->stream == SNDRV_PCM_STREAM_CAPTURE)) + return -EINVAL; + + if (!(dev->capability & DWC_I2S_PLAY) && + (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) + return -EINVAL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = &dev->play_dma_data; + else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + dma_data = &dev->capture_dma_data; + + snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)dma_data); + + return 0; +} + +static int dw_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + struct i2s_clk_config_data *config = &dev->config; + u32 ccr, xfer_resolution, ch_reg, irq; + int ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + config->data_width = 16; + ccr = 0x00; + xfer_resolution = 0x02; + break; + + case SNDRV_PCM_FORMAT_S24_LE: + config->data_width = 24; + ccr = 0x08; + xfer_resolution = 0x04; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + config->data_width = 32; + ccr = 0x10; + xfer_resolution = 0x05; + break; + + default: + dev_err(dev->dev, "designware-i2s: unsuppted PCM fmt"); + return -EINVAL; + } + + config->chan_nr = params_channels(params); + + switch (config->chan_nr) { + case EIGHT_CHANNEL_SUPPORT: + ch_reg = 3; + case SIX_CHANNEL_SUPPORT: + ch_reg = 2; + case FOUR_CHANNEL_SUPPORT: + ch_reg = 1; + case TWO_CHANNEL_SUPPORT: + ch_reg = 0; + break; + default: + dev_err(dev->dev, "channel not supported\n"); + } + + i2s_disable_channels(dev, substream->stream); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + i2s_write_reg(dev->i2s_base, TCR(ch_reg), xfer_resolution); + i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); + irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); + i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); + i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); + } else { + i2s_write_reg(dev->i2s_base, RCR(ch_reg), xfer_resolution); + i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); + irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); + i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03); + i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); + } + + i2s_write_reg(dev->i2s_base, CCR, ccr); + + config->sample_rate = params_rate(params); + + if (!dev->i2s_clk_cfg) + return -EINVAL; + + ret = dev->i2s_clk_cfg(config); + if (ret < 0) { + dev_err(dev->dev, "runtime audio clk config fail\n"); + return ret; + } + + return 0; +} + +static void dw_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int dw_i2s_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + dev->active++; + i2s_start(dev, substream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dev->active--; + i2s_stop(dev, substream); + break; + default: + ret = -EINVAL; + break; + } + return ret; +} + +static struct snd_soc_dai_ops dw_i2s_dai_ops = { + .startup = dw_i2s_startup, + .shutdown = dw_i2s_shutdown, + .hw_params = dw_i2s_hw_params, + .trigger = dw_i2s_trigger, +}; + +#ifdef CONFIG_PM + +static int dw_i2s_suspend(struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + clk_disable(dev->clk); + return 0; +} + +static int dw_i2s_resume(struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + clk_enable(dev->clk); + return 0; +} + +#else +#define dw_i2s_suspend NULL +#define dw_i2s_resume NULL +#endif + +static int dw_i2s_probe(struct platform_device *pdev) +{ + const struct i2s_platform_data *pdata = pdev->dev.platform_data; + struct dw_i2s_dev *dev; + struct resource *res; + int ret; + unsigned int cap; + struct snd_soc_dai_driver *dw_i2s_dai; + + if (!pdata) { + dev_err(&pdev->dev, "Invalid platform data\n"); + return -EINVAL; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "no i2s resource defined\n"); + return -ENODEV; + } + + if (!devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), pdev->name)) { + dev_err(&pdev->dev, "i2s region already claimed\n"); + return -EBUSY; + } + + dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL); + if (!dev) { + dev_warn(&pdev->dev, "kzalloc fail\n"); + return -ENOMEM; + } + + dev->i2s_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + if (!dev->i2s_base) { + dev_err(&pdev->dev, "ioremap fail for i2s_region\n"); + return -ENOMEM; + } + + cap = pdata->cap; + dev->capability = cap; + dev->i2s_clk_cfg = pdata->i2s_clk_cfg; + + /* Set DMA slaves info */ + + dev->play_dma_data.data = pdata->play_dma_data; + dev->capture_dma_data.data = pdata->capture_dma_data; + dev->play_dma_data.addr = res->start + I2S_TXDMA; + dev->capture_dma_data.addr = res->start + I2S_RXDMA; + dev->play_dma_data.max_burst = 16; + dev->capture_dma_data.max_burst = 16; + dev->play_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + dev->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + dev->play_dma_data.filter = pdata->filter; + dev->capture_dma_data.filter = pdata->filter; + + dev->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(dev->clk)) + return PTR_ERR(dev->clk); + + ret = clk_enable(dev->clk); + if (ret < 0) + goto err_clk_put; + + dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL); + if (!dw_i2s_dai) { + dev_err(&pdev->dev, "mem allocation failed for dai driver\n"); + ret = -ENOMEM; + goto err_clk_disable; + } + + if (cap & DWC_I2S_PLAY) { + dev_dbg(&pdev->dev, " SPEAr: play supported\n"); + dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->playback.channels_max = pdata->channel; + dw_i2s_dai->playback.formats = pdata->snd_fmts; + dw_i2s_dai->playback.rates = pdata->snd_rates; + } + + if (cap & DWC_I2S_RECORD) { + dev_dbg(&pdev->dev, "SPEAr: record supported\n"); + dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->capture.channels_max = pdata->channel; + dw_i2s_dai->capture.formats = pdata->snd_fmts; + dw_i2s_dai->capture.rates = pdata->snd_rates; + } + + dw_i2s_dai->ops = &dw_i2s_dai_ops; + dw_i2s_dai->suspend = dw_i2s_suspend; + dw_i2s_dai->resume = dw_i2s_resume; + + dev->dev = &pdev->dev; + dev_set_drvdata(&pdev->dev, dev); + ret = snd_soc_register_dai(&pdev->dev, dw_i2s_dai); + if (ret != 0) { + dev_err(&pdev->dev, "not able to register dai\n"); + goto err_set_drvdata; + } + + return 0; + +err_set_drvdata: + dev_set_drvdata(&pdev->dev, NULL); +err_clk_disable: + clk_disable(dev->clk); +err_clk_put: + clk_put(dev->clk); + return ret; +} + +static int dw_i2s_remove(struct platform_device *pdev) +{ + struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_dai(&pdev->dev); + dev_set_drvdata(&pdev->dev, NULL); + + clk_put(dev->clk); + + return 0; +} + +static struct platform_driver dw_i2s_driver = { + .probe = dw_i2s_probe, + .remove = dw_i2s_remove, + .driver = { + .name = "designware-i2s", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(dw_i2s_driver); + +MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>"); +MODULE_DESCRIPTION("DESIGNWARE I2S SoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:designware_i2s"); diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c index 0678637abd66..bdffab33e160 100644 --- a/sound/soc/ep93xx/ep93xx-ac97.c +++ b/sound/soc/ep93xx/ep93xx-ac97.c @@ -87,17 +87,13 @@ * struct ep93xx_ac97_info - EP93xx AC97 controller info structure * @lock: mutex serializing access to the bus (slot 1 & 2 ops) * @dev: pointer to the platform device dev structure - * @mem: physical memory resource for the registers * @regs: mapped AC97 controller registers - * @irq: AC97 interrupt number * @done: bus ops wait here for an interrupt */ struct ep93xx_ac97_info { struct mutex lock; struct device *dev; - struct resource *mem; void __iomem *regs; - int irq; struct completion done; }; @@ -359,66 +355,50 @@ static struct snd_soc_dai_driver ep93xx_ac97_dai = { static int __devinit ep93xx_ac97_probe(struct platform_device *pdev) { struct ep93xx_ac97_info *info; + struct resource *res; + unsigned int irq; int ret; - info = kzalloc(sizeof(struct ep93xx_ac97_info), GFP_KERNEL); + info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; - dev_set_drvdata(&pdev->dev, info); - - mutex_init(&info->lock); - init_completion(&info->done); - info->dev = &pdev->dev; + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) + return -ENODEV; - info->mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!info->mem) { - ret = -ENXIO; - goto fail_free_info; - } + info->regs = devm_request_and_ioremap(&pdev->dev, res); + if (!info->regs) + return -ENXIO; - info->irq = platform_get_irq(pdev, 0); - if (!info->irq) { - ret = -ENXIO; - goto fail_free_info; - } + irq = platform_get_irq(pdev, 0); + if (!irq) + return -ENODEV; - if (!request_mem_region(info->mem->start, resource_size(info->mem), - pdev->name)) { - ret = -EBUSY; - goto fail_free_info; - } + ret = devm_request_irq(&pdev->dev, irq, ep93xx_ac97_interrupt, + IRQF_TRIGGER_HIGH, pdev->name, info); + if (ret) + goto fail; - info->regs = ioremap(info->mem->start, resource_size(info->mem)); - if (!info->regs) { - ret = -ENOMEM; - goto fail_release_mem; - } + dev_set_drvdata(&pdev->dev, info); - ret = request_irq(info->irq, ep93xx_ac97_interrupt, IRQF_TRIGGER_HIGH, - pdev->name, info); - if (ret) - goto fail_unmap_mem; + mutex_init(&info->lock); + init_completion(&info->done); + info->dev = &pdev->dev; ep93xx_ac97_info = info; platform_set_drvdata(pdev, info); ret = snd_soc_register_dai(&pdev->dev, &ep93xx_ac97_dai); if (ret) - goto fail_free_irq; + goto fail; return 0; -fail_free_irq: +fail: platform_set_drvdata(pdev, NULL); - free_irq(info->irq, info); -fail_unmap_mem: - iounmap(info->regs); -fail_release_mem: - release_mem_region(info->mem->start, resource_size(info->mem)); -fail_free_info: - kfree(info); - + ep93xx_ac97_info = NULL; + dev_set_drvdata(&pdev->dev, NULL); return ret; } @@ -431,11 +411,9 @@ static int __devexit ep93xx_ac97_remove(struct platform_device *pdev) /* disable the AC97 controller */ ep93xx_ac97_write_reg(info, AC97GCR, 0); - free_irq(info->irq, info); - iounmap(info->regs); - release_mem_region(info->mem->start, resource_size(info->mem)); platform_set_drvdata(pdev, NULL); - kfree(info); + ep93xx_ac97_info = NULL; + dev_set_drvdata(&pdev->dev, NULL); return 0; } diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index f7a62348e3fe..8df8f6dc474f 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -63,7 +63,6 @@ struct ep93xx_i2s_info { struct clk *sclk; struct clk *lrclk; struct ep93xx_pcm_dma_params *dma_params; - struct resource *mem; void __iomem *regs; }; @@ -373,38 +372,22 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) struct resource *res; int err; - info = kzalloc(sizeof(struct ep93xx_i2s_info), GFP_KERNEL); - if (!info) { - err = -ENOMEM; - goto fail; - } - - dev_set_drvdata(&pdev->dev, info); - info->dma_params = ep93xx_i2s_dma_params; + info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); + if (!info) + return -ENOMEM; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - err = -ENODEV; - goto fail_free_info; - } + if (!res) + return -ENODEV; - info->mem = request_mem_region(res->start, resource_size(res), - pdev->name); - if (!info->mem) { - err = -EBUSY; - goto fail_free_info; - } - - info->regs = ioremap(info->mem->start, resource_size(info->mem)); - if (!info->regs) { - err = -ENXIO; - goto fail_release_mem; - } + info->regs = devm_request_and_ioremap(&pdev->dev, res); + if (!info->regs) + return -ENXIO; info->mclk = clk_get(&pdev->dev, "mclk"); if (IS_ERR(info->mclk)) { err = PTR_ERR(info->mclk); - goto fail_unmap_mem; + goto fail; } info->sclk = clk_get(&pdev->dev, "sclk"); @@ -419,6 +402,9 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) goto fail_put_sclk; } + dev_set_drvdata(&pdev->dev, info); + info->dma_params = ep93xx_i2s_dma_params; + err = snd_soc_register_dai(&pdev->dev, &ep93xx_i2s_dai); if (err) goto fail_put_lrclk; @@ -426,17 +412,12 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) return 0; fail_put_lrclk: + dev_set_drvdata(&pdev->dev, NULL); clk_put(info->lrclk); fail_put_sclk: clk_put(info->sclk); fail_put_mclk: clk_put(info->mclk); -fail_unmap_mem: - iounmap(info->regs); -fail_release_mem: - release_mem_region(info->mem->start, resource_size(info->mem)); -fail_free_info: - kfree(info); fail: return err; } @@ -446,12 +427,10 @@ static int __devexit ep93xx_i2s_remove(struct platform_device *pdev) struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); + dev_set_drvdata(&pdev->dev, NULL); clk_put(info->lrclk); clk_put(info->sclk); clk_put(info->mclk); - iounmap(info->regs); - release_mem_region(info->mem->start, resource_size(info->mem)); - kfree(info); return 0; } diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index 162dbb74f4cc..4eea98b42bc8 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -136,7 +136,7 @@ static struct snd_pcm_ops ep93xx_pcm_ops = { .hw_params = ep93xx_pcm_hw_params, .hw_free = ep93xx_pcm_hw_free, .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer, + .pointer = snd_dmaengine_pcm_pointer_no_residue, .mmap = ep93xx_pcm_mmap, }; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index d754d34d68a6..d70133086ac3 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,18 +1,31 @@ -config SND_MPC52xx_DMA +config SND_SOC_FSL_SSI tristate -# ASoC platform support for the Freescale PowerPC SOCs that have an SSI and -# an Elo DMA controller, such as the MPC8610 and P1022. You will still need to -# select a platform driver and a codec driver. -config SND_SOC_POWERPC_SSI +config SND_SOC_FSL_UTILS tristate + +menuconfig SND_POWERPC_SOC + tristate "SoC Audio for Freescale PowerPC CPUs" depends on FSL_SOC + help + Say Y or M if you want to add support for codecs attached to + the PowerPC CPUs. + +if SND_POWERPC_SOC + +config SND_MPC52xx_DMA + tristate + +config SND_SOC_POWERPC_DMA + tristate config SND_SOC_MPC8610_HPCD tristate "ALSA SoC support for the Freescale MPC8610 HPCD board" # I2C is necessary for the CS4270 driver depends on MPC8610_HPCD && I2C - select SND_SOC_POWERPC_SSI + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + select SND_SOC_POWERPC_DMA select SND_SOC_CS4270 select SND_SOC_CS4270_VD33_ERRATA default y if MPC8610_HPCD @@ -23,7 +36,9 @@ config SND_SOC_P1022_DS tristate "ALSA SoC support for the Freescale P1022 DS board" # I2C is necessary for the WM8776 driver depends on P1022_DS && I2C - select SND_SOC_POWERPC_SSI + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + select SND_SOC_POWERPC_DMA select SND_SOC_WM8776 default y if P1022_DS help @@ -65,3 +80,103 @@ config SND_MPC52xx_SOC_EFIKA help Say Y if you want to add support for sound on the Efika. +endif # SND_POWERPC_SOC + +menuconfig SND_IMX_SOC + tristate "SoC Audio for Freescale i.MX CPUs" + depends on ARCH_MXC + help + Say Y or M if you want to add support for codecs attached to + the i.MX CPUs. + +if SND_IMX_SOC + +config SND_SOC_IMX_SSI + tristate + +config SND_SOC_IMX_PCM + tristate + +config SND_SOC_IMX_PCM_FIQ + tristate + select FIQ + select SND_SOC_IMX_PCM + +config SND_SOC_IMX_PCM_DMA + tristate + select SND_SOC_DMAENGINE_PCM + select SND_SOC_IMX_PCM + +config SND_SOC_IMX_AUDMUX + tristate + +config SND_MXC_SOC_WM1133_EV1 + tristate "Audio on the i.MX31ADS with WM1133-EV1 fitted" + depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL + select SND_SOC_WM8350 + select SND_SOC_IMX_PCM_FIQ + select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_SSI + help + Enable support for audio on the i.MX31ADS with the WM1133-EV1 + PMIC board with WM8835x fitted. + +config SND_SOC_MX27VIS_AIC32X4 + tristate "SoC audio support for Visstrim M10 boards" + depends on MACH_IMX27_VISSTRIM_M10 && I2C + select SND_SOC_TLV320AIC32X4 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_SSI + help + Say Y if you want to add support for SoC audio on Visstrim SM10 + board with TLV320AIC32X4 codec. + +config SND_SOC_PHYCORE_AC97 + tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" + depends on MACH_PCM043 || MACH_PCA100 + select SND_SOC_AC97_BUS + select SND_SOC_WM9712 + select SND_SOC_IMX_PCM_FIQ + select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_SSI + help + Say Y if you want to add support for SoC audio on Phytec phyCORE + and phyCARD boards in AC97 mode + +config SND_SOC_EUKREA_TLV320 + tristate "Eukrea TLV320" + depends on MACH_EUKREA_MBIMX27_BASEBOARD \ + || MACH_EUKREA_MBIMXSD25_BASEBOARD \ + || MACH_EUKREA_MBIMXSD35_BASEBOARD \ + || MACH_EUKREA_MBIMXSD51_BASEBOARD + depends on I2C + select SND_SOC_TLV320AIC23 + select SND_SOC_IMX_PCM_FIQ + select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_SSI + help + Enable I2S based access to the TLV320AIC23B codec attached + to the SSI interface + +config SND_SOC_IMX_SGTL5000 + tristate "SoC Audio support for i.MX boards with sgtl5000" + depends on OF && I2C + select SND_SOC_SGTL5000 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + help + Say Y if you want to add support for SoC audio on an i.MX board with + a sgtl5000 codec. + +config SND_SOC_IMX_MC13783 + tristate "SoC Audio support for I.MX boards with mc13783" + depends on MFD_MC13783 + select SND_SOC_IMX_SSI + select SND_SOC_IMX_AUDMUX + select SND_SOC_MC13783 + select SND_SOC_IMX_PCM_DMA + +endif # SND_IMX_SOC diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index b4a38c0ac58c..5f3cf3f52ea0 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -8,8 +8,11 @@ obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o # Freescale PowerPC SSI/DMA Platform Support snd-soc-fsl-ssi-objs := fsl_ssi.o +snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o -obj-$(CONFIG_SND_SOC_POWERPC_SSI) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o +obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o +obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o +obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o # MPC5200 Platform Support obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o @@ -20,3 +23,29 @@ obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o obj-$(CONFIG_SND_MPC52xx_SOC_PCM030) += pcm030-audio-fabric.o obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o +# i.MX Platform Support +snd-soc-imx-ssi-objs := imx-ssi.o +snd-soc-imx-audmux-objs := imx-audmux.o + +obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o +obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o + +obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o +snd-soc-imx-pcm-y := imx-pcm.o +snd-soc-imx-pcm-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm-fiq.o +snd-soc-imx-pcm-$(CONFIG_SND_SOC_IMX_PCM_DMA) += imx-pcm-dma.o + +# i.MX Machine Support +snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o +snd-soc-phycore-ac97-objs := phycore-ac97.o +snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o +snd-soc-wm1133-ev1-objs := wm1133-ev1.o +snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-mc13783-objs := imx-mc13783.o + +obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o +obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o +obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o +obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o +obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 7d4475cfdb24..efb9ede01208 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -7,7 +7,7 @@ * which is Copyright 2009 Simtec Electronics * and on sound/soc/imx/phycore-ac97.c which is * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de> - * + * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2eb407fa3b48..4ed2afd47782 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -11,11 +11,15 @@ */ #include <linux/init.h> +#include <linux/io.h> #include <linux/module.h> #include <linux/interrupt.h> +#include <linux/clk.h> #include <linux/device.h> #include <linux/delay.h> #include <linux/slab.h> +#include <linux/of_address.h> +#include <linux/of_irq.h> #include <linux/of_platform.h> #include <sound/core.h> @@ -25,6 +29,26 @@ #include <sound/soc.h> #include "fsl_ssi.h" +#include "imx-pcm.h" + +#ifdef PPC +#define read_ssi(addr) in_be32(addr) +#define write_ssi(val, addr) out_be32(addr, val) +#define write_ssi_mask(addr, clear, set) clrsetbits_be32(addr, clear, set) +#elif defined ARM +#define read_ssi(addr) readl(addr) +#define write_ssi(val, addr) writel(val, addr) +/* + * FIXME: Proper locking should be added at write_ssi_mask caller level + * to ensure this register read/modify/write sequence is race free. + */ +static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set) +{ + u32 val = readl(addr); + val = (val & ~clear) | set; + writel(val, addr); +} +#endif /** * FSLSSI_I2S_RATES: sample rates supported by the I2S @@ -94,6 +118,13 @@ struct fsl_ssi_private { struct device_attribute dev_attr; struct platform_device *pdev; + bool new_binding; + bool ssi_on_imx; + struct clk *clk; + struct platform_device *imx_pcm_pdev; + struct imx_pcm_dma_params dma_params_tx; + struct imx_pcm_dma_params dma_params_rx; + struct { unsigned int rfrc; unsigned int tfrc; @@ -145,7 +176,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) were interrupted for. We mask it with the Interrupt Enable register so that we only check for events that we're interested in. */ - sisr = in_be32(&ssi->sisr) & SIER_FLAGS; + sisr = read_ssi(&ssi->sisr) & SIER_FLAGS; if (sisr & CCSR_SSI_SISR_RFRC) { ssi_private->stats.rfrc++; @@ -260,7 +291,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) /* Clear the bits that we set */ if (sisr2) - out_be32(&ssi->sisr, sisr2); + write_ssi(sisr2, &ssi->sisr); return ret; } @@ -295,7 +326,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * SSI needs to be disabled before updating the registers we set * here. */ - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); /* * Program the SSI into I2S Slave Non-Network Synchronous mode. @@ -303,20 +334,18 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * * FIXME: Little-endian samples require a different shift dir */ - clrsetbits_be32(&ssi->scr, + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE | (synchronous ? CCSR_SSI_SCR_SYN : 0)); - out_be32(&ssi->stcr, - CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | + write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS | - CCSR_SSI_STCR_TSCKP); + CCSR_SSI_STCR_TSCKP, &ssi->stcr); - out_be32(&ssi->srcr, - CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 | + write_ssi(CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 | CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS | - CCSR_SSI_SRCR_RSCKP); + CCSR_SSI_SRCR_RSCKP, &ssi->srcr); /* * The DC and PM bits are only used if the SSI is the clock @@ -324,7 +353,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, */ /* Enable the interrupts and DMA requests */ - out_be32(&ssi->sier, SIER_FLAGS); + write_ssi(SIER_FLAGS, &ssi->sier); /* * Set the watermark for transmit FIFI 0 and receive FIFO 0. We @@ -339,9 +368,9 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * make this value larger (and maybe we should), but this way * data will be written to memory as soon as it's available. */ - out_be32(&ssi->sfcsr, - CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) | - CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2)); + write_ssi(CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) | + CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2), + &ssi->sfcsr); /* * We keep the SSI disabled because if we enable it, then the @@ -393,6 +422,12 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, ssi_private->second_stream = substream; } + if (ssi_private->ssi_on_imx) + snd_soc_dai_set_dma_data(dai, substream, + (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + &ssi_private->dma_params_tx : + &ssi_private->dma_params_rx); + return 0; } @@ -417,7 +452,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, unsigned int sample_size = snd_pcm_format_width(params_format(hw_params)); u32 wl = CCSR_SSI_SxCCR_WL(sample_size); - int enabled = in_be32(&ssi->scr) & CCSR_SSI_SCR_SSIEN; + int enabled = read_ssi(&ssi->scr) & CCSR_SSI_SCR_SSIEN; /* * If we're in synchronous mode, and the SSI is already enabled, @@ -439,9 +474,9 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, /* In synchronous mode, the SSI uses STCCR for capture */ if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) || ssi_private->cpu_dai_drv.symmetric_rates) - clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); + write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); else - clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); + write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); return 0; } @@ -466,19 +501,19 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - setbits32(&ssi->scr, + write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); else - setbits32(&ssi->scr, + write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - clrbits32(&ssi->scr, CCSR_SSI_SCR_TE); + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_TE, 0); else - clrbits32(&ssi->scr, CCSR_SSI_SCR_RE); + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); break; default: @@ -510,7 +545,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, if (!ssi_private->first_stream) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); } } @@ -622,12 +657,6 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) if (!of_device_is_available(np)) return -ENODEV; - /* Check for a codec-handle property. */ - if (!of_get_property(np, "codec-handle", NULL)) { - dev_err(&pdev->dev, "missing codec-handle property\n"); - return -ENODEV; - } - /* We only support the SSI in "I2S Slave" mode */ sprop = of_get_property(np, "fsl,mode", NULL); if (!sprop || strcmp(sprop, "i2s-slave")) { @@ -692,6 +721,50 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) /* Older 8610 DTs didn't have the fifo-depth property */ ssi_private->fifo_depth = 8; + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx21-ssi")) { + u32 dma_events[2]; + ssi_private->ssi_on_imx = true; + + ssi_private->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(ssi_private->clk)) { + ret = PTR_ERR(ssi_private->clk); + dev_err(&pdev->dev, "could not get clock: %d\n", ret); + goto error_irq; + } + clk_prepare_enable(ssi_private->clk); + + /* + * We have burstsize be "fifo_depth - 2" to match the SSI + * watermark setting in fsl_ssi_startup(). + */ + ssi_private->dma_params_tx.burstsize = + ssi_private->fifo_depth - 2; + ssi_private->dma_params_rx.burstsize = + ssi_private->fifo_depth - 2; + ssi_private->dma_params_tx.dma_addr = + ssi_private->ssi_phys + offsetof(struct ccsr_ssi, stx0); + ssi_private->dma_params_rx.dma_addr = + ssi_private->ssi_phys + offsetof(struct ccsr_ssi, srx0); + /* + * TODO: This is a temporary solution and should be changed + * to use generic DMA binding later when the helplers get in. + */ + ret = of_property_read_u32_array(pdev->dev.of_node, + "fsl,ssi-dma-events", dma_events, 2); + if (ret) { + dev_err(&pdev->dev, "could not get dma events\n"); + goto error_clk; + } + ssi_private->dma_params_tx.dma = dma_events[0]; + ssi_private->dma_params_rx.dma = dma_events[1]; + + ssi_private->dma_params_tx.shared_peripheral = + of_device_is_compatible(of_get_parent(np), + "fsl,spba-bus"); + ssi_private->dma_params_rx.shared_peripheral = + ssi_private->dma_params_tx.shared_peripheral; + } + /* Initialize the the device_attribute structure */ dev_attr = &ssi_private->dev_attr; sysfs_attr_init(&dev_attr->attr); @@ -715,6 +788,26 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) goto error_dev; } + if (ssi_private->ssi_on_imx) { + ssi_private->imx_pcm_pdev = + platform_device_register_simple("imx-pcm-audio", + -1, NULL, 0); + if (IS_ERR(ssi_private->imx_pcm_pdev)) { + ret = PTR_ERR(ssi_private->imx_pcm_pdev); + goto error_dev; + } + } + + /* + * If codec-handle property is missing from SSI node, we assume + * that the machine driver uses new binding which does not require + * SSI driver to trigger machine driver's probe. + */ + if (!of_get_property(np, "codec-handle", NULL)) { + ssi_private->new_binding = true; + goto done; + } + /* Trigger the machine driver's probe function. The platform driver * name of the machine driver is taken from /compatible property of the * device tree. We also pass the address of the CPU DAI driver @@ -736,15 +829,24 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) goto error_dai; } +done: return 0; error_dai: + if (ssi_private->ssi_on_imx) + platform_device_unregister(ssi_private->imx_pcm_pdev); snd_soc_unregister_dai(&pdev->dev); error_dev: dev_set_drvdata(&pdev->dev, NULL); device_remove_file(&pdev->dev, dev_attr); +error_clk: + if (ssi_private->ssi_on_imx) { + clk_disable_unprepare(ssi_private->clk); + clk_put(ssi_private->clk); + } + error_irq: free_irq(ssi_private->irq, ssi_private); @@ -764,7 +866,13 @@ static int fsl_ssi_remove(struct platform_device *pdev) { struct fsl_ssi_private *ssi_private = dev_get_drvdata(&pdev->dev); - platform_device_unregister(ssi_private->pdev); + if (!ssi_private->new_binding) + platform_device_unregister(ssi_private->pdev); + if (ssi_private->ssi_on_imx) { + platform_device_unregister(ssi_private->imx_pcm_pdev); + clk_disable_unprepare(ssi_private->clk); + clk_put(ssi_private->clk); + } snd_soc_unregister_dai(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); @@ -779,6 +887,7 @@ static int fsl_ssi_remove(struct platform_device *pdev) static const struct of_device_id fsl_ssi_ids[] = { { .compatible = "fsl,mpc8610-ssi", }, + { .compatible = "fsl,imx21-ssi", }, {} }; MODULE_DEVICE_TABLE(of, fsl_ssi_ids); diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c new file mode 100644 index 000000000000..b9e42b503a37 --- /dev/null +++ b/sound/soc/fsl/fsl_utils.c @@ -0,0 +1,91 @@ +/** + * Freescale ALSA SoC Machine driver utility + * + * Author: Timur Tabi <timur@freescale.com> + * + * Copyright 2010 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/module.h> +#include <linux/of_address.h> +#include <sound/soc.h> + +#include "fsl_utils.h" + +/** + * fsl_asoc_get_dma_channel - determine the dma channel for a SSI node + * + * @ssi_np: pointer to the SSI device tree node + * @name: name of the phandle pointing to the dma channel + * @dai: ASoC DAI link pointer to be filled with platform_name + * @dma_channel_id: dma channel id to be returned + * @dma_id: dma id to be returned + * + * This function determines the dma and channel id for given SSI node. It + * also discovers the platform_name for the ASoC DAI link. + */ +int fsl_asoc_get_dma_channel(struct device_node *ssi_np, + const char *name, + struct snd_soc_dai_link *dai, + unsigned int *dma_channel_id, + unsigned int *dma_id) +{ + struct resource res; + struct device_node *dma_channel_np, *dma_np; + const u32 *iprop; + int ret; + + dma_channel_np = of_parse_phandle(ssi_np, name, 0); + if (!dma_channel_np) + return -EINVAL; + + if (!of_device_is_compatible(dma_channel_np, "fsl,ssi-dma-channel")) { + of_node_put(dma_channel_np); + return -EINVAL; + } + + /* Determine the dev_name for the device_node. This code mimics the + * behavior of of_device_make_bus_id(). We need this because ASoC uses + * the dev_name() of the device to match the platform (DMA) device with + * the CPU (SSI) device. It's all ugly and hackish, but it works (for + * now). + * + * dai->platform name should already point to an allocated buffer. + */ + ret = of_address_to_resource(dma_channel_np, 0, &res); + if (ret) { + of_node_put(dma_channel_np); + return ret; + } + snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", + (unsigned long long) res.start, dma_channel_np->name); + + iprop = of_get_property(dma_channel_np, "cell-index", NULL); + if (!iprop) { + of_node_put(dma_channel_np); + return -EINVAL; + } + *dma_channel_id = be32_to_cpup(iprop); + + dma_np = of_get_parent(dma_channel_np); + iprop = of_get_property(dma_np, "cell-index", NULL); + if (!iprop) { + of_node_put(dma_np); + return -EINVAL; + } + *dma_id = be32_to_cpup(iprop); + + of_node_put(dma_np); + of_node_put(dma_channel_np); + + return 0; +} +EXPORT_SYMBOL(fsl_asoc_get_dma_channel); + +MODULE_AUTHOR("Timur Tabi <timur@freescale.com>"); +MODULE_DESCRIPTION("Freescale ASoC utility code"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/fsl_utils.h b/sound/soc/fsl/fsl_utils.h new file mode 100644 index 000000000000..b2951126527c --- /dev/null +++ b/sound/soc/fsl/fsl_utils.h @@ -0,0 +1,26 @@ +/** + * Freescale ALSA SoC Machine driver utility + * + * Author: Timur Tabi <timur@freescale.com> + * + * Copyright 2010 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _FSL_UTILS_H +#define _FSL_UTILS_H + +#define DAI_NAME_SIZE 32 + +struct snd_soc_dai_link; +struct device_node; + +int fsl_asoc_get_dma_channel(struct device_node *ssi_np, const char *name, + struct snd_soc_dai_link *dai, + unsigned int *dma_channel_id, + unsigned int *dma_id); + +#endif /* _FSL_UTILS_H */ diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index f23700359c67..e7c800ebbd75 100644 --- a/sound/soc/imx/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -26,6 +26,7 @@ #include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/slab.h> +#include <linux/pinctrl/consumer.h> #include "imx-audmux.h" @@ -155,7 +156,7 @@ static void __init audmux_debugfs_init(void) return; } - for (i = 0; i < MX31_AUDMUX_PORT6_SSI_PINS_6 + 1; i++) { + for (i = 0; i < MX31_AUDMUX_PORT7_SSI_PINS_7 + 1; i++) { snprintf(buf, sizeof(buf), "ssi%d", i); if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, (void *)i, &audmux_debugfs_fops)) @@ -249,6 +250,7 @@ EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); static int __devinit imx_audmux_probe(struct platform_device *pdev) { struct resource *res; + struct pinctrl *pinctrl; const struct of_device_id *of_id = of_match_device(imx_audmux_dt_ids, &pdev->dev); @@ -257,6 +259,12 @@ static int __devinit imx_audmux_probe(struct platform_device *pdev) if (!audmux_base) return -EADDRNOTAVAIL; + pinctrl = devm_pinctrl_get_select_default(&pdev->dev); + if (IS_ERR(pinctrl)) { + dev_err(&pdev->dev, "setup pinctrl failed!"); + return PTR_ERR(pinctrl); + } + audmux_clk = clk_get(&pdev->dev, "audmux"); if (IS_ERR(audmux_clk)) { dev_dbg(&pdev->dev, "cannot get clock: %ld\n", diff --git a/sound/soc/imx/imx-audmux.h b/sound/soc/fsl/imx-audmux.h index 04ebbab8d7b9..b8ff44b9dafa 100644 --- a/sound/soc/imx/imx-audmux.h +++ b/sound/soc/fsl/imx-audmux.h @@ -14,6 +14,7 @@ #define MX31_AUDMUX_PORT4_SSI_PINS_4 3 #define MX31_AUDMUX_PORT5_SSI_PINS_5 4 #define MX31_AUDMUX_PORT6_SSI_PINS_6 5 +#define MX31_AUDMUX_PORT7_SSI_PINS_7 6 #define MX51_AUDMUX_PORT1_SSI0 0 #define MX51_AUDMUX_PORT2_SSI1 1 diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c new file mode 100644 index 000000000000..549b31fdc9dd --- /dev/null +++ b/sound/soc/fsl/imx-mc13783.c @@ -0,0 +1,173 @@ +/* + * imx-mc13783.c -- SoC audio for imx based boards with mc13783 codec + * + * Copyright 2012 Philippe Retornaz, <philippe.retornaz@epfl.ch> + * + * Heavly based on phycore-mc13783: + * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <asm/mach-types.h> + +#include "../codecs/mc13783.h" +#include "imx-ssi.h" +#include "imx-audmux.h" + +#define FMT_SSI (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) + +static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xfffffffc, 0xfffffffc, + 4, 16); + if (ret) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, MC13783_CLK_CLIA, 26000000, 0); + if (ret) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x0, 0xfffffffc, 2, 16); + if (ret) + return ret; + + return 0; +} + +static struct snd_soc_ops imx_mc13783_hifi_ops = { + .hw_params = imx_mc13783_hifi_hw_params, +}; + +static struct snd_soc_dai_link imx_mc13783_dai_mc13783[] = { + { + .name = "MC13783", + .stream_name = "Sound", + .codec_dai_name = "mc13783-hifi", + .codec_name = "mc13783-codec", + .cpu_dai_name = "imx-ssi.0", + .platform_name = "imx-pcm-audio.0", + .ops = &imx_mc13783_hifi_ops, + .symmetric_rates = 1, + .dai_fmt = FMT_SSI, + }, +}; + +static const struct snd_soc_dapm_widget imx_mc13783_widget[] = { + SND_SOC_DAPM_MIC("Mic", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route imx_mc13783_routes[] = { + {"Speaker", NULL, "LSP"}, + {"Headphone", NULL, "HSL"}, + {"Headphone", NULL, "HSR"}, + + {"MC1LIN", NULL, "MC1 Bias"}, + {"MC2IN", NULL, "MC2 Bias"}, + {"MC1 Bias", NULL, "Mic"}, + {"MC2 Bias", NULL, "Mic"}, +}; + +static struct snd_soc_card imx_mc13783 = { + .name = "imx_mc13783", + .dai_link = imx_mc13783_dai_mc13783, + .num_links = ARRAY_SIZE(imx_mc13783_dai_mc13783), + .dapm_widgets = imx_mc13783_widget, + .num_dapm_widgets = ARRAY_SIZE(imx_mc13783_widget), + .dapm_routes = imx_mc13783_routes, + .num_dapm_routes = ARRAY_SIZE(imx_mc13783_routes), +}; + +static int __devinit imx_mc13783_probe(struct platform_device *pdev) +{ + int ret; + + imx_mc13783.dev = &pdev->dev; + + ret = snd_soc_register_card(&imx_mc13783); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + return ret; + } + + if (machine_is_mx31_3ds()) { + imx_audmux_v2_configure_port(MX31_AUDMUX_PORT4_SSI_PINS_4, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0) | + IMX_AUDMUX_V2_PDCR_MODE(1) | + IMX_AUDMUX_V2_PDCR_INMMASK(0xfc)); + imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) | + IMX_AUDMUX_V2_PTCR_TCLKDIR | + IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_RFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_RCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4), + IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT4_SSI_PINS_4)); + } else if (machine_is_mx27_3ds()) { + imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, + IMX_AUDMUX_V1_PCR_SYN | + IMX_AUDMUX_V1_PCR_TFSDIR | + IMX_AUDMUX_V1_PCR_TCLKDIR | + IMX_AUDMUX_V1_PCR_RFSDIR | + IMX_AUDMUX_V1_PCR_RCLKDIR | + IMX_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | + IMX_AUDMUX_V1_PCR_RFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | + IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) + ); + imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR3_SSI_PINS_4, + IMX_AUDMUX_V1_PCR_SYN | + IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) + ); + } + + return ret; +} + +static int __devexit imx_mc13783_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&imx_mc13783); + + return 0; +} + +static struct platform_driver imx_mc13783_audio_driver = { + .driver = { + .name = "imx_mc13783", + .owner = THIS_MODULE, + }, + .probe = imx_mc13783_probe, + .remove = __devexit_p(imx_mc13783_remove) +}; + +module_platform_driver(imx_mc13783_audio_driver); + +MODULE_AUTHOR("Sascha Hauer <s.hauer@pengutronix.de>"); +MODULE_AUTHOR("Philippe Retornaz <philippe.retornaz@epfl.ch"); +MODULE_DESCRIPTION("imx with mc13783 codec ALSA SoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:imx_mc13783"); diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/fsl/imx-pcm-dma.c index 6b818de2fc03..48f9d886f020 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -109,7 +109,8 @@ static int snd_imx_open(struct snd_pcm_substream *substream) dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); dma_data = kzalloc(sizeof(*dma_data), GFP_KERNEL); - dma_data->peripheral_type = IMX_DMATYPE_SSI; + dma_data->peripheral_type = dma_params->shared_peripheral ? + IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI; dma_data->priority = DMA_PRIO_HIGH; dma_data->dma_request = dma_params->dma; @@ -140,7 +141,7 @@ static struct snd_pcm_ops imx_pcm_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_imx_pcm_hw_params, .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer, + .pointer = snd_dmaengine_pcm_pointer_no_residue, .mmap = snd_imx_pcm_mmap, }; diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 456b7d723d66..ee27ba3933bd 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -29,6 +29,7 @@ #include <asm/fiq.h> +#include <mach/irqs.h> #include <mach/ssi.h> #include "imx-ssi.h" diff --git a/sound/soc/imx/imx-pcm.c b/sound/soc/fsl/imx-pcm.c index 93dc360b1777..93dc360b1777 100644 --- a/sound/soc/imx/imx-pcm.c +++ b/sound/soc/fsl/imx-pcm.c diff --git a/sound/soc/imx/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index b5f5c3acf34d..83c0ed7d55c9 100644 --- a/sound/soc/imx/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -22,6 +22,7 @@ struct imx_pcm_dma_params { int dma; unsigned long dma_addr; int burstsize; + bool shared_peripheral; /* The peripheral is on SPBA bus */ }; int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c new file mode 100644 index 000000000000..fb21b17f17f5 --- /dev/null +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -0,0 +1,220 @@ +/* + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/of_i2c.h> +#include <linux/clk.h> +#include <sound/soc.h> + +#include "../codecs/sgtl5000.h" +#include "imx-audmux.h" + +#define DAI_NAME_SIZE 32 + +struct imx_sgtl5000_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + struct clk *codec_clk; + unsigned int clk_frequency; +}; + +static int imx_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct imx_sgtl5000_data *data = container_of(rtd->card, + struct imx_sgtl5000_data, card); + struct device *dev = rtd->card->dev; + int ret; + + ret = snd_soc_dai_set_sysclk(rtd->codec_dai, SGTL5000_SYSCLK, + data->clk_frequency, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "could not set codec driver clock params\n"); + return ret; + } + + return 0; +} + +static const struct snd_soc_dapm_widget imx_sgtl5000_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_LINE("Line In Jack", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Line Out Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static int __devinit imx_sgtl5000_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct i2c_client *codec_dev; + struct imx_sgtl5000_data *data; + int int_port, ext_port; + int ret; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(&pdev->dev, "audmux internal port setup failed\n"); + return ret; + } + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(&pdev->dev, "audmux external port setup failed\n"); + return ret; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + return -EINVAL; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + data->codec_clk = clk_get(&codec_dev->dev, NULL); + if (IS_ERR(data->codec_clk)) { + /* assuming clock enabled by default */ + data->codec_clk = NULL; + ret = of_property_read_u32(codec_np, "clock-frequency", + &data->clk_frequency); + if (ret) { + dev_err(&codec_dev->dev, + "clock-frequency missing or invalid\n"); + goto fail; + } + } else { + data->clk_frequency = clk_get_rate(data->codec_clk); + clk_prepare_enable(data->codec_clk); + } + + data->dai.name = "HiFi"; + data->dai.stream_name = "HiFi"; + data->dai.codec_dai_name = "sgtl5000"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.platform_name = "imx-pcm-audio"; + data->dai.init = &imx_sgtl5000_dai_init; + data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto clk_fail; + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto clk_fail; + data->card.num_links = 1; + data->card.dai_link = &data->dai; + data->card.dapm_widgets = imx_sgtl5000_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets); + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto clk_fail; + } + + platform_set_drvdata(pdev, data); +clk_fail: + clk_put(data->codec_clk); +fail: + if (ssi_np) + of_node_put(ssi_np); + if (codec_np) + of_node_put(codec_np); + + return ret; +} + +static int __devexit imx_sgtl5000_remove(struct platform_device *pdev) +{ + struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); + + if (data->codec_clk) { + clk_disable_unprepare(data->codec_clk); + clk_put(data->codec_clk); + } + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_sgtl5000_dt_ids[] = { + { .compatible = "fsl,imx-audio-sgtl5000", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_sgtl5000_dt_ids); + +static struct platform_driver imx_sgtl5000_driver = { + .driver = { + .name = "imx-sgtl5000", + .owner = THIS_MODULE, + .of_match_table = imx_sgtl5000_dt_ids, + }, + .probe = imx_sgtl5000_probe, + .remove = __devexit_p(imx_sgtl5000_remove), +}; +module_platform_driver(imx_sgtl5000_driver); + +MODULE_AUTHOR("Shawn Guo <shawn.guo@linaro.org>"); +MODULE_DESCRIPTION("Freescale i.MX SGTL5000 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-sgtl5000"); diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 4f81ed456325..28dd76c7cb1c 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -28,7 +28,7 @@ * value. When we read the same register two times (and the register still * contains the same value) these status bits are not set. We work * around this by not polling these bits but only wait a fixed delay. - * + * */ #include <linux/clk.h> @@ -543,7 +543,7 @@ static int imx_ssi_probe(struct platform_device *pdev) ret); goto failed_clk; } - clk_enable(ssi->clk); + clk_prepare_enable(ssi->clk); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { @@ -641,7 +641,7 @@ failed_ac97: failed_ioremap: release_mem_region(res->start, resource_size(res)); failed_get_resource: - clk_disable(ssi->clk); + clk_disable_unprepare(ssi->clk); clk_put(ssi->clk); failed_clk: kfree(ssi); @@ -664,7 +664,7 @@ static int __devexit imx_ssi_remove(struct platform_device *pdev) iounmap(ssi->base); release_mem_region(res->start, resource_size(res)); - clk_disable(ssi->clk); + clk_disable_unprepare(ssi->clk); clk_put(ssi->clk); kfree(ssi); diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/fsl/imx-ssi.h index 5744e86ca878..5744e86ca878 100644 --- a/sound/soc/imx/imx-ssi.h +++ b/sound/soc/fsl/imx-ssi.h diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 3fea5a15ffe8..60bcba1bc30e 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -14,18 +14,16 @@ #include <linux/interrupt.h> #include <linux/of_device.h> #include <linux/slab.h> -#include <linux/of_i2c.h> #include <sound/soc.h> #include <asm/fsl_guts.h> #include "fsl_dma.h" #include "fsl_ssi.h" +#include "fsl_utils.h" /* There's only one global utilities register */ static phys_addr_t guts_phys; -#define DAI_NAME_SIZE 32 - /** * mpc8610_hpcd_data: machine-specific ASoC device data * @@ -43,7 +41,6 @@ struct mpc8610_hpcd_data { unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */ unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ char codec_dai_name[DAI_NAME_SIZE]; - char codec_name[DAI_NAME_SIZE]; char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */ }; @@ -181,141 +178,6 @@ static struct snd_soc_ops mpc8610_hpcd_ops = { }; /** - * get_node_by_phandle_name - get a node by its phandle name - * - * This function takes a node, the name of a property in that node, and a - * compatible string. Assuming the property is a phandle to another node, - * it returns that node, (optionally) if that node is compatible. - * - * If the property is not a phandle, or the node it points to is not compatible - * with the specific string, then NULL is returned. - */ -static struct device_node *get_node_by_phandle_name(struct device_node *np, - const char *name, - const char *compatible) -{ - const phandle *ph; - int len; - - ph = of_get_property(np, name, &len); - if (!ph || (len != sizeof(phandle))) - return NULL; - - np = of_find_node_by_phandle(*ph); - if (!np) - return NULL; - - if (compatible && !of_device_is_compatible(np, compatible)) { - of_node_put(np); - return NULL; - } - - return np; -} - -/** - * get_parent_cell_index -- return the cell-index of the parent of a node - * - * Return the value of the cell-index property of the parent of the given - * node. This is used for DMA channel nodes that need to know the DMA ID - * of the controller they are on. - */ -static int get_parent_cell_index(struct device_node *np) -{ - struct device_node *parent = of_get_parent(np); - const u32 *iprop; - - if (!parent) - return -1; - - iprop = of_get_property(parent, "cell-index", NULL); - of_node_put(parent); - - if (!iprop) - return -1; - - return be32_to_cpup(iprop); -} - -/** - * codec_node_dev_name - determine the dev_name for a codec node - * - * This function determines the dev_name for an I2C node. This is the name - * that would be returned by dev_name() if this device_node were part of a - * 'struct device' It's ugly and hackish, but it works. - * - * The dev_name for such devices include the bus number and I2C address. For - * example, "cs4270.0-004f". - */ -static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) -{ - const u32 *iprop; - int addr; - char temp[DAI_NAME_SIZE]; - struct i2c_client *i2c; - - of_modalias_node(np, temp, DAI_NAME_SIZE); - - iprop = of_get_property(np, "reg", NULL); - if (!iprop) - return -EINVAL; - - addr = be32_to_cpup(iprop); - - /* We need the adapter number */ - i2c = of_find_i2c_device_by_node(np); - if (!i2c) - return -ENODEV; - - snprintf(buf, len, "%s.%u-%04x", temp, i2c->adapter->nr, addr); - - return 0; -} - -static int get_dma_channel(struct device_node *ssi_np, - const char *name, - struct snd_soc_dai_link *dai, - unsigned int *dma_channel_id, - unsigned int *dma_id) -{ - struct resource res; - struct device_node *dma_channel_np; - const u32 *iprop; - int ret; - - dma_channel_np = get_node_by_phandle_name(ssi_np, name, - "fsl,ssi-dma-channel"); - if (!dma_channel_np) - return -EINVAL; - - /* Determine the dev_name for the device_node. This code mimics the - * behavior of of_device_make_bus_id(). We need this because ASoC uses - * the dev_name() of the device to match the platform (DMA) device with - * the CPU (SSI) device. It's all ugly and hackish, but it works (for - * now). - * - * dai->platform name should already point to an allocated buffer. - */ - ret = of_address_to_resource(dma_channel_np, 0, &res); - if (ret) - return ret; - snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", - (unsigned long long) res.start, dma_channel_np->name); - - iprop = of_get_property(dma_channel_np, "cell-index", NULL); - if (!iprop) { - of_node_put(dma_channel_np); - return -EINVAL; - } - - *dma_channel_id = be32_to_cpup(iprop); - *dma_id = get_parent_cell_index(dma_channel_np); - of_node_put(dma_channel_np); - - return 0; -} - -/** * mpc8610_hpcd_probe: platform probe function for the machine driver * * Although this is a machine driver, the SSI node is the "master" node with @@ -352,16 +214,8 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev); machine_data->dai[0].ops = &mpc8610_hpcd_ops; - /* Determine the codec name, it will be used as the codec DAI name */ - ret = codec_node_dev_name(codec_np, machine_data->codec_name, - DAI_NAME_SIZE); - if (ret) { - dev_err(&pdev->dev, "invalid codec node %s\n", - codec_np->full_name); - ret = -EINVAL; - goto error; - } - machine_data->dai[0].codec_name = machine_data->codec_name; + /* ASoC core can match codec with device node */ + machine_data->dai[0].codec_of_node = codec_np; /* The DAI name from the codec (snd_soc_dai_driver.name) */ machine_data->dai[0].codec_dai_name = "cs4270-hifi"; @@ -458,9 +312,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) /* Find the playback DMA channel to use. */ machine_data->dai[0].platform_name = machine_data->platform_name[0]; - ret = get_dma_channel(np, "fsl,playback-dma", &machine_data->dai[0], - &machine_data->dma_channel_id[0], - &machine_data->dma_id[0]); + ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma", + &machine_data->dai[0], + &machine_data->dma_channel_id[0], + &machine_data->dma_id[0]); if (ret) { dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n"); goto error; @@ -468,9 +323,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) /* Find the capture DMA channel to use. */ machine_data->dai[1].platform_name = machine_data->platform_name[1]; - ret = get_dma_channel(np, "fsl,capture-dma", &machine_data->dai[1], - &machine_data->dma_channel_id[1], - &machine_data->dma_id[1]); + ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma", + &machine_data->dai[1], + &machine_data->dma_channel_id[1], + &machine_data->dma_id[1]); if (ret) { dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n"); goto error; diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c index f6d04ad4bb39..f6d04ad4bb39 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/fsl/mx27vis-aic32x4.c diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 982a1c944983..50adf4032bcc 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -14,12 +14,12 @@ #include <linux/interrupt.h> #include <linux/of_device.h> #include <linux/slab.h> -#include <linux/of_i2c.h> #include <sound/soc.h> #include <asm/fsl_guts.h> #include "fsl_dma.h" #include "fsl_ssi.h" +#include "fsl_utils.h" /* P1022-specific PMUXCR and DMUXCR bit definitions */ @@ -57,8 +57,6 @@ static inline void guts_set_dmuxcr(struct ccsr_guts __iomem *guts, /* There's only one global utilities register */ static phys_addr_t guts_phys; -#define DAI_NAME_SIZE 32 - /** * machine_data: machine-specific ASoC device data * @@ -75,7 +73,6 @@ struct machine_data { unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */ unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ - char codec_name[DAI_NAME_SIZE]; char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */ }; @@ -191,136 +188,6 @@ static struct snd_soc_ops p1022_ds_ops = { }; /** - * get_node_by_phandle_name - get a node by its phandle name - * - * This function takes a node, the name of a property in that node, and a - * compatible string. Assuming the property is a phandle to another node, - * it returns that node, (optionally) if that node is compatible. - * - * If the property is not a phandle, or the node it points to is not compatible - * with the specific string, then NULL is returned. - */ -static struct device_node *get_node_by_phandle_name(struct device_node *np, - const char *name, const char *compatible) -{ - np = of_parse_phandle(np, name, 0); - if (!np) - return NULL; - - if (!of_device_is_compatible(np, compatible)) { - of_node_put(np); - return NULL; - } - - return np; -} - -/** - * get_parent_cell_index -- return the cell-index of the parent of a node - * - * Return the value of the cell-index property of the parent of the given - * node. This is used for DMA channel nodes that need to know the DMA ID - * of the controller they are on. - */ -static int get_parent_cell_index(struct device_node *np) -{ - struct device_node *parent = of_get_parent(np); - const u32 *iprop; - int ret = -1; - - if (!parent) - return -1; - - iprop = of_get_property(parent, "cell-index", NULL); - if (iprop) - ret = be32_to_cpup(iprop); - - of_node_put(parent); - - return ret; -} - -/** - * codec_node_dev_name - determine the dev_name for a codec node - * - * This function determines the dev_name for an I2C node. This is the name - * that would be returned by dev_name() if this device_node were part of a - * 'struct device' It's ugly and hackish, but it works. - * - * The dev_name for such devices include the bus number and I2C address. For - * example, "cs4270-codec.0-004f". - */ -static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) -{ - const u32 *iprop; - int addr; - char temp[DAI_NAME_SIZE]; - struct i2c_client *i2c; - - of_modalias_node(np, temp, DAI_NAME_SIZE); - - iprop = of_get_property(np, "reg", NULL); - if (!iprop) - return -EINVAL; - - addr = be32_to_cpup(iprop); - - /* We need the adapter number */ - i2c = of_find_i2c_device_by_node(np); - if (!i2c) - return -ENODEV; - - snprintf(buf, len, "%s.%u-%04x", temp, i2c->adapter->nr, addr); - - return 0; -} - -static int get_dma_channel(struct device_node *ssi_np, - const char *name, - struct snd_soc_dai_link *dai, - unsigned int *dma_channel_id, - unsigned int *dma_id) -{ - struct resource res; - struct device_node *dma_channel_np; - const u32 *iprop; - int ret; - - dma_channel_np = get_node_by_phandle_name(ssi_np, name, - "fsl,ssi-dma-channel"); - if (!dma_channel_np) - return -EINVAL; - - /* Determine the dev_name for the device_node. This code mimics the - * behavior of of_device_make_bus_id(). We need this because ASoC uses - * the dev_name() of the device to match the platform (DMA) device with - * the CPU (SSI) device. It's all ugly and hackish, but it works (for - * now). - * - * dai->platform name should already point to an allocated buffer. - */ - ret = of_address_to_resource(dma_channel_np, 0, &res); - if (ret) { - of_node_put(dma_channel_np); - return ret; - } - snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", - (unsigned long long) res.start, dma_channel_np->name); - - iprop = of_get_property(dma_channel_np, "cell-index", NULL); - if (!iprop) { - of_node_put(dma_channel_np); - return -EINVAL; - } - - *dma_channel_id = be32_to_cpup(iprop); - *dma_id = get_parent_cell_index(dma_channel_np); - of_node_put(dma_channel_np); - - return 0; -} - -/** * p1022_ds_probe: platform probe function for the machine driver * * Although this is a machine driver, the SSI node is the "master" node with @@ -357,15 +224,8 @@ static int p1022_ds_probe(struct platform_device *pdev) mdata->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev); mdata->dai[0].ops = &p1022_ds_ops; - /* Determine the codec name, it will be used as the codec DAI name */ - ret = codec_node_dev_name(codec_np, mdata->codec_name, DAI_NAME_SIZE); - if (ret) { - dev_err(&pdev->dev, "invalid codec node %s\n", - codec_np->full_name); - ret = -EINVAL; - goto error; - } - mdata->dai[0].codec_name = mdata->codec_name; + /* ASoC core can match codec with device node */ + mdata->dai[0].codec_of_node = codec_np; /* We register two DAIs per SSI, one for playback and the other for * capture. We support codecs that have separate DAIs for both playback @@ -462,9 +322,9 @@ static int p1022_ds_probe(struct platform_device *pdev) /* Find the playback DMA channel to use. */ mdata->dai[0].platform_name = mdata->platform_name[0]; - ret = get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0], - &mdata->dma_channel_id[0], - &mdata->dma_id[0]); + ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0], + &mdata->dma_channel_id[0], + &mdata->dma_id[0]); if (ret) { dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n"); goto error; @@ -472,9 +332,9 @@ static int p1022_ds_probe(struct platform_device *pdev) /* Find the capture DMA channel to use. */ mdata->dai[1].platform_name = mdata->platform_name[1]; - ret = get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1], - &mdata->dma_channel_id[1], - &mdata->dma_id[1]); + ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1], + &mdata->dma_channel_id[1], + &mdata->dma_id[1]); if (ret) { dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n"); goto error; diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/fsl/phycore-ac97.c index f8da6dd115ed..f8da6dd115ed 100644 --- a/sound/soc/imx/phycore-ac97.c +++ b/sound/soc/fsl/phycore-ac97.c diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index fe54a69073e5..fe54a69073e5 100644 --- a/sound/soc/imx/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig new file mode 100644 index 000000000000..610f61251640 --- /dev/null +++ b/sound/soc/generic/Kconfig @@ -0,0 +1,4 @@ +config SND_SIMPLE_CARD + tristate "ASoC Simple sound card support" + help + This option enables generic simple sound card support diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile new file mode 100644 index 000000000000..9c3b246792bf --- /dev/null +++ b/sound/soc/generic/Makefile @@ -0,0 +1,3 @@ +snd-soc-simple-card-objs := simple-card.o + +obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c new file mode 100644 index 000000000000..b4b4cab30232 --- /dev/null +++ b/sound/soc/generic/simple-card.c @@ -0,0 +1,114 @@ +/* + * ASoC simple sound card support + * + * Copyright (C) 2012 Renesas Solutions Corp. + * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/platform_device.h> +#include <linux/module.h> +#include <sound/simple_card.h> + +#define asoc_simple_get_card_info(p) \ + container_of(p->dai_link, struct asoc_simple_card_info, snd_link) + +static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct asoc_simple_card_info *cinfo = asoc_simple_get_card_info(rtd); + struct asoc_simple_dai_init_info *iinfo = cinfo->init; + struct snd_soc_dai *codec = rtd->codec_dai; + struct snd_soc_dai *cpu = rtd->cpu_dai; + unsigned int cpu_daifmt = iinfo->fmt | iinfo->cpu_daifmt; + unsigned int codec_daifmt = iinfo->fmt | iinfo->codec_daifmt; + int ret; + + if (codec_daifmt) { + ret = snd_soc_dai_set_fmt(codec, codec_daifmt); + if (ret < 0) + return ret; + } + + if (iinfo->sysclk) { + ret = snd_soc_dai_set_sysclk(codec, 0, iinfo->sysclk, 0); + if (ret < 0) + return ret; + } + + if (cpu_daifmt) { + ret = snd_soc_dai_set_fmt(cpu, cpu_daifmt); + if (ret < 0) + return ret; + } + + return 0; +} + +static int asoc_simple_card_probe(struct platform_device *pdev) +{ + struct asoc_simple_card_info *cinfo = pdev->dev.platform_data; + + if (!cinfo) { + dev_err(&pdev->dev, "no info for asoc-simple-card\n"); + return -EINVAL; + } + + if (!cinfo->name || + !cinfo->card || + !cinfo->cpu_dai || + !cinfo->codec || + !cinfo->platform || + !cinfo->codec_dai) { + dev_err(&pdev->dev, "insufficient asoc_simple_card_info settings\n"); + return -EINVAL; + } + + /* + * init snd_soc_dai_link + */ + cinfo->snd_link.name = cinfo->name; + cinfo->snd_link.stream_name = cinfo->name; + cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai; + cinfo->snd_link.platform_name = cinfo->platform; + cinfo->snd_link.codec_name = cinfo->codec; + cinfo->snd_link.codec_dai_name = cinfo->codec_dai; + + /* enable snd_link.init if cinfo has settings */ + if (cinfo->init) + cinfo->snd_link.init = asoc_simple_card_dai_init; + + /* + * init snd_soc_card + */ + cinfo->snd_card.name = cinfo->card; + cinfo->snd_card.owner = THIS_MODULE; + cinfo->snd_card.dai_link = &cinfo->snd_link; + cinfo->snd_card.num_links = 1; + cinfo->snd_card.dev = &pdev->dev; + + return snd_soc_register_card(&cinfo->snd_card); +} + +static int asoc_simple_card_remove(struct platform_device *pdev) +{ + struct asoc_simple_card_info *cinfo = pdev->dev.platform_data; + + return snd_soc_unregister_card(&cinfo->snd_card); +} + +static struct platform_driver asoc_simple_card = { + .driver = { + .name = "asoc-simple-card", + }, + .probe = asoc_simple_card_probe, + .remove = asoc_simple_card_remove, +}; + +module_platform_driver(asoc_simple_card); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("ASoC Simple Sound Card"); +MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>"); diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig deleted file mode 100644 index 810acaa09009..000000000000 --- a/sound/soc/imx/Kconfig +++ /dev/null @@ -1,79 +0,0 @@ -menuconfig SND_IMX_SOC - tristate "SoC Audio for Freescale i.MX CPUs" - depends on ARCH_MXC - help - Say Y or M if you want to add support for codecs attached to - the i.MX SSI interface. - - -if SND_IMX_SOC - -config SND_SOC_IMX_SSI - tristate - -config SND_SOC_IMX_PCM - tristate - -config SND_MXC_SOC_FIQ - tristate - select FIQ - select SND_SOC_IMX_PCM - -config SND_MXC_SOC_MX2 - select SND_SOC_DMAENGINE_PCM - tristate - select SND_SOC_IMX_PCM - -config SND_SOC_IMX_AUDMUX - tristate - -config SND_MXC_SOC_WM1133_EV1 - tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted" - depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL - select SND_SOC_WM8350 - select SND_MXC_SOC_FIQ - select SND_SOC_IMX_AUDMUX - select SND_SOC_IMX_SSI - help - Enable support for audio on the i.MX31ADS with the WM1133-EV1 - PMIC board with WM8835x fitted. - -config SND_SOC_MX27VIS_AIC32X4 - tristate "SoC audio support for Visstrim M10 boards" - depends on MACH_IMX27_VISSTRIM_M10 && I2C - select SND_SOC_TLV320AIC32X4 - select SND_MXC_SOC_MX2 - select SND_SOC_IMX_AUDMUX - select SND_SOC_IMX_SSI - help - Say Y if you want to add support for SoC audio on Visstrim SM10 - board with TLV320AIC32X4 codec. - -config SND_SOC_PHYCORE_AC97 - tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" - depends on MACH_PCM043 || MACH_PCA100 - select SND_SOC_AC97_BUS - select SND_SOC_WM9712 - select SND_MXC_SOC_FIQ - select SND_SOC_IMX_AUDMUX - select SND_SOC_IMX_SSI - help - Say Y if you want to add support for SoC audio on Phytec phyCORE - and phyCARD boards in AC97 mode - -config SND_SOC_EUKREA_TLV320 - tristate "Eukrea TLV320" - depends on MACH_EUKREA_MBIMX27_BASEBOARD \ - || MACH_EUKREA_MBIMXSD25_BASEBOARD \ - || MACH_EUKREA_MBIMXSD35_BASEBOARD \ - || MACH_EUKREA_MBIMXSD51_BASEBOARD - depends on I2C - select SND_SOC_TLV320AIC23 - select SND_MXC_SOC_FIQ - select SND_SOC_IMX_AUDMUX - select SND_SOC_IMX_SSI - help - Enable I2S based access to the TLV320AIC23B codec attached - to the SSI interface - -endif # SND_IMX_SOC diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile deleted file mode 100644 index f5db3e92d0d1..000000000000 --- a/sound/soc/imx/Makefile +++ /dev/null @@ -1,22 +0,0 @@ -# i.MX Platform Support -snd-soc-imx-ssi-objs := imx-ssi.o -snd-soc-imx-audmux-objs := imx-audmux.o - -obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o -obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o - -obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o -snd-soc-imx-pcm-y := imx-pcm.o -snd-soc-imx-pcm-$(CONFIG_SND_MXC_SOC_FIQ) += imx-pcm-fiq.o -snd-soc-imx-pcm-$(CONFIG_SND_MXC_SOC_MX2) += imx-pcm-dma-mx2.o - -# i.MX Machine Support -snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o -snd-soc-phycore-ac97-objs := phycore-ac97.o -snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o -snd-soc-wm1133-ev1-objs := wm1133-ev1.o - -obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o -obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o -obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o -obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index a5af7c42e62b..41349670adab 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -346,7 +346,7 @@ static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s) /* Playback */ dma_config = &i2s->pcm_config_playback.dma_config; - dma_config->src_width = JZ4740_DMA_WIDTH_32BIT, + dma_config->src_width = JZ4740_DMA_WIDTH_32BIT; dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE; dma_config->request_type = JZ4740_DMA_TYPE_AIC_TRANSMIT; dma_config->flags = JZ4740_DMA_SRC_AUTOINC; @@ -355,7 +355,7 @@ static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s) /* Capture */ dma_config = &i2s->pcm_config_capture.dma_config; - dma_config->dst_width = JZ4740_DMA_WIDTH_32BIT, + dma_config->dst_width = JZ4740_DMA_WIDTH_32BIT; dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE; dma_config->request_type = JZ4740_DMA_TYPE_AIC_RECEIVE; dma_config->flags = JZ4740_DMA_DST_AUTOINC; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 3cb9aa4299d3..7646dd7f30cb 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -17,6 +17,7 @@ #include <linux/slab.h> #include <linux/mbus.h> #include <linux/delay.h> +#include <linux/clk.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -449,7 +450,21 @@ static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->burst = data->burst; - return snd_soc_register_dai(&pdev->dev, &kirkwood_i2s_dai); + priv->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(priv->clk)) { + dev_err(&pdev->dev, "no clock\n"); + err = PTR_ERR(priv->clk); + goto err_ioremap; + } + clk_prepare_enable(priv->clk); + + err = snd_soc_register_dai(&pdev->dev, &kirkwood_i2s_dai); + if (!err) + return 0; + dev_err(&pdev->dev, "snd_soc_register_dai failed\n"); + + clk_disable_unprepare(priv->clk); + clk_put(priv->clk); err_ioremap: iounmap(priv->io); @@ -466,6 +481,10 @@ static __devexit int kirkwood_i2s_dev_remove(struct platform_device *pdev) struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); + + clk_disable_unprepare(priv->clk); + clk_put(priv->clk); + iounmap(priv->io); release_mem_region(priv->mem->start, SZ_16K); kfree(priv); diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index 9047436b3937..f9084d83e6bd 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -123,6 +123,7 @@ struct kirkwood_dma_data { void __iomem *io; int irq; int burst; + struct clk *clk; }; #endif diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index e373fbbc97a0..f82d766cbf9e 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -141,7 +141,7 @@ static struct snd_pcm_ops mxs_pcm_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_mxs_pcm_hw_params, .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer, + .pointer = snd_dmaengine_pcm_pointer_no_residue, .mmap = snd_mxs_pcm_mmap, }; @@ -220,28 +220,16 @@ static struct snd_soc_platform_driver mxs_soc_platform = { .pcm_free = mxs_pcm_free, }; -static int __devinit mxs_soc_platform_probe(struct platform_device *pdev) +int __devinit mxs_pcm_platform_register(struct device *dev) { - return snd_soc_register_platform(&pdev->dev, &mxs_soc_platform); + return snd_soc_register_platform(dev, &mxs_soc_platform); } +EXPORT_SYMBOL_GPL(mxs_pcm_platform_register); -static int __devexit mxs_soc_platform_remove(struct platform_device *pdev) +void __devexit mxs_pcm_platform_unregister(struct device *dev) { - snd_soc_unregister_platform(&pdev->dev); - - return 0; + snd_soc_unregister_platform(dev); } - -static struct platform_driver mxs_pcm_driver = { - .driver = { - .name = "mxs-pcm-audio", - .owner = THIS_MODULE, - }, - .probe = mxs_soc_platform_probe, - .remove = __devexit_p(mxs_soc_platform_remove), -}; - -module_platform_driver(mxs_pcm_driver); +EXPORT_SYMBOL_GPL(mxs_pcm_platform_unregister); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:mxs-pcm-audio"); diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h index 5f01a9124b3d..35ba2ca42384 100644 --- a/sound/soc/mxs/mxs-pcm.h +++ b/sound/soc/mxs/mxs-pcm.h @@ -24,4 +24,7 @@ struct mxs_pcm_dma_params { int chan_num; }; +int mxs_pcm_platform_register(struct device *dev); +void mxs_pcm_platform_unregister(struct device *dev); + #endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 53f4fd8feced..aba71bfa33b1 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -18,6 +18,8 @@ #include <linux/module.h> #include <linux/init.h> +#include <linux/of.h> +#include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/dma-mapping.h> @@ -25,6 +27,7 @@ #include <linux/delay.h> #include <linux/time.h> #include <linux/fsl/mxs-dma.h> +#include <linux/pinctrl/consumer.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -620,34 +623,61 @@ static irqreturn_t mxs_saif_irq(int irq, void *dev_id) return IRQ_HANDLED; } -static int mxs_saif_probe(struct platform_device *pdev) +static int __devinit mxs_saif_probe(struct platform_device *pdev) { + struct device_node *np = pdev->dev.of_node; struct resource *iores, *dmares; struct mxs_saif *saif; struct mxs_saif_platform_data *pdata; + struct pinctrl *pinctrl; int ret = 0; - if (pdev->id >= ARRAY_SIZE(mxs_saif)) + + if (!np && pdev->id >= ARRAY_SIZE(mxs_saif)) return -EINVAL; saif = devm_kzalloc(&pdev->dev, sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; - mxs_saif[pdev->id] = saif; - saif->id = pdev->id; - - pdata = pdev->dev.platform_data; - if (pdata && !pdata->master_mode) { - saif->master_id = pdata->master_id; - if (saif->master_id < 0 || - saif->master_id >= ARRAY_SIZE(mxs_saif) || - saif->master_id == saif->id) { - dev_err(&pdev->dev, "get wrong master id\n"); - return -EINVAL; + if (np) { + struct device_node *master; + saif->id = of_alias_get_id(np, "saif"); + if (saif->id < 0) + return saif->id; + /* + * If there is no "fsl,saif-master" phandle, it's a saif + * master. Otherwise, it's a slave and its phandle points + * to the master. + */ + master = of_parse_phandle(np, "fsl,saif-master", 0); + if (!master) { + saif->master_id = saif->id; + } else { + saif->master_id = of_alias_get_id(master, "saif"); + if (saif->master_id < 0) + return saif->master_id; } } else { - saif->master_id = saif->id; + saif->id = pdev->id; + pdata = pdev->dev.platform_data; + if (pdata && !pdata->master_mode) + saif->master_id = pdata->master_id; + else + saif->master_id = saif->id; + } + + if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { + dev_err(&pdev->dev, "get wrong master id\n"); + return -EINVAL; + } + + mxs_saif[saif->id] = saif; + + pinctrl = devm_pinctrl_get_select_default(&pdev->dev); + if (IS_ERR(pinctrl)) { + ret = PTR_ERR(pinctrl); + return ret; } saif->clk = clk_get(&pdev->dev, NULL); @@ -669,12 +699,19 @@ static int mxs_saif_probe(struct platform_device *pdev) dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmares) { - ret = -ENODEV; - dev_err(&pdev->dev, "failed to get dma resource: %d\n", - ret); - goto failed_get_resource; + /* + * TODO: This is a temporary solution and should be changed + * to use generic DMA binding later when the helplers get in. + */ + ret = of_property_read_u32(np, "fsl,saif-dma-channel", + &saif->dma_param.chan_num); + if (ret) { + dev_err(&pdev->dev, "failed to get dma channel\n"); + goto failed_get_resource; + } + } else { + saif->dma_param.chan_num = dmares->start; } - saif->dma_param.chan_num = dmares->start; saif->irq = platform_get_irq(pdev, 0); if (saif->irq < 0) { @@ -708,24 +745,14 @@ static int mxs_saif_probe(struct platform_device *pdev) goto failed_get_resource; } - saif->soc_platform_pdev = platform_device_alloc( - "mxs-pcm-audio", pdev->id); - if (!saif->soc_platform_pdev) { - ret = -ENOMEM; - goto failed_pdev_alloc; - } - - platform_set_drvdata(saif->soc_platform_pdev, saif); - ret = platform_device_add(saif->soc_platform_pdev); + ret = mxs_pcm_platform_register(&pdev->dev); if (ret) { - dev_err(&pdev->dev, "failed to add soc platform device\n"); - goto failed_pdev_add; + dev_err(&pdev->dev, "register PCM failed: %d\n", ret); + goto failed_pdev_alloc; } return 0; -failed_pdev_add: - platform_device_put(saif->soc_platform_pdev); failed_pdev_alloc: snd_soc_unregister_dai(&pdev->dev); failed_get_resource: @@ -738,13 +765,19 @@ static int __devexit mxs_saif_remove(struct platform_device *pdev) { struct mxs_saif *saif = platform_get_drvdata(pdev); - platform_device_unregister(saif->soc_platform_pdev); + mxs_pcm_platform_unregister(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); clk_put(saif->clk); return 0; } +static const struct of_device_id mxs_saif_dt_ids[] = { + { .compatible = "fsl,imx28-saif", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, mxs_saif_dt_ids); + static struct platform_driver mxs_saif_driver = { .probe = mxs_saif_probe, .remove = __devexit_p(mxs_saif_remove), @@ -752,6 +785,7 @@ static struct platform_driver mxs_saif_driver = { .driver = { .name = "mxs-saif", .owner = THIS_MODULE, + .of_match_table = mxs_saif_dt_ids, }, }; diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index 12c91e4eb941..3cb342e5bc90 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -123,7 +123,6 @@ struct mxs_saif { unsigned int cur_rate; unsigned int ongoing; - struct platform_device *soc_platform_pdev; u32 fifo_underrun; u32 fifo_overrun; }; diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 60f052b7cf22..215113b05f7d 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -18,6 +18,8 @@ #include <linux/module.h> #include <linux/device.h> +#include <linux/of.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -90,7 +92,7 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { .codec_dai_name = "sgtl5000", .codec_name = "sgtl5000.0-000a", .cpu_dai_name = "mxs-saif.0", - .platform_name = "mxs-pcm-audio.0", + .platform_name = "mxs-saif.0", .ops = &mxs_sgtl5000_hifi_ops, }, { .name = "HiFi Rx", @@ -98,7 +100,7 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { .codec_dai_name = "sgtl5000", .codec_name = "sgtl5000.0-000a", .cpu_dai_name = "mxs-saif.1", - .platform_name = "mxs-pcm-audio.1", + .platform_name = "mxs-saif.1", .ops = &mxs_sgtl5000_hifi_ops, }, }; @@ -110,11 +112,48 @@ static struct snd_soc_card mxs_sgtl5000 = { .num_links = ARRAY_SIZE(mxs_sgtl5000_dai), }; +static int __devinit mxs_sgtl5000_probe_dt(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *saif_np[2], *codec_np; + int i, ret = 0; + + if (!np) + return 1; /* no device tree */ + + saif_np[0] = of_parse_phandle(np, "saif-controllers", 0); + saif_np[1] = of_parse_phandle(np, "saif-controllers", 1); + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!saif_np[0] || !saif_np[1] || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < 2; i++) { + mxs_sgtl5000_dai[i].codec_name = NULL; + mxs_sgtl5000_dai[i].codec_of_node = codec_np; + mxs_sgtl5000_dai[i].cpu_dai_name = NULL; + mxs_sgtl5000_dai[i].cpu_of_node = saif_np[i]; + mxs_sgtl5000_dai[i].platform_name = NULL; + mxs_sgtl5000_dai[i].platform_of_node = saif_np[i]; + } + + of_node_put(codec_np); + of_node_put(saif_np[0]); + of_node_put(saif_np[1]); + + return ret; +} + static int __devinit mxs_sgtl5000_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mxs_sgtl5000; int ret; + ret = mxs_sgtl5000_probe_dt(pdev); + if (ret < 0) + return ret; + /* * Set an init clock(11.28Mhz) for sgtl5000 initialization(i2c r/w). * The Sgtl5000 sysclk is derived from saif0 mclk and it's range @@ -148,10 +187,17 @@ static int __devexit mxs_sgtl5000_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id mxs_sgtl5000_dt_ids[] = { + { .compatible = "fsl,mxs-audio-sgtl5000", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, mxs_sgtl5000_dt_ids); + static struct platform_driver mxs_sgtl5000_audio_driver = { .driver = { .name = "mxs-sgtl5000", .owner = THIS_MODULE, + .of_match_table = mxs_sgtl5000_dt_ids, }, .probe = mxs_sgtl5000_probe, .remove = __devexit_p(mxs_sgtl5000_remove), diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index deafbfaacdbf..57a2fa751085 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -109,10 +109,12 @@ config SND_OMAP_SOC_OMAP_ABE_TWL6040 - PandaBoard (4430) - PandaBoardES (4460) -config SND_OMAP_SOC_OMAP4_HDMI - tristate "SoC Audio support for Texas Instruments OMAP4 HDMI" - depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS && ARCH_OMAP4 +config SND_OMAP_SOC_OMAP_HDMI + tristate "SoC Audio support for Texas Instruments OMAP HDMI" + depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS select SND_OMAP_SOC_HDMI + select SND_SOC_OMAP_HDMI_CODEC + select OMAP4_DSS_HDMI_AUDIO help Say Y if you want to add support for SoC HDMI audio on Texas Instruments OMAP4 chips diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 1d656bce01d4..0e14dd322565 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -25,7 +25,7 @@ snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o snd-soc-zoom2-objs := zoom2.o snd-soc-igep0020-objs := igep0020.o -snd-soc-omap4-hdmi-objs := omap4-hdmi-card.o +snd-soc-omap-hdmi-card-objs := omap-hdmi-card.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o @@ -41,4 +41,4 @@ obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o obj-$(CONFIG_SND_OMAP_SOC_IGEP0020) += snd-soc-igep0020.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP4_HDMI) += snd-soc-omap4-hdmi.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP_HDMI) += snd-soc-omap-hdmi-card.o diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index e5f44440d1b9..34835e8a9160 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -109,6 +109,47 @@ static void omap_mcbsp_dump_reg(struct omap_mcbsp *mcbsp) dev_dbg(mcbsp->dev, "***********************\n"); } +static irqreturn_t omap_mcbsp_irq_handler(int irq, void *dev_id) +{ + struct omap_mcbsp *mcbsp = dev_id; + u16 irqst; + + irqst = MCBSP_READ(mcbsp, IRQST); + dev_dbg(mcbsp->dev, "IRQ callback : 0x%x\n", irqst); + + if (irqst & RSYNCERREN) + dev_err(mcbsp->dev, "RX Frame Sync Error!\n"); + if (irqst & RFSREN) + dev_dbg(mcbsp->dev, "RX Frame Sync\n"); + if (irqst & REOFEN) + dev_dbg(mcbsp->dev, "RX End Of Frame\n"); + if (irqst & RRDYEN) + dev_dbg(mcbsp->dev, "RX Buffer Threshold Reached\n"); + if (irqst & RUNDFLEN) + dev_err(mcbsp->dev, "RX Buffer Underflow!\n"); + if (irqst & ROVFLEN) + dev_err(mcbsp->dev, "RX Buffer Overflow!\n"); + + if (irqst & XSYNCERREN) + dev_err(mcbsp->dev, "TX Frame Sync Error!\n"); + if (irqst & XFSXEN) + dev_dbg(mcbsp->dev, "TX Frame Sync\n"); + if (irqst & XEOFEN) + dev_dbg(mcbsp->dev, "TX End Of Frame\n"); + if (irqst & XRDYEN) + dev_dbg(mcbsp->dev, "TX Buffer threshold Reached\n"); + if (irqst & XUNDFLEN) + dev_err(mcbsp->dev, "TX Buffer Underflow!\n"); + if (irqst & XOVFLEN) + dev_err(mcbsp->dev, "TX Buffer Overflow!\n"); + if (irqst & XEMPTYEOFEN) + dev_dbg(mcbsp->dev, "TX Buffer empty at end of frame\n"); + + MCBSP_WRITE(mcbsp, IRQST, irqst); + + return IRQ_HANDLED; +} + static irqreturn_t omap_mcbsp_tx_irq_handler(int irq, void *dev_id) { struct omap_mcbsp *mcbsp_tx = dev_id; @@ -176,6 +217,10 @@ void omap_mcbsp_config(struct omap_mcbsp *mcbsp, /* Enable wakeup behavior */ if (mcbsp->pdata->has_wakeup) MCBSP_WRITE(mcbsp, WAKEUPEN, XRDYEN | RRDYEN); + + /* Enable TX/RX sync error interrupts by default */ + if (mcbsp->irq) + MCBSP_WRITE(mcbsp, IRQEN, RSYNCERREN | XSYNCERREN); } /** @@ -489,23 +534,25 @@ int omap_mcbsp_request(struct omap_mcbsp *mcbsp) MCBSP_WRITE(mcbsp, SPCR1, 0); MCBSP_WRITE(mcbsp, SPCR2, 0); - err = request_irq(mcbsp->tx_irq, omap_mcbsp_tx_irq_handler, - 0, "McBSP", (void *)mcbsp); - if (err != 0) { - dev_err(mcbsp->dev, "Unable to request TX IRQ %d " - "for McBSP%d\n", mcbsp->tx_irq, - mcbsp->id); - goto err_clk_disable; - } + if (mcbsp->irq) { + err = request_irq(mcbsp->irq, omap_mcbsp_irq_handler, 0, + "McBSP", (void *)mcbsp); + if (err != 0) { + dev_err(mcbsp->dev, "Unable to request IRQ\n"); + goto err_clk_disable; + } + } else { + err = request_irq(mcbsp->tx_irq, omap_mcbsp_tx_irq_handler, 0, + "McBSP TX", (void *)mcbsp); + if (err != 0) { + dev_err(mcbsp->dev, "Unable to request TX IRQ\n"); + goto err_clk_disable; + } - if (mcbsp->rx_irq) { - err = request_irq(mcbsp->rx_irq, - omap_mcbsp_rx_irq_handler, - 0, "McBSP", (void *)mcbsp); + err = request_irq(mcbsp->rx_irq, omap_mcbsp_rx_irq_handler, 0, + "McBSP RX", (void *)mcbsp); if (err != 0) { - dev_err(mcbsp->dev, "Unable to request RX IRQ %d " - "for McBSP%d\n", mcbsp->rx_irq, - mcbsp->id); + dev_err(mcbsp->dev, "Unable to request RX IRQ\n"); goto err_free_irq; } } @@ -542,9 +589,16 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) if (mcbsp->pdata->has_wakeup) MCBSP_WRITE(mcbsp, WAKEUPEN, 0); - if (mcbsp->rx_irq) + /* Disable interrupt requests */ + if (mcbsp->irq) + MCBSP_WRITE(mcbsp, IRQEN, 0); + + if (mcbsp->irq) { + free_irq(mcbsp->irq, (void *)mcbsp); + } else { free_irq(mcbsp->rx_irq, (void *)mcbsp); - free_irq(mcbsp->tx_irq, (void *)mcbsp); + free_irq(mcbsp->tx_irq, (void *)mcbsp); + } reg_cache = mcbsp->reg_cache; @@ -754,7 +808,7 @@ THRESHOLD_PROP_BUILDER(max_tx_thres); THRESHOLD_PROP_BUILDER(max_rx_thres); static const char *dma_op_modes[] = { - "element", "threshold", "frame", + "element", "threshold", }; static ssize_t dma_op_mode_show(struct device *dev, @@ -949,13 +1003,24 @@ int __devinit omap_mcbsp_init(struct platform_device *pdev) else mcbsp->phys_dma_base = res->start; - mcbsp->tx_irq = platform_get_irq_byname(pdev, "tx"); - mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx"); - - /* From OMAP4 there will be a single irq line */ - if (mcbsp->tx_irq == -ENXIO) { - mcbsp->tx_irq = platform_get_irq(pdev, 0); - mcbsp->rx_irq = 0; + /* + * OMAP1, 2 uses two interrupt lines: TX, RX + * OMAP2430, OMAP3 SoC have combined IRQ line as well. + * OMAP4 and newer SoC only have the combined IRQ line. + * Use the combined IRQ if available since it gives better debugging + * possibilities. + */ + mcbsp->irq = platform_get_irq_byname(pdev, "common"); + if (mcbsp->irq == -ENXIO) { + mcbsp->tx_irq = platform_get_irq_byname(pdev, "tx"); + + if (mcbsp->tx_irq == -ENXIO) { + mcbsp->irq = platform_get_irq(pdev, 0); + mcbsp->tx_irq = 0; + } else { + mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx"); + mcbsp->irq = 0; + } } res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index a944fcc9073c..262a6152111f 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -217,17 +217,20 @@ enum { /********************** McBSP DMA operating modes **************************/ #define MCBSP_DMA_MODE_ELEMENT 0 #define MCBSP_DMA_MODE_THRESHOLD 1 -#define MCBSP_DMA_MODE_FRAME 2 -/********************** McBSP WAKEUPEN bit definitions *********************/ +/********************** McBSP WAKEUPEN/IRQST/IRQEN bit definitions *********/ #define RSYNCERREN BIT(0) #define RFSREN BIT(1) #define REOFEN BIT(2) #define RRDYEN BIT(3) +#define RUNDFLEN BIT(4) +#define ROVFLEN BIT(5) #define XSYNCERREN BIT(7) #define XFSXEN BIT(8) #define XEOFEN BIT(9) #define XRDYEN BIT(10) +#define XUNDFLEN BIT(11) +#define XOVFLEN BIT(12) #define XEMPTYEOFEN BIT(14) /* Clock signal muxing options */ @@ -295,6 +298,7 @@ struct omap_mcbsp { int configured; u8 free; + int irq; int rx_irq; int tx_irq; diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 93bb8eee22b3..9d93793d3077 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -40,6 +40,11 @@ #include "omap-pcm.h" #include "../codecs/twl6040.h" +struct abe_twl6040 { + int jack_detection; /* board can detect jack events */ + int mclk_freq; /* MCLK frequency speed for twl6040 */ +}; + static int omap_abe_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -47,13 +52,13 @@ static int omap_abe_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = codec->card; - struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev); + struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); int clk_id, freq; int ret; clk_id = twl6040_get_clk_id(rtd->codec); if (clk_id == TWL6040_SYSCLK_SEL_HPPLL) - freq = pdata->mclk_freq; + freq = priv->mclk_freq; else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL) freq = 32768; else @@ -128,6 +133,9 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_MIC("Main Handset Mic", NULL), SND_SOC_DAPM_MIC("Sub Handset Mic", NULL), SND_SOC_DAPM_LINE("Line In", NULL), + + /* Digital microphones */ + SND_SOC_DAPM_MIC("Digital Mic", NULL), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -173,6 +181,7 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_card *card = codec->card; struct snd_soc_dapm_context *dapm = &codec->dapm; struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev); + struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); int hs_trim; int ret = 0; @@ -196,7 +205,7 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) TWL6040_HSF_TRIM_RIGHT(hs_trim)); /* Headset jack detection only if it is supported */ - if (pdata->jack_detection) { + if (priv->jack_detection) { ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, &hs_jack); if (ret) @@ -210,10 +219,6 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) return ret; } -static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Digital Mic", NULL), -}; - static const struct snd_soc_dapm_route dmic_audio_map[] = { {"DMic", NULL, "Digital Mic"}, {"Digital Mic", NULL, "Digital Mic1 Bias"}, @@ -223,19 +228,13 @@ static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - ret = snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets, - ARRAY_SIZE(dmic_dapm_widgets)); - if (ret) - return ret; return snd_soc_dapm_add_routes(dapm, dmic_audio_map, ARRAY_SIZE(dmic_audio_map)); } /* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link twl6040_dmic_dai[] = { +static struct snd_soc_dai_link abe_twl6040_dai_links[] = { { .name = "TWL6040", .stream_name = "TWL6040", @@ -258,19 +257,6 @@ static struct snd_soc_dai_link twl6040_dmic_dai[] = { }, }; -static struct snd_soc_dai_link twl6040_only_dai[] = { - { - .name = "TWL6040", - .stream_name = "TWL6040", - .cpu_dai_name = "omap-mcpdm", - .codec_dai_name = "twl6040-legacy", - .platform_name = "omap-pcm-audio", - .codec_name = "twl6040-codec", - .init = omap_abe_twl6040_init, - .ops = &omap_abe_ops, - }, -}; - /* Audio machine driver */ static struct snd_soc_card omap_abe_card = { .owner = THIS_MODULE, @@ -285,6 +271,8 @@ static __devinit int omap_abe_probe(struct platform_device *pdev) { struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev); struct snd_soc_card *card = &omap_abe_card; + struct abe_twl6040 *priv; + int num_links = 0; int ret; card->dev = &pdev->dev; @@ -294,6 +282,10 @@ static __devinit int omap_abe_probe(struct platform_device *pdev) return -ENODEV; } + priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + if (pdata->card_name) { card->name = pdata->card_name; } else { @@ -301,18 +293,24 @@ static __devinit int omap_abe_probe(struct platform_device *pdev) return -ENODEV; } - if (!pdata->mclk_freq) { + priv->jack_detection = pdata->jack_detection; + priv->mclk_freq = pdata->mclk_freq; + + + if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency missing\n"); return -ENODEV; } - if (pdata->has_dmic) { - card->dai_link = twl6040_dmic_dai; - card->num_links = ARRAY_SIZE(twl6040_dmic_dai); - } else { - card->dai_link = twl6040_only_dai; - card->num_links = ARRAY_SIZE(twl6040_only_dai); - } + if (pdata->has_dmic) + num_links = 2; + else + num_links = 1; + + card->dai_link = abe_twl6040_dai_links; + card->num_links = num_links; + + snd_soc_card_set_drvdata(card, priv); ret = snd_soc_register_card(card); if (ret) diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 4dcb5a7e40e8..75f5dca0e8d2 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -32,6 +32,7 @@ #include <linux/io.h> #include <linux/slab.h> #include <linux/pm_runtime.h> +#include <linux/of_device.h> #include <plat/dma.h> #include <sound/core.h> @@ -528,10 +529,17 @@ static int __devexit asoc_dmic_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id omap_dmic_of_match[] = { + { .compatible = "ti,omap4-dmic", }, + { } +}; +MODULE_DEVICE_TABLE(of, omap_dmic_of_match); + static struct platform_driver asoc_dmic_driver = { .driver = { .name = "omap-dmic", .owner = THIS_MODULE, + .of_match_table = omap_dmic_of_match, }, .probe = asoc_dmic_probe, .remove = __devexit_p(asoc_dmic_remove), diff --git a/sound/soc/omap/omap-hdmi-card.c b/sound/soc/omap/omap-hdmi-card.c new file mode 100644 index 000000000000..eaa2ea0e3f81 --- /dev/null +++ b/sound/soc/omap/omap-hdmi-card.c @@ -0,0 +1,87 @@ +/* + * omap-hdmi-card.c + * + * OMAP ALSA SoC machine driver for TI OMAP HDMI + * Copyright (C) 2011 Texas Instruments Incorporated - http://www.ti.com/ + * Author: Ricardo Neri <ricardo.neri@ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <asm/mach-types.h> +#include <video/omapdss.h> + +#define DRV_NAME "omap-hdmi-audio" + +static struct snd_soc_dai_link omap_hdmi_dai = { + .name = "HDMI", + .stream_name = "HDMI", + .cpu_dai_name = "omap-hdmi-audio-dai", + .platform_name = "omap-pcm-audio", + .codec_name = "hdmi-audio-codec", + .codec_dai_name = "omap-hdmi-hifi", +}; + +static struct snd_soc_card snd_soc_omap_hdmi = { + .name = "OMAPHDMI", + .owner = THIS_MODULE, + .dai_link = &omap_hdmi_dai, + .num_links = 1, +}; + +static __devinit int omap_hdmi_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_omap_hdmi; + int ret; + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + card->dev = NULL; + return ret; + } + return 0; +} + +static int __devexit omap_hdmi_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + card->dev = NULL; + return 0; +} + +static struct platform_driver omap_hdmi_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + .probe = omap_hdmi_probe, + .remove = __devexit_p(omap_hdmi_remove), +}; + +module_platform_driver(omap_hdmi_driver); + +MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>"); +MODULE_DESCRIPTION("OMAP HDMI machine ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c index 38e0defa7078..a08245d9203c 100644 --- a/sound/soc/omap/omap-hdmi.c +++ b/sound/soc/omap/omap-hdmi.c @@ -30,21 +30,28 @@ #include <sound/pcm_params.h> #include <sound/initval.h> #include <sound/soc.h> +#include <sound/asound.h> +#include <sound/asoundef.h> +#include <video/omapdss.h> #include <plat/dma.h> #include "omap-pcm.h" #include "omap-hdmi.h" -#define DRV_NAME "hdmi-audio-dai" +#define DRV_NAME "omap-hdmi-audio-dai" -static struct omap_pcm_dma_data omap_hdmi_dai_dma_params = { - .name = "HDMI playback", - .sync_mode = OMAP_DMA_SYNC_PACKET, +struct hdmi_priv { + struct omap_pcm_dma_data dma_params; + struct omap_dss_audio dss_audio; + struct snd_aes_iec958 iec; + struct snd_cea_861_aud_if cea; + struct omap_dss_device *dssdev; }; static int omap_hdmi_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { + struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai); int err; /* * Make sure that the period bytes are multiple of the DMA packet size. @@ -52,46 +59,201 @@ static int omap_hdmi_dai_startup(struct snd_pcm_substream *substream, */ err = snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 128); - if (err < 0) + if (err < 0) { + dev_err(dai->dev, "could not apply constraint\n"); return err; + } + if (!priv->dssdev->driver->audio_supported(priv->dssdev)) { + dev_err(dai->dev, "audio not supported\n"); + return -ENODEV; + } return 0; } +static int omap_hdmi_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai); + + return priv->dssdev->driver->audio_enable(priv->dssdev); +} + static int omap_hdmi_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai); + struct snd_aes_iec958 *iec = &priv->iec; + struct snd_cea_861_aud_if *cea = &priv->cea; int err = 0; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - omap_hdmi_dai_dma_params.packet_size = 16; + priv->dma_params.packet_size = 16; break; case SNDRV_PCM_FORMAT_S24_LE: - omap_hdmi_dai_dma_params.packet_size = 32; + priv->dma_params.packet_size = 32; break; default: - err = -EINVAL; + dev_err(dai->dev, "format not supported!\n"); + return -EINVAL; } - omap_hdmi_dai_dma_params.data_type = OMAP_DMA_DATA_TYPE_S32; + priv->dma_params.data_type = OMAP_DMA_DATA_TYPE_S32; snd_soc_dai_set_dma_data(dai, substream, - &omap_hdmi_dai_dma_params); + &priv->dma_params); + + /* + * fill the IEC-60958 channel status word + */ + + /* specify IEC-60958-3 (commercial use) */ + iec->status[0] &= ~IEC958_AES0_PROFESSIONAL; + + /* specify that the audio is LPCM*/ + iec->status[0] &= ~IEC958_AES0_NONAUDIO; + + iec->status[0] |= IEC958_AES0_CON_NOT_COPYRIGHT; + + iec->status[0] |= IEC958_AES0_CON_EMPHASIS_NONE; + + iec->status[0] |= IEC958_AES1_PRO_MODE_NOTID; + + iec->status[1] = IEC958_AES1_CON_GENERAL; + + iec->status[2] |= IEC958_AES2_CON_SOURCE_UNSPEC; + + iec->status[2] |= IEC958_AES2_CON_CHANNEL_UNSPEC; + + switch (params_rate(params)) { + case 32000: + iec->status[3] |= IEC958_AES3_CON_FS_32000; + break; + case 44100: + iec->status[3] |= IEC958_AES3_CON_FS_44100; + break; + case 48000: + iec->status[3] |= IEC958_AES3_CON_FS_48000; + break; + case 88200: + iec->status[3] |= IEC958_AES3_CON_FS_88200; + break; + case 96000: + iec->status[3] |= IEC958_AES3_CON_FS_96000; + break; + case 176400: + iec->status[3] |= IEC958_AES3_CON_FS_176400; + break; + case 192000: + iec->status[3] |= IEC958_AES3_CON_FS_192000; + break; + default: + dev_err(dai->dev, "rate not supported!\n"); + return -EINVAL; + } + + /* specify the clock accuracy */ + iec->status[3] |= IEC958_AES3_CON_CLOCK_1000PPM; + + /* + * specify the word length. The same word length value can mean + * two different lengths. Hence, we need to specify the maximum + * word length as well. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + iec->status[4] |= IEC958_AES4_CON_WORDLEN_20_16; + iec->status[4] &= ~IEC958_AES4_CON_MAX_WORDLEN_24; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iec->status[4] |= IEC958_AES4_CON_WORDLEN_24_20; + iec->status[4] |= IEC958_AES4_CON_MAX_WORDLEN_24; + break; + default: + dev_err(dai->dev, "format not supported!\n"); + return -EINVAL; + } + + /* + * Fill the CEA-861 audio infoframe (see spec for details) + */ + + cea->db1_ct_cc = (params_channels(params) - 1) + & CEA861_AUDIO_INFOFRAME_DB1CC; + cea->db1_ct_cc |= CEA861_AUDIO_INFOFRAME_DB1CT_FROM_STREAM; + + cea->db2_sf_ss = CEA861_AUDIO_INFOFRAME_DB2SF_FROM_STREAM; + cea->db2_sf_ss |= CEA861_AUDIO_INFOFRAME_DB2SS_FROM_STREAM; + + cea->db3 = 0; /* not used, all zeros */ + + /* + * The OMAP HDMI IP requires to use the 8-channel channel code when + * transmitting more than two channels. + */ + if (params_channels(params) == 2) + cea->db4_ca = 0x0; + else + cea->db4_ca = 0x13; + + cea->db5_dminh_lsv = CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PROHIBITED; + /* the expression is trivial but makes clear what we are doing */ + cea->db5_dminh_lsv |= (0 & CEA861_AUDIO_INFOFRAME_DB5_LSV); + + priv->dss_audio.iec = iec; + priv->dss_audio.cea = cea; + + err = priv->dssdev->driver->audio_config(priv->dssdev, + &priv->dss_audio); return err; } +static int omap_hdmi_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai); + int err = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + err = priv->dssdev->driver->audio_start(priv->dssdev); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + priv->dssdev->driver->audio_stop(priv->dssdev); + break; + default: + err = -EINVAL; + } + return err; +} + +static void omap_hdmi_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai); + + priv->dssdev->driver->audio_disable(priv->dssdev); +} + static const struct snd_soc_dai_ops omap_hdmi_dai_ops = { .startup = omap_hdmi_dai_startup, .hw_params = omap_hdmi_dai_hw_params, + .prepare = omap_hdmi_dai_prepare, + .trigger = omap_hdmi_dai_trigger, + .shutdown = omap_hdmi_dai_shutdown, }; static struct snd_soc_dai_driver omap_hdmi_dai = { .playback = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = OMAP_HDMI_RATES, .formats = OMAP_HDMI_FORMATS, }, @@ -102,31 +264,77 @@ static __devinit int omap_hdmi_probe(struct platform_device *pdev) { int ret; struct resource *hdmi_rsrc; + struct hdmi_priv *hdmi_data; + bool hdmi_dev_found = false; + + hdmi_data = devm_kzalloc(&pdev->dev, sizeof(*hdmi_data), GFP_KERNEL); + if (hdmi_data == NULL) { + dev_err(&pdev->dev, "Cannot allocate memory for HDMI data\n"); + return -ENOMEM; + } hdmi_rsrc = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!hdmi_rsrc) { dev_err(&pdev->dev, "Cannot obtain IORESOURCE_MEM HDMI\n"); - return -EINVAL; + return -ENODEV; } - omap_hdmi_dai_dma_params.port_addr = hdmi_rsrc->start + hdmi_data->dma_params.port_addr = hdmi_rsrc->start + OMAP_HDMI_AUDIO_DMA_PORT; hdmi_rsrc = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!hdmi_rsrc) { dev_err(&pdev->dev, "Cannot obtain IORESOURCE_DMA HDMI\n"); - return -EINVAL; + return -ENODEV; } - omap_hdmi_dai_dma_params.dma_req = hdmi_rsrc->start; + hdmi_data->dma_params.dma_req = hdmi_rsrc->start; + hdmi_data->dma_params.name = "HDMI playback"; + hdmi_data->dma_params.sync_mode = OMAP_DMA_SYNC_PACKET; + + /* + * TODO: We assume that there is only one DSS HDMI device. Future + * OMAP implementations may support more than one HDMI devices and + * we should provided separate audio support for all of them. + */ + /* Find an HDMI device. */ + for_each_dss_dev(hdmi_data->dssdev) { + omap_dss_get_device(hdmi_data->dssdev); + if (!hdmi_data->dssdev->driver) { + omap_dss_put_device(hdmi_data->dssdev); + continue; + } + + if (hdmi_data->dssdev->type == OMAP_DISPLAY_TYPE_HDMI) { + hdmi_dev_found = true; + break; + } + } + + if (!hdmi_dev_found) { + dev_err(&pdev->dev, "no driver for HDMI display found\n"); + return -ENODEV; + } + + dev_set_drvdata(&pdev->dev, hdmi_data); ret = snd_soc_register_dai(&pdev->dev, &omap_hdmi_dai); + return ret; } static int __devexit omap_hdmi_remove(struct platform_device *pdev) { + struct hdmi_priv *hdmi_data = dev_get_drvdata(&pdev->dev); + snd_soc_unregister_dai(&pdev->dev); + + if (hdmi_data == NULL) { + dev_err(&pdev->dev, "cannot obtain HDMi data\n"); + return -ENODEV; + } + + omap_dss_put_device(hdmi_data->dssdev); return 0; } diff --git a/sound/soc/omap/omap-hdmi.h b/sound/soc/omap/omap-hdmi.h index 34c298d5057e..6ad2bf4f2697 100644 --- a/sound/soc/omap/omap-hdmi.h +++ b/sound/soc/omap/omap-hdmi.h @@ -28,7 +28,9 @@ #define OMAP_HDMI_AUDIO_DMA_PORT 0x8c #define OMAP_HDMI_RATES (SNDRV_PCM_RATE_32000 | \ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000) #define OMAP_HDMI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 6912ac7cb625..1046083e90a0 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -71,18 +71,17 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream) dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */ - if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) - /* - * Configure McBSP threshold based on either: - * packet_size, when the sDMA is in packet mode, or - * based on the period size. - */ - if (dma_data->packet_size) - words = dma_data->packet_size; - else - words = snd_pcm_lib_period_bytes(substream) / - (mcbsp->wlen / 8); + /* + * Configure McBSP threshold based on either: + * packet_size, when the sDMA is in packet mode, or based on the + * period size in THRESHOLD mode, otherwise use McBSP threshold = 1 + * for mono streams. + */ + if (dma_data->packet_size) + words = dma_data->packet_size; + else if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) + words = snd_pcm_lib_period_bytes(substream) / + (mcbsp->wlen / 8); else words = 1; @@ -139,13 +138,15 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, if (mcbsp->pdata->buffer_size) { /* * Rule for the buffer size. We should not allow - * smaller buffer than the FIFO size to avoid underruns + * smaller buffer than the FIFO size to avoid underruns. + * This applies only for the playback stream. */ - snd_pcm_hw_rule_add(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, - omap_mcbsp_hwrule_min_buffersize, - mcbsp, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + omap_mcbsp_hwrule_min_buffersize, + mcbsp, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); /* Make sure, that the period size is always even */ snd_pcm_hw_constraint_step(substream->runtime, 0, @@ -230,6 +231,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, unsigned int format, div, framesize, master; dma_data = &mcbsp->dma_data[substream->stream]; + channels = params_channels(params); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -245,7 +247,6 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, } if (mcbsp->pdata->buffer_size) { dma_data->set_threshold = omap_mcbsp_set_threshold; - /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */ if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) { int period_words, max_thrsh; @@ -283,6 +284,10 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, } else { sync_mode = OMAP_DMA_SYNC_FRAME; } + } else if (channels > 1) { + /* Use packet mode for non mono streams */ + pkt_size = channels; + sync_mode = OMAP_DMA_SYNC_PACKET; } } @@ -301,7 +306,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, regs->rcr1 &= ~(RFRLEN1(0x7f) | RWDLEN1(7)); regs->xcr1 &= ~(XFRLEN1(0x7f) | XWDLEN1(7)); format = mcbsp->fmt & SND_SOC_DAIFMT_FORMAT_MASK; - wpf = channels = params_channels(params); + wpf = channels; if (channels == 2 && (format == SND_SOC_DAIFMT_I2S || format == SND_SOC_DAIFMT_LEFT_J)) { /* Use dual-phase frames */ diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 39705561131a..2c66e2498a45 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -33,6 +33,7 @@ #include <linux/irq.h> #include <linux/slab.h> #include <linux/pm_runtime.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> @@ -507,10 +508,17 @@ static int __devexit asoc_mcpdm_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id omap_mcpdm_of_match[] = { + { .compatible = "ti,omap4-mcpdm", }, + { } +}; +MODULE_DEVICE_TABLE(of, omap_mcpdm_of_match); + static struct platform_driver asoc_mcpdm_driver = { .driver = { .name = "omap-mcpdm", .owner = THIS_MODULE, + .of_match_table = omap_mcpdm_of_match, }, .probe = asoc_mcpdm_probe, @@ -519,6 +527,7 @@ static struct platform_driver asoc_mcpdm_driver = { module_platform_driver(asoc_mcpdm_driver); +MODULE_ALIAS("platform:omap-mcpdm"); MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>"); MODULE_DESCRIPTION("OMAP PDM SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap4-hdmi-card.c b/sound/soc/omap/omap4-hdmi-card.c deleted file mode 100644 index 28d689b2714d..000000000000 --- a/sound/soc/omap/omap4-hdmi-card.c +++ /dev/null @@ -1,121 +0,0 @@ -/* - * omap4-hdmi-card.c - * - * OMAP ALSA SoC machine driver for TI OMAP4 HDMI - * Copyright (C) 2011 Texas Instruments Incorporated - http://www.ti.com/ - * Author: Ricardo Neri <ricardo.neri@ti.com> - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include <linux/module.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <asm/mach-types.h> -#include <video/omapdss.h> - -#define DRV_NAME "omap4-hdmi-audio" - -static int omap4_hdmi_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - int i; - struct omap_overlay_manager *mgr = NULL; - struct device *dev = substream->pcm->card->dev; - - /* Find DSS HDMI device */ - for (i = 0; i < omap_dss_get_num_overlay_managers(); i++) { - mgr = omap_dss_get_overlay_manager(i); - if (mgr && mgr->device - && mgr->device->type == OMAP_DISPLAY_TYPE_HDMI) - break; - } - - if (i == omap_dss_get_num_overlay_managers()) { - dev_err(dev, "HDMI display device not found!\n"); - return -ENODEV; - } - - /* Make sure HDMI is power-on to avoid L3 interconnect errors */ - if (mgr->device->state != OMAP_DSS_DISPLAY_ACTIVE) { - dev_err(dev, "HDMI display is not active!\n"); - return -EIO; - } - - return 0; -} - -static struct snd_soc_ops omap4_hdmi_dai_ops = { - .hw_params = omap4_hdmi_dai_hw_params, -}; - -static struct snd_soc_dai_link omap4_hdmi_dai = { - .name = "HDMI", - .stream_name = "HDMI", - .cpu_dai_name = "hdmi-audio-dai", - .platform_name = "omap-pcm-audio", - .codec_name = "omapdss_hdmi", - .codec_dai_name = "hdmi-audio-codec", - .ops = &omap4_hdmi_dai_ops, -}; - -static struct snd_soc_card snd_soc_omap4_hdmi = { - .name = "OMAP4HDMI", - .owner = THIS_MODULE, - .dai_link = &omap4_hdmi_dai, - .num_links = 1, -}; - -static __devinit int omap4_hdmi_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &snd_soc_omap4_hdmi; - int ret; - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - card->dev = NULL; - return ret; - } - return 0; -} - -static int __devexit omap4_hdmi_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - card->dev = NULL; - return 0; -} - -static struct platform_driver omap4_hdmi_driver = { - .driver = { - .name = "omap4-hdmi-audio", - .owner = THIS_MODULE, - }, - .probe = omap4_hdmi_probe, - .remove = __devexit_p(omap4_hdmi_remove), -}; - -module_platform_driver(omap4_hdmi_driver); - -MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>"); -MODULE_DESCRIPTION("OMAP4 HDMI machine ASoC driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index a0f7d3cfa470..4d2e46fae77c 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -8,6 +8,15 @@ config SND_PXA2XX_SOC the PXA2xx AC97, I2S or SSP interface. You will also need to select the audio interfaces to support below. +config SND_MMP_SOC + bool "Soc Audio for Marvell MMP chips" + depends on ARCH_MMP + select SND_SOC_DMAENGINE_PCM + select SND_ARM + help + Say Y if you want to add support for codecs attached to + the MMP SSPA interface. + config SND_PXA2XX_AC97 tristate select SND_AC97_CODEC @@ -26,6 +35,9 @@ config SND_PXA_SOC_SSP tristate select PXA_SSP +config SND_MMP_SOC_SSPA + tristate + config SND_PXA2XX_SOC_CORGI tristate "SoC Audio support for Sharp Zaurus SL-C7x0" depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx @@ -138,6 +150,26 @@ config SND_SOC_TAVOREVB3 Say Y if you want to add support for SoC audio on the Marvell Saarb reference platform. +config SND_PXA910_SOC + tristate "SoC Audio for Marvell PXA910 chip" + depends on ARCH_MMP && SND + select SND_PCM + help + Say Y if you want to add support for SoC audio on the + Marvell PXA910 reference platform. + +config SND_SOC_TTC_DKB + bool "SoC Audio support for TTC DKB" + depends on SND_PXA910_SOC && MACH_TTC_DKB + select PXA_SSP + select SND_PXA_SOC_SSP + select SND_MMP_SOC + select MFD_88PM860X + select SND_SOC_88PM860X + help + Say Y if you want to add support for SoC audio on TTC DKB + + config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE @@ -194,3 +226,13 @@ config SND_PXA2XX_SOC_IMOTE2 help Say Y if you want to add support for SoC audio on the IMote 2. + +config SND_MMP_SOC_BROWNSTONE + tristate "SoC Audio support for Marvell Brownstone" + depends on SND_MMP_SOC && MACH_BROWNSTONE + select SND_MMP_SOC_SSPA + select MFD_WM8994 + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on the + Marvell Brownstone reference platform. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index af357623be9d..d8a265d2d5d7 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -3,11 +3,15 @@ snd-soc-pxa2xx-objs := pxa2xx-pcm.o snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o snd-soc-pxa-ssp-objs := pxa-ssp.o +snd-soc-mmp-objs := mmp-pcm.o +snd-soc-mmp-sspa-objs := mmp-sspa.o obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o +obj-$(CONFIG_SND_MMP_SOC) += snd-soc-mmp.o +obj-$(CONFIG_SND_MMP_SOC_SSPA) += snd-soc-mmp-sspa.o # PXA Machine Support snd-soc-corgi-objs := corgi.o @@ -28,6 +32,8 @@ snd-soc-mioa701-objs := mioa701_wm9713.o snd-soc-z2-objs := z2.o snd-soc-imote2-objs := imote2.o snd-soc-raumfeld-objs := raumfeld.o +snd-soc-brownstone-objs := brownstone.o +snd-soc-ttc-dkb-objs := ttc-dkb.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -47,3 +53,5 @@ obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o +obj-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o +obj-$(CONFIG_SND_SOC_TTC_DKB) += snd-soc-ttc-dkb.o diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c new file mode 100644 index 000000000000..5e666e03d333 --- /dev/null +++ b/sound/soc/pxa/brownstone.c @@ -0,0 +1,174 @@ +/* + * linux/sound/soc/pxa/brownstone.c + * + * Copyright (C) 2011 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + */ + +#include <linux/module.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> + +#include "../codecs/wm8994.h" +#include "mmp-sspa.h" + +static const struct snd_kcontrol_new brownstone_dapm_control[] = { + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static const struct snd_soc_dapm_widget brownstone_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), +}; + +static const struct snd_soc_dapm_route brownstone_audio_map[] = { + {"Ext Spk", NULL, "SPKOUTLP"}, + {"Ext Spk", NULL, "SPKOUTLN"}, + {"Ext Spk", NULL, "SPKOUTRP"}, + {"Ext Spk", NULL, "SPKOUTRN"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + + {"DMIC1DAT", NULL, "MICBIAS1"}, + {"MICBIAS1", NULL, "Main Mic"}, +}; + +static int brownstone_wm8994_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); + snd_soc_dapm_enable_pin(dapm, "Main Mic"); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(dapm, "HPOUT2P"); + snd_soc_dapm_nc_pin(dapm, "HPOUT2N"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT1N"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT1P"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT2N"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT2P"); + snd_soc_dapm_nc_pin(dapm, "IN1LN"); + snd_soc_dapm_nc_pin(dapm, "IN1LP"); + snd_soc_dapm_nc_pin(dapm, "IN1RP"); + snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(dapm, "IN2RN"); + snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(dapm, "IN2LN"); + + snd_soc_dapm_sync(dapm); + + return 0; +} + +static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int freq_out, sspa_mclk, sysclk; + int sspa_div; + + if (params_rate(params) > 11025) { + freq_out = params_rate(params) * 512; + sysclk = params_rate(params) * 256; + sspa_mclk = params_rate(params) * 64; + } else { + freq_out = params_rate(params) * 1024; + sysclk = params_rate(params) * 512; + sspa_mclk = params_rate(params) * 64; + } + sspa_div = freq_out; + do_div(sspa_div, sspa_mclk); + + snd_soc_dai_set_sysclk(cpu_dai, MMP_SSPA_CLK_AUDIO, freq_out, 0); + snd_soc_dai_set_pll(cpu_dai, MMP_SYSCLK, 0, freq_out, sysclk); + snd_soc_dai_set_pll(cpu_dai, MMP_SSPA_CLK, 0, freq_out, sspa_mclk); + + /* set wm8994 sysclk */ + snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK1, sysclk, 0); + + return 0; +} + +/* machine stream operations */ +static struct snd_soc_ops brownstone_ops = { + .hw_params = brownstone_wm8994_hw_params, +}; + +static struct snd_soc_dai_link brownstone_wm8994_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + .cpu_dai_name = "mmp-sspa-dai.0", + .codec_dai_name = "wm8994-aif1", + .platform_name = "mmp-pcm-audio", + .codec_name = "wm8994-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ops = &brownstone_ops, + .init = brownstone_wm8994_init, +}, +}; + +/* audio machine driver */ +static struct snd_soc_card brownstone = { + .name = "brownstone", + .dai_link = brownstone_wm8994_dai, + .num_links = ARRAY_SIZE(brownstone_wm8994_dai), + + .controls = brownstone_dapm_control, + .num_controls = ARRAY_SIZE(brownstone_dapm_control), + .dapm_widgets = brownstone_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(brownstone_dapm_widgets), + .dapm_routes = brownstone_audio_map, + .num_dapm_routes = ARRAY_SIZE(brownstone_audio_map), +}; + +static int __devinit brownstone_probe(struct platform_device *pdev) +{ + int ret; + + brownstone.dev = &pdev->dev; + ret = snd_soc_register_card(&brownstone); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; +} + +static int __devexit brownstone_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&brownstone); + return 0; +} + +static struct platform_driver mmp_driver = { + .driver = { + .name = "brownstone-audio", + .owner = THIS_MODULE, + }, + .probe = brownstone_probe, + .remove = __devexit_p(brownstone_remove), +}; + +module_platform_driver(mmp_driver); + +MODULE_AUTHOR("Leo Yan <leoy@marvell.com>"); +MODULE_DESCRIPTION("ALSA SoC Brownstone"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 9c585af59b5f..8687c1c65d29 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -186,36 +186,27 @@ static struct snd_soc_card mioa701 = { .num_links = ARRAY_SIZE(mioa701_dai), }; -static struct platform_device *mioa701_snd_device; - -static int mioa701_wm9713_probe(struct platform_device *pdev) +static int __devinit mioa701_wm9713_probe(struct platform_device *pdev) { - int ret; + int rc; if (!machine_is_mioa701()) return -ENODEV; - dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will" - "lead to overheating and possible destruction of your device." - "Do not use without a good knowledge of mio's board design!\n"); - - mioa701_snd_device = platform_device_alloc("soc-audio", -1); - if (!mioa701_snd_device) - return -ENOMEM; - - platform_set_drvdata(mioa701_snd_device, &mioa701); - - ret = platform_device_add(mioa701_snd_device); - if (!ret) - return 0; - - platform_device_put(mioa701_snd_device); - return ret; + mioa701.dev = &pdev->dev; + rc = snd_soc_register_card(&mioa701); + if (!rc) + dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will" + "lead to overheating and possible destruction of your device." + " Do not use without a good knowledge of mio's board design!\n"); + return rc; } static int __devexit mioa701_wm9713_remove(struct platform_device *pdev) { - platform_device_unregister(mioa701_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); return 0; } diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c new file mode 100644 index 000000000000..73ac5463c9e4 --- /dev/null +++ b/sound/soc/pxa/mmp-pcm.c @@ -0,0 +1,297 @@ +/* + * linux/sound/soc/pxa/mmp-pcm.c + * + * Copyright (C) 2011 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + */ +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/dmaengine.h> +#include <linux/platform_data/mmp_audio.h> +#include <sound/pxa2xx-lib.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <mach/sram.h> +#include <sound/dmaengine_pcm.h> + +struct mmp_dma_data { + int ssp_id; + struct resource *dma_res; +}; + +#define MMP_PCM_INFO (SNDRV_PCM_INFO_MMAP | \ + SNDRV_PCM_INFO_MMAP_VALID | \ + SNDRV_PCM_INFO_INTERLEAVED | \ + SNDRV_PCM_INFO_PAUSE | \ + SNDRV_PCM_INFO_RESUME) + +#define MMP_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_pcm_hardware mmp_pcm_hardware[] = { + { + .info = MMP_PCM_INFO, + .formats = MMP_PCM_FORMATS, + .period_bytes_min = 1024, + .period_bytes_max = 2048, + .periods_min = 2, + .periods_max = 32, + .buffer_bytes_max = 4096, + .fifo_size = 32, + }, + { + .info = MMP_PCM_INFO, + .formats = MMP_PCM_FORMATS, + .period_bytes_min = 1024, + .period_bytes_max = 2048, + .periods_min = 2, + .periods_max = 32, + .buffer_bytes_max = 4096, + .fifo_size = 32, + }, +}; + +static int mmp_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct pxa2xx_pcm_dma_params *dma_params; + struct dma_slave_config slave_config; + int ret; + + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + if (!dma_params) + return 0; + + ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config); + if (ret) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + slave_config.dst_addr = dma_params->dev_addr; + slave_config.dst_maxburst = 4; + } else { + slave_config.src_addr = dma_params->dev_addr; + slave_config.src_maxburst = 4; + } + + ret = dmaengine_slave_config(chan, &slave_config); + if (ret) + return ret; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static bool filter(struct dma_chan *chan, void *param) +{ + struct mmp_dma_data *dma_data = param; + bool found = false; + char *devname; + + devname = kasprintf(GFP_KERNEL, "%s.%d", dma_data->dma_res->name, + dma_data->ssp_id); + if ((strcmp(dev_name(chan->device->dev), devname) == 0) && + (chan->chan_id == dma_data->dma_res->start)) { + found = true; + } + + kfree(devname); + return found; +} + +static int mmp_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct platform_device *pdev = to_platform_device(rtd->platform->dev); + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct mmp_dma_data *dma_data; + struct resource *r; + int ret; + + r = platform_get_resource(pdev, IORESOURCE_DMA, substream->stream); + if (!r) + return -EBUSY; + + snd_soc_set_runtime_hwparams(substream, + &mmp_pcm_hardware[substream->stream]); + dma_data = devm_kzalloc(&pdev->dev, + sizeof(struct mmp_dma_data), GFP_KERNEL); + if (dma_data == NULL) + return -ENOMEM; + + dma_data->dma_res = r; + dma_data->ssp_id = cpu_dai->id; + + ret = snd_dmaengine_pcm_open(substream, filter, dma_data); + if (ret) { + devm_kfree(&pdev->dev, dma_data); + return ret; + } + + snd_dmaengine_pcm_set_data(substream, dma_data); + return 0; +} + +static int mmp_pcm_close(struct snd_pcm_substream *substream) +{ + struct mmp_dma_data *dma_data = snd_dmaengine_pcm_get_data(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct platform_device *pdev = to_platform_device(rtd->platform->dev); + + snd_dmaengine_pcm_close(substream); + devm_kfree(&pdev->dev, dma_data); + return 0; +} + +static int mmp_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long off = vma->vm_pgoff; + + vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + return remap_pfn_range(vma, vma->vm_start, + __phys_to_pfn(runtime->dma_addr) + off, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +struct snd_pcm_ops mmp_pcm_ops = { + .open = mmp_pcm_open, + .close = mmp_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = mmp_pcm_hw_params, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, + .mmap = mmp_pcm_mmap, +}; + +static void mmp_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + struct gen_pool *gpool; + + gpool = sram_get_gpool("asram"); + if (!gpool) + return; + + for (stream = 0; stream < 2; stream++) { + size_t size = mmp_pcm_hardware[stream].buffer_bytes_max; + + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + gen_pool_free(gpool, (unsigned long)buf->area, size); + buf->area = NULL; + } + + return; +} + +static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream, + int stream) +{ + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = mmp_pcm_hardware[stream].buffer_bytes_max; + struct gen_pool *gpool; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = substream->pcm->card->dev; + buf->private_data = NULL; + + gpool = sram_get_gpool("asram"); + if (!gpool) + return -ENOMEM; + + buf->area = (unsigned char *)gen_pool_alloc(gpool, size); + if (!buf->area) + return -ENOMEM; + buf->addr = gen_pool_virt_to_phys(gpool, (unsigned long)buf->area); + buf->bytes = size; + return 0; +} + +int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm_substream *substream; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0, stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + + ret = mmp_pcm_preallocate_dma_buffer(substream, stream); + if (ret) + goto err; + } + + return 0; + +err: + mmp_pcm_free_dma_buffers(pcm); + return ret; +} + +struct snd_soc_platform_driver mmp_soc_platform = { + .ops = &mmp_pcm_ops, + .pcm_new = mmp_pcm_new, + .pcm_free = mmp_pcm_free_dma_buffers, +}; + +static __devinit int mmp_pcm_probe(struct platform_device *pdev) +{ + struct mmp_audio_platdata *pdata = pdev->dev.platform_data; + + if (pdata) { + mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].buffer_bytes_max = + pdata->buffer_max_playback; + mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].period_bytes_max = + pdata->period_max_playback; + mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].buffer_bytes_max = + pdata->buffer_max_capture; + mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].period_bytes_max = + pdata->period_max_capture; + } + return snd_soc_register_platform(&pdev->dev, &mmp_soc_platform); +} + +static int __devexit mmp_pcm_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver mmp_pcm_driver = { + .driver = { + .name = "mmp-pcm-audio", + .owner = THIS_MODULE, + }, + + .probe = mmp_pcm_probe, + .remove = __devexit_p(mmp_pcm_remove), +}; + +module_platform_driver(mmp_pcm_driver); + +MODULE_AUTHOR("Leo Yan <leoy@marvell.com>"); +MODULE_DESCRIPTION("MMP Soc Audio DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c new file mode 100644 index 000000000000..4d6cb8a30fc8 --- /dev/null +++ b/sound/soc/pxa/mmp-sspa.c @@ -0,0 +1,480 @@ +/* + * linux/sound/soc/pxa/mmp-sspa.c + * Base on pxa2xx-ssp.c + * + * Copyright (C) 2011 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/slab.h> +#include <linux/pxa2xx_ssp.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/pxa2xx-lib.h> +#include "mmp-sspa.h" + +/* + * SSPA audio private data + */ +struct sspa_priv { + struct ssp_device *sspa; + struct pxa2xx_pcm_dma_params *dma_params; + struct clk *audio_clk; + struct clk *sysclk; + int dai_fmt; + int running_cnt; +}; + +static void mmp_sspa_write_reg(struct ssp_device *sspa, u32 reg, u32 val) +{ + __raw_writel(val, sspa->mmio_base + reg); +} + +static u32 mmp_sspa_read_reg(struct ssp_device *sspa, u32 reg) +{ + return __raw_readl(sspa->mmio_base + reg); +} + +static void mmp_sspa_tx_enable(struct ssp_device *sspa) +{ + unsigned int sspa_sp; + + sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP); + sspa_sp |= SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); +} + +static void mmp_sspa_tx_disable(struct ssp_device *sspa) +{ + unsigned int sspa_sp; + + sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP); + sspa_sp &= ~SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); +} + +static void mmp_sspa_rx_enable(struct ssp_device *sspa) +{ + unsigned int sspa_sp; + + sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP); + sspa_sp |= SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp); +} + +static void mmp_sspa_rx_disable(struct ssp_device *sspa) +{ + unsigned int sspa_sp; + + sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP); + sspa_sp &= ~SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp); +} + +static int mmp_sspa_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai); + + clk_enable(priv->sysclk); + clk_enable(priv->sspa->clk); + + return 0; +} + +static void mmp_sspa_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai); + + clk_disable(priv->sspa->clk); + clk_disable(priv->sysclk); + + return; +} + +/* + * Set the SSP ports SYSCLK. + */ +static int mmp_sspa_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + int ret = 0; + + switch (clk_id) { + case MMP_SSPA_CLK_AUDIO: + ret = clk_set_rate(priv->audio_clk, freq); + if (ret) + return ret; + break; + case MMP_SSPA_CLK_PLL: + case MMP_SSPA_CLK_VCXO: + /* not support yet */ + return -EINVAL; + default: + return -EINVAL; + } + + return 0; +} + +static int mmp_sspa_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + int ret = 0; + + switch (pll_id) { + case MMP_SYSCLK: + ret = clk_set_rate(priv->sysclk, freq_out); + if (ret) + return ret; + break; + case MMP_SSPA_CLK: + ret = clk_set_rate(priv->sspa->clk, freq_out); + if (ret) + return ret; + break; + default: + return -ENODEV; + } + + return 0; +} + +/* + * Set up the sspa dai format. The sspa port must be inactive + * before calling this function as the physical + * interface format is changed. + */ +static int mmp_sspa_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct ssp_device *sspa = sspa_priv->sspa; + u32 sspa_sp, sspa_ctrl; + + /* check if we need to change anything at all */ + if (sspa_priv->dai_fmt == fmt) + return 0; + + /* we can only change the settings if the port is not in use */ + if ((mmp_sspa_read_reg(sspa, SSPA_TXSP) & SSPA_SP_S_EN) || + (mmp_sspa_read_reg(sspa, SSPA_RXSP) & SSPA_SP_S_EN)) { + dev_err(&sspa->pdev->dev, + "can't change hardware dai format: stream is in use\n"); + return -EINVAL; + } + + /* reset port settings */ + sspa_sp = SSPA_SP_WEN | SSPA_SP_S_RST | SSPA_SP_FFLUSH; + sspa_ctrl = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + sspa_sp |= SSPA_SP_MSL; + break; + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + sspa_sp |= SSPA_SP_FSP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + sspa_sp |= SSPA_TXSP_FPER(63); + sspa_sp |= SSPA_SP_FWID(31); + sspa_ctrl |= SSPA_CTL_XDATDLY(1); + break; + default: + return -EINVAL; + } + + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); + mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp); + + sspa_sp &= ~(SSPA_SP_S_RST | SSPA_SP_FFLUSH); + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); + mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp); + + /* + * FIXME: hw issue, for the tx serial port, + * can not config the master/slave mode; + * so must clean this bit. + * The master/slave mode has been set in the + * rx port. + */ + sspa_sp &= ~SSPA_SP_MSL; + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); + + mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl); + mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl); + + /* Since we are configuring the timings for the format by hand + * we have to defer some things until hw_params() where we + * know parameters like the sample size. + */ + sspa_priv->dai_fmt = fmt; + return 0; +} + +/* + * Set the SSPA audio DMA parameters and sample size. + * Can be called multiple times by oss emulation. + */ +static int mmp_sspa_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai); + struct ssp_device *sspa = sspa_priv->sspa; + struct pxa2xx_pcm_dma_params *dma_params; + u32 sspa_ctrl; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_TXCTL); + else + sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_RXCTL); + + sspa_ctrl &= ~SSPA_CTL_XFRLEN1_MASK; + sspa_ctrl |= SSPA_CTL_XFRLEN1(params_channels(params) - 1); + sspa_ctrl &= ~SSPA_CTL_XWDLEN1_MASK; + sspa_ctrl |= SSPA_CTL_XWDLEN1(SSPA_CTL_32_BITS); + sspa_ctrl &= ~SSPA_CTL_XSSZ1_MASK; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_8_BITS); + break; + case SNDRV_PCM_FORMAT_S16_LE: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_16_BITS); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_20_BITS); + break; + case SNDRV_PCM_FORMAT_S24_3LE: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_24_BITS); + break; + case SNDRV_PCM_FORMAT_S32_LE: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_32_BITS); + break; + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl); + mmp_sspa_write_reg(sspa, SSPA_TXFIFO_LL, 0x1); + } else { + mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl); + mmp_sspa_write_reg(sspa, SSPA_RXFIFO_UL, 0x0); + } + + dma_params = &sspa_priv->dma_params[substream->stream]; + dma_params->dev_addr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + (sspa->phys_base + SSPA_TXD) : + (sspa->phys_base + SSPA_RXD); + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_params); + return 0; +} + +static int mmp_sspa_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai); + struct ssp_device *sspa = sspa_priv->sspa; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * whatever playback or capture, must enable rx. + * this is a hw issue, so need check if rx has been + * enabled or not; if has been enabled by another + * stream, do not enable again. + */ + if (!sspa_priv->running_cnt) + mmp_sspa_rx_enable(sspa); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mmp_sspa_tx_enable(sspa); + + sspa_priv->running_cnt++; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sspa_priv->running_cnt--; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mmp_sspa_tx_disable(sspa); + + /* have no capture stream, disable rx port */ + if (!sspa_priv->running_cnt) + mmp_sspa_rx_disable(sspa); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static int mmp_sspa_probe(struct snd_soc_dai *dai) +{ + struct sspa_priv *priv = dev_get_drvdata(dai->dev); + + snd_soc_dai_set_drvdata(dai, priv); + return 0; + +} + +#define MMP_SSPA_RATES SNDRV_PCM_RATE_8000_192000 +#define MMP_SSPA_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops mmp_sspa_dai_ops = { + .startup = mmp_sspa_startup, + .shutdown = mmp_sspa_shutdown, + .trigger = mmp_sspa_trigger, + .hw_params = mmp_sspa_hw_params, + .set_sysclk = mmp_sspa_set_dai_sysclk, + .set_pll = mmp_sspa_set_dai_pll, + .set_fmt = mmp_sspa_set_dai_fmt, +}; + +struct snd_soc_dai_driver mmp_sspa_dai = { + .probe = mmp_sspa_probe, + .playback = { + .channels_min = 1, + .channels_max = 128, + .rates = MMP_SSPA_RATES, + .formats = MMP_SSPA_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = MMP_SSPA_RATES, + .formats = MMP_SSPA_FORMATS, + }, + .ops = &mmp_sspa_dai_ops, +}; + +static __devinit int asoc_mmp_sspa_probe(struct platform_device *pdev) +{ + struct sspa_priv *priv; + struct resource *res; + + priv = devm_kzalloc(&pdev->dev, + sizeof(struct sspa_priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->sspa = devm_kzalloc(&pdev->dev, + sizeof(struct ssp_device), GFP_KERNEL); + if (priv->sspa == NULL) + return -ENOMEM; + + priv->dma_params = devm_kzalloc(&pdev->dev, + 2 * sizeof(struct pxa2xx_pcm_dma_params), GFP_KERNEL); + if (priv->dma_params == NULL) + return -ENOMEM; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) + return -ENOMEM; + + priv->sspa->mmio_base = devm_request_and_ioremap(&pdev->dev, res); + if (priv->sspa->mmio_base == NULL) + return -ENODEV; + + priv->sspa->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(priv->sspa->clk)) + return PTR_ERR(priv->sspa->clk); + + priv->audio_clk = clk_get(NULL, "mmp-audio"); + if (IS_ERR(priv->audio_clk)) + return PTR_ERR(priv->audio_clk); + + priv->sysclk = clk_get(NULL, "mmp-sysclk"); + if (IS_ERR(priv->sysclk)) { + clk_put(priv->audio_clk); + return PTR_ERR(priv->sysclk); + } + clk_enable(priv->audio_clk); + priv->dai_fmt = (unsigned int) -1; + platform_set_drvdata(pdev, priv); + + return snd_soc_register_dai(&pdev->dev, &mmp_sspa_dai); +} + +static int __devexit asoc_mmp_sspa_remove(struct platform_device *pdev) +{ + struct sspa_priv *priv = platform_get_drvdata(pdev); + + clk_disable(priv->audio_clk); + clk_put(priv->audio_clk); + clk_put(priv->sysclk); + snd_soc_unregister_dai(&pdev->dev); + return 0; +} + +static struct platform_driver asoc_mmp_sspa_driver = { + .driver = { + .name = "mmp-sspa-dai", + .owner = THIS_MODULE, + }, + .probe = asoc_mmp_sspa_probe, + .remove = __devexit_p(asoc_mmp_sspa_remove), +}; + +module_platform_driver(asoc_mmp_sspa_driver); + +MODULE_AUTHOR("Leo Yan <leoy@marvell.com>"); +MODULE_DESCRIPTION("MMP SSPA SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/mmp-sspa.h b/sound/soc/pxa/mmp-sspa.h new file mode 100644 index 000000000000..ea365cb9e784 --- /dev/null +++ b/sound/soc/pxa/mmp-sspa.h @@ -0,0 +1,92 @@ +/* + * linux/sound/soc/pxa/mmp-sspa.h + * + * Copyright (C) 2011 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#ifndef _MMP_SSPA_H +#define _MMP_SSPA_H + +/* + * SSPA Registers + */ +#define SSPA_RXD (0x00) +#define SSPA_RXID (0x04) +#define SSPA_RXCTL (0x08) +#define SSPA_RXSP (0x0c) +#define SSPA_RXFIFO_UL (0x10) +#define SSPA_RXINT_MASK (0x14) +#define SSPA_RXC (0x18) +#define SSPA_RXFIFO_NOFS (0x1c) +#define SSPA_RXFIFO_SIZE (0x20) + +#define SSPA_TXD (0x80) +#define SSPA_TXID (0x84) +#define SSPA_TXCTL (0x88) +#define SSPA_TXSP (0x8c) +#define SSPA_TXFIFO_LL (0x90) +#define SSPA_TXINT_MASK (0x94) +#define SSPA_TXC (0x98) +#define SSPA_TXFIFO_NOFS (0x9c) +#define SSPA_TXFIFO_SIZE (0xa0) + +/* SSPA Control Register */ +#define SSPA_CTL_XPH (1 << 31) /* Read Phase */ +#define SSPA_CTL_XFIG (1 << 15) /* Transmit Zeros when FIFO Empty */ +#define SSPA_CTL_JST (1 << 3) /* Audio Sample Justification */ +#define SSPA_CTL_XFRLEN2_MASK (7 << 24) +#define SSPA_CTL_XFRLEN2(x) ((x) << 24) /* Transmit Frame Length in Phase 2 */ +#define SSPA_CTL_XWDLEN2_MASK (7 << 21) +#define SSPA_CTL_XWDLEN2(x) ((x) << 21) /* Transmit Word Length in Phase 2 */ +#define SSPA_CTL_XDATDLY(x) ((x) << 19) /* Tansmit Data Delay */ +#define SSPA_CTL_XSSZ2_MASK (7 << 16) +#define SSPA_CTL_XSSZ2(x) ((x) << 16) /* Transmit Sample Audio Size */ +#define SSPA_CTL_XFRLEN1_MASK (7 << 8) +#define SSPA_CTL_XFRLEN1(x) ((x) << 8) /* Transmit Frame Length in Phase 1 */ +#define SSPA_CTL_XWDLEN1_MASK (7 << 5) +#define SSPA_CTL_XWDLEN1(x) ((x) << 5) /* Transmit Word Length in Phase 1 */ +#define SSPA_CTL_XSSZ1_MASK (7 << 0) +#define SSPA_CTL_XSSZ1(x) ((x) << 0) /* XSSZ1 */ + +#define SSPA_CTL_8_BITS (0x0) /* Sample Size */ +#define SSPA_CTL_12_BITS (0x1) +#define SSPA_CTL_16_BITS (0x2) +#define SSPA_CTL_20_BITS (0x3) +#define SSPA_CTL_24_BITS (0x4) +#define SSPA_CTL_32_BITS (0x5) + +/* SSPA Serial Port Register */ +#define SSPA_SP_WEN (1 << 31) /* Write Configuration Enable */ +#define SSPA_SP_MSL (1 << 18) /* Master Slave Configuration */ +#define SSPA_SP_CLKP (1 << 17) /* CLKP Polarity Clock Edge Select */ +#define SSPA_SP_FSP (1 << 16) /* FSP Polarity Clock Edge Select */ +#define SSPA_SP_FFLUSH (1 << 2) /* FIFO Flush */ +#define SSPA_SP_S_RST (1 << 1) /* Active High Reset Signal */ +#define SSPA_SP_S_EN (1 << 0) /* Serial Clock Domain Enable */ +#define SSPA_SP_FWID(x) ((x) << 20) /* Frame-Sync Width */ +#define SSPA_TXSP_FPER(x) ((x) << 4) /* Frame-Sync Active */ + +/* sspa clock sources */ +#define MMP_SSPA_CLK_PLL 0 +#define MMP_SSPA_CLK_VCXO 1 +#define MMP_SSPA_CLK_AUDIO 3 + +/* sspa pll id */ +#define MMP_SYSCLK 0 +#define MMP_SSPA_CLK 1 + +#endif /* _MMP_SSPA_H */ diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index fd04ce139031..4da5fc55c7ee 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -33,7 +33,6 @@ #include <mach/hardware.h> #include <mach/dma.h> -#include <mach/audio.h> #include "../../arm/pxa2xx-pcm.h" #include "pxa-ssp.h" @@ -85,14 +84,12 @@ struct pxa2xx_pcm_dma_data { char name[20]; }; -static struct pxa2xx_pcm_dma_params * -pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) +static void pxa_ssp_set_dma_params(struct ssp_device *ssp, int width4, + int out, struct pxa2xx_pcm_dma_params *dma_data) { struct pxa2xx_pcm_dma_data *dma; - dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL); - if (dma == NULL) - return NULL; + dma = container_of(dma_data, struct pxa2xx_pcm_dma_data, params); snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id, width4 ? "32-bit" : "16-bit", out ? "out" : "in"); @@ -103,8 +100,6 @@ pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) (DCMD_INCTRGADDR | DCMD_FLOWSRC)) | (width4 ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16; dma->params.dev_addr = ssp->phys_base + SSDR; - - return &dma->params; } static int pxa_ssp_startup(struct snd_pcm_substream *substream, @@ -112,6 +107,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, { struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); struct ssp_device *ssp = priv->ssp; + struct pxa2xx_pcm_dma_data *dma; int ret = 0; if (!cpu_dai->active) { @@ -119,8 +115,10 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, pxa_ssp_disable(ssp); } - kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); - snd_soc_dai_set_dma_data(cpu_dai, substream, NULL); + dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL); + if (!dma) + return -ENOMEM; + snd_soc_dai_set_dma_data(cpu_dai, substream, &dma->params); return ret; } @@ -195,7 +193,7 @@ static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) { u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { + if (ssp->type == PXA25x_SSP) { sscr0 &= ~0x0000ff00; sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ } else { @@ -213,7 +211,7 @@ static u32 pxa_ssp_get_scr(struct ssp_device *ssp) u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); u32 div; - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) + if (ssp->type == PXA25x_SSP) div = ((sscr0 >> 8) & 0xff) * 2 + 2; else div = ((sscr0 >> 8) & 0xfff) + 1; @@ -243,7 +241,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, break; case PXA_SSP_CLK_PLL: /* Internal PLL is fixed */ - if (cpu_is_pxa25x()) + if (ssp->type == PXA25x_SSP) priv->sysclk = 1843200; else priv->sysclk = 13000000; @@ -267,11 +265,11 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, /* The SSP clock must be disabled when changing SSP clock mode * on PXA2xx. On PXA3xx it must be enabled when doing so. */ - if (!cpu_is_pxa3xx()) + if (ssp->type != PXA3xx_SSP) clk_disable(ssp->clk); val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; pxa_ssp_write_reg(ssp, SSCR0, val); - if (!cpu_is_pxa3xx()) + if (ssp->type != PXA3xx_SSP) clk_enable(ssp->clk); return 0; @@ -295,24 +293,20 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, case PXA_SSP_AUDIO_DIV_SCDB: val = pxa_ssp_read_reg(ssp, SSACD); val &= ~SSACD_SCDB; -#if defined(CONFIG_PXA3xx) - if (cpu_is_pxa3xx()) + if (ssp->type == PXA3xx_SSP) val &= ~SSACD_SCDX8; -#endif switch (div) { case PXA_SSP_CLK_SCDB_1: val |= SSACD_SCDB; break; case PXA_SSP_CLK_SCDB_4: break; -#if defined(CONFIG_PXA3xx) case PXA_SSP_CLK_SCDB_8: - if (cpu_is_pxa3xx()) + if (ssp->type == PXA3xx_SSP) val |= SSACD_SCDX8; else return -EINVAL; break; -#endif default: return -EINVAL; } @@ -338,10 +332,8 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, struct ssp_device *ssp = priv->ssp; u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; -#if defined(CONFIG_PXA3xx) - if (cpu_is_pxa3xx()) + if (ssp->type == PXA3xx_SSP) pxa_ssp_write_reg(ssp, SSACDD, 0); -#endif switch (freq_out) { case 5622000: @@ -366,11 +358,10 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, break; default: -#ifdef CONFIG_PXA3xx /* PXA3xx has a clock ditherer which can be used to generate * a wider range of frequencies - calculate a value for it. */ - if (cpu_is_pxa3xx()) { + if (ssp->type == PXA3xx_SSP) { u32 val; u64 tmp = 19968; tmp *= 1000000; @@ -387,7 +378,6 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, val, freq_out); break; } -#endif return -EINVAL; } @@ -573,18 +563,13 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); - /* generate correct DMA params */ - kfree(dma_data); - /* Network mode with one active slot (ttsa == 1) can be used * to force 16-bit frame width on the wire (for S16_LE), even * with two channels. Use 16-bit DMA transfers for this case. */ - dma_data = pxa_ssp_get_dma_params(ssp, - ((chn == 2) && (ttsa != 1)) || (width == 32), - substream->stream == SNDRV_PCM_STREAM_PLAYBACK); - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + pxa_ssp_set_dma_params(ssp, + ((chn == 2) && (ttsa != 1)) || (width == 32), + substream->stream == SNDRV_PCM_STREAM_PLAYBACK, dma_data); /* we can only change the settings if the port is not in use */ if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) @@ -596,10 +581,8 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: -#ifdef CONFIG_PXA3xx - if (cpu_is_pxa3xx()) + if (ssp->type == PXA3xx_SSP) sscr0 |= SSCR0_FPCKE; -#endif sscr0 |= SSCR0_DataSize(16); break; case SNDRV_PCM_FORMAT_S24_LE: @@ -624,9 +607,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, * trying and failing a lot; some of the registers * needed for that mode are only available on PXA3xx. */ - -#ifdef CONFIG_PXA3xx - if (!cpu_is_pxa3xx()) + if (ssp->type != PXA3xx_SSP) return -EINVAL; sspsp |= SSPSP_SFRMWDTH(width * 2); @@ -634,9 +615,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sspsp |= SSPSP_EDMYSTOP(3); sspsp |= SSPSP_DMYSTOP(3); sspsp |= SSPSP_DMYSTRT(1); -#else - return -EINVAL; -#endif } else { /* The frame width is the width the LRCLK is * asserted for; the delay is expressed in diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index d08583790d23..3075a426124c 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -166,7 +166,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, struct pxa2xx_pcm_dma_params *dma_data; BUG_ON(IS_ERR(clk_i2s)); - clk_enable(clk_i2s); + clk_prepare_enable(clk_i2s); clk_ena = 1; pxa_i2s_wait(); @@ -259,7 +259,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, SACR0 &= ~SACR0_ENB; pxa_i2s_wait(); if (clk_ena) { - clk_disable(clk_i2s); + clk_disable_unprepare(clk_i2s); clk_ena = 0; } } diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c new file mode 100644 index 000000000000..935491a8a770 --- /dev/null +++ b/sound/soc/pxa/ttc-dkb.c @@ -0,0 +1,173 @@ +/* + * linux/sound/soc/pxa/ttc_dkb.c + * + * Copyright (C) 2012 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <sound/pcm_params.h> +#include "../codecs/88pm860x-codec.h" + +static struct snd_soc_jack hs_jack, mic_jack; + +static struct snd_soc_jack_pin hs_jack_pins[] = { + { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, +}; + +static struct snd_soc_jack_pin mic_jack_pins[] = { + { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, +}; + +/* ttc machine dapm widgets */ +static const struct snd_soc_dapm_widget ttc_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_LINE("Lineout Out 1", NULL), + SND_SOC_DAPM_LINE("Lineout Out 2", NULL), + SND_SOC_DAPM_SPK("Ext Speaker", NULL), + SND_SOC_DAPM_MIC("Ext Mic 1", NULL), + SND_SOC_DAPM_MIC("Headset Mic 2", NULL), + SND_SOC_DAPM_MIC("Ext Mic 3", NULL), +}; + +/* ttc machine audio map */ +static const struct snd_soc_dapm_route ttc_audio_map[] = { + {"Headset Stereophone", NULL, "HS1"}, + {"Headset Stereophone", NULL, "HS2"}, + + {"Ext Speaker", NULL, "LSP"}, + {"Ext Speaker", NULL, "LSN"}, + + {"Lineout Out 1", NULL, "LINEOUT1"}, + {"Lineout Out 2", NULL, "LINEOUT2"}, + + {"MIC1P", NULL, "Mic1 Bias"}, + {"MIC1N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Ext Mic 1"}, + + {"MIC2P", NULL, "Mic1 Bias"}, + {"MIC2N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Headset Mic 2"}, + + {"MIC3P", NULL, "Mic3 Bias"}, + {"MIC3N", NULL, "Mic3 Bias"}, + {"Mic3 Bias", NULL, "Ext Mic 3"}, +}; + +static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + /* connected pins */ + snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); + snd_soc_dapm_enable_pin(dapm, "Ext Mic 1"); + snd_soc_dapm_enable_pin(dapm, "Ext Mic 3"); + snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); + snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); + + /* Headset jack detection */ + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE + | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, + &hs_jack); + snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE, + &mic_jack); + snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), + mic_jack_pins); + + /* headphone, microphone detection & headset short detection */ + pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE, + SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2); + pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE); + + return 0; +} + +/* ttc/td-dkb digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link ttc_pm860x_hifi_dai[] = { +{ + .name = "88pm860x i2s", + .stream_name = "audio playback", + .codec_name = "88pm860x-codec", + .platform_name = "mmp-pcm-audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "88pm860x-i2s", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + .init = ttc_pm860x_init, +}, +}; + +/* ttc/td audio machine driver */ +static struct snd_soc_card ttc_dkb_card = { + .name = "ttc-dkb-hifi", + .dai_link = ttc_pm860x_hifi_dai, + .num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai), + + .dapm_widgets = ttc_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ttc_dapm_widgets), + .dapm_routes = ttc_audio_map, + .num_dapm_routes = ARRAY_SIZE(ttc_audio_map), +}; + +static int __devinit ttc_dkb_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &ttc_dkb_card; + int ret; + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + + return ret; +} + +static int __devexit ttc_dkb_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver ttc_dkb_driver = { + .driver = { + .name = "ttc-dkb-audio", + .owner = THIS_MODULE, + }, + .probe = ttc_dkb_probe, + .remove = __devexit_p(ttc_dkb_remove), +}; + +module_platform_driver(ttc_dkb_driver); + +/* Module information */ +MODULE_AUTHOR("Qiao Zhou, <zhouqiao@marvell.com>"); +MODULE_DESCRIPTION("ALSA SoC TTC DKB"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:ttc-dkb-audio"); diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index ddc6cde14e2a..f3ebc38c10fe 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -74,7 +74,7 @@ static void dma_enqueue(struct snd_pcm_substream *substream) struct runtime_data *prtd = substream->runtime->private_data; dma_addr_t pos = prtd->dma_pos; unsigned int limit; - struct samsung_dma_prep_info dma_info; + struct samsung_dma_prep dma_info; pr_debug("Entered %s\n", __func__); @@ -146,7 +146,8 @@ static int dma_hw_params(struct snd_pcm_substream *substream, unsigned long totbytes = params_buffer_bytes(params); struct s3c_dma_params *dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - struct samsung_dma_info dma_info; + struct samsung_dma_req req; + struct samsung_dma_config config; pr_debug("Entered %s\n", __func__); @@ -166,16 +167,17 @@ static int dma_hw_params(struct snd_pcm_substream *substream, prtd->params->ops = samsung_dma_get_ops(); - dma_info.cap = (samsung_dma_has_circular() ? + req.cap = (samsung_dma_has_circular() ? DMA_CYCLIC : DMA_SLAVE); - dma_info.client = prtd->params->client; - dma_info.direction = + req.client = prtd->params->client; + config.direction = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); - dma_info.width = prtd->params->dma_size; - dma_info.fifo = prtd->params->dma_addr; + config.width = prtd->params->dma_size; + config.fifo = prtd->params->dma_addr; prtd->params->ch = prtd->params->ops->request( - prtd->params->channel, &dma_info); + prtd->params->channel, &req); + prtd->params->ops->config(prtd->params->ch, &config); } snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index e7416851bf7d..ee52c8a00779 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -23,10 +23,10 @@ static int littlemill_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; int ret; - if (dapm->dev != codec_dai->dev) + if (dapm->dev != aif1_dai->dev) return 0; switch (level) { @@ -36,7 +36,7 @@ static int littlemill_set_bias_level(struct snd_soc_card *card, * then do so now, otherwise these are noops. */ if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, + ret = snd_soc_dai_set_pll(aif1_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK2, 32768, sample_rate * 512); if (ret < 0) { @@ -44,7 +44,7 @@ static int littlemill_set_bias_level(struct snd_soc_card *card, return ret; } - ret = snd_soc_dai_set_sysclk(codec_dai, + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_FLL1, sample_rate * 512, SND_SOC_CLOCK_IN); @@ -66,25 +66,25 @@ static int littlemill_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; int ret; - if (dapm->dev != codec_dai->dev) + if (dapm->dev != aif1_dai->dev) return 0; switch (level) { case SND_SOC_BIAS_STANDBY: - ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK2, + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2, 32768, SND_SOC_CLOCK_IN); if (ret < 0) { - pr_err("Failed to switch away from FLL: %d\n", ret); + pr_err("Failed to switch away from FLL1: %d\n", ret); return ret; } - ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, + ret = snd_soc_dai_set_pll(aif1_dai, WM8994_FLL1, 0, 0, 0); if (ret < 0) { - pr_err("Failed to stop FLL: %d\n", ret); + pr_err("Failed to stop FLL1: %d\n", ret); return ret; } break; @@ -131,6 +131,14 @@ static struct snd_soc_ops littlemill_ops = { .hw_params = littlemill_hw_params, }; +static const struct snd_soc_pcm_stream baseband_params = { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 2, + .channels_max = 2, +}; + static struct snd_soc_dai_link littlemill_dai[] = { { .name = "CPU", @@ -143,6 +151,69 @@ static struct snd_soc_dai_link littlemill_dai[] = { | SND_SOC_DAIFMT_CBM_CFM, .ops = &littlemill_ops, }, + { + .name = "Baseband", + .stream_name = "Baseband", + .cpu_dai_name = "wm8994-aif2", + .codec_dai_name = "wm1250-ev1", + .codec_name = "wm1250-ev1.1-0027", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + .params = &baseband_params, + }, +}; + +static int bbclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_card *card = w->dapm->card; + struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai; + int ret; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = snd_soc_dai_set_pll(aif2_dai, WM8994_FLL2, + WM8994_FLL_SRC_BCLK, 64 * 8000, + 8000 * 256); + if (ret < 0) { + pr_err("Failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(aif2_dai, WM8994_SYSCLK_FLL2, + 8000 * 256, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK: %d\n", ret); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMD: + ret = snd_soc_dai_set_sysclk(aif2_dai, WM8994_SYSCLK_MCLK2, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to switch away from FLL2: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(aif2_dai, WM8994_FLL2, + 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL2: %d\n", ret); + return ret; + } + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_kcontrol_new controls[] = { + SOC_DAPM_PIN_SWITCH("WM1250 Input"), + SOC_DAPM_PIN_SWITCH("WM1250 Output"), }; static struct snd_soc_dapm_widget widgets[] = { @@ -150,6 +221,10 @@ static struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_MIC("AMIC", NULL), SND_SOC_DAPM_MIC("DMIC", NULL), + + SND_SOC_DAPM_SUPPLY_S("Baseband Clock", -1, SND_SOC_NOPM, 0, 0, + bbclk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), }; static struct snd_soc_dapm_route audio_paths[] = { @@ -162,6 +237,8 @@ static struct snd_soc_dapm_route audio_paths[] = { { "DMIC", NULL, "MICBIAS2" }, /* Default for DMICBIAS jumper */ { "DMIC1DAT", NULL, "DMIC" }, { "DMIC2DAT", NULL, "DMIC" }, + + { "AIF2CLK", NULL, "Baseband Clock" }, }; static struct snd_soc_jack littlemill_headset; @@ -169,10 +246,16 @@ static struct snd_soc_jack littlemill_headset; static int littlemill_late_probe(struct snd_soc_card *card) { struct snd_soc_codec *codec = card->rtd[0].codec; - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai; int ret; - ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK2, + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(aif2_dai, WM8994_SYSCLK_MCLK2, 32768, SND_SOC_CLOCK_IN); if (ret < 0) return ret; @@ -204,6 +287,8 @@ static struct snd_soc_card littlemill = { .set_bias_level = littlemill_set_bias_level, .set_bias_level_post = littlemill_set_bias_level_post, + .controls = controls, + .num_controls = ARRAY_SIZE(controls), .dapm_widgets = widgets, .num_dapm_widgets = ARRAY_SIZE(widgets), .dapm_routes = audio_paths, diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index 4adff934f771..6abf341c4a2a 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -21,33 +21,6 @@ #define MCLK1_RATE (44100 * 512) #define CLKOUT_RATE (44100 * 256) -static int lowland_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops lowland_ops = { - .hw_params = lowland_hw_params, -}; - static struct snd_soc_jack lowland_headset; /* Headset jack detection DAPM pins */ @@ -101,6 +74,25 @@ static int lowland_wm5100_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int lowland_wm9081_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + + snd_soc_dapm_nc_pin(&codec->dapm, "LINEOUT"); + + /* At any time the WM9081 is active it will have this clock */ + return snd_soc_codec_set_sysclk(codec, WM9081_SYSCLK_MCLK, 0, + CLKOUT_RATE, 0); +} + +static const struct snd_soc_pcm_stream sub_params = { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = 44100, + .rate_max = 44100, + .channels_min = 2, + .channels_max = 2, +}; + static struct snd_soc_dai_link lowland_dai[] = { { .name = "CPU", @@ -109,7 +101,8 @@ static struct snd_soc_dai_link lowland_dai[] = { .codec_dai_name = "wm5100-aif1", .platform_name = "samsung-audio", .codec_name = "wm5100.1-001a", - .ops = &lowland_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .init = lowland_wm5100_init, }, { @@ -118,24 +111,20 @@ static struct snd_soc_dai_link lowland_dai[] = { .cpu_dai_name = "wm5100-aif2", .codec_dai_name = "wm1250-ev1", .codec_name = "wm1250-ev1.1-0027", - .ops = &lowland_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ignore_suspend = 1, }, -}; - -static int lowland_wm9081_init(struct snd_soc_dapm_context *dapm) -{ - snd_soc_dapm_nc_pin(dapm, "LINEOUT"); - - /* At any time the WM9081 is active it will have this clock */ - return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, - CLKOUT_RATE, 0); -} - -static struct snd_soc_aux_dev lowland_aux_dev[] = { { - .name = "wm9081", + .name = "Sub Speaker", + .stream_name = "Sub Speaker", + .cpu_dai_name = "wm5100-aif3", + .codec_dai_name = "wm9081-hifi", .codec_name = "wm9081.1-006c", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + .params = &sub_params, .init = lowland_wm9081_init, }, }; @@ -180,8 +169,6 @@ static struct snd_soc_card lowland = { .owner = THIS_MODULE, .dai_link = lowland_dai, .num_links = ARRAY_SIZE(lowland_dai), - .aux_dev = lowland_aux_dev, - .num_aux_devs = ARRAY_SIZE(lowland_aux_dev), .codec_conf = lowland_codec_conf, .num_configs = ARRAY_SIZE(lowland_codec_conf), diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 79fbeea99d46..ac7701b3c5dc 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -25,7 +25,6 @@ #include <sound/soc.h> #include <sound/pcm_params.h> -#include <mach/regs-gpio.h> #include <mach/dma.h> #include "dma.h" @@ -83,12 +82,9 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk; - /* Configure the I2S pins in correct mode */ - s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); - s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); - s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); return 0; } diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index c4aa4d412fbf..0aae3a3883dc 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -23,7 +23,6 @@ #include <sound/soc.h> #include <sound/pcm_params.h> -#include <mach/regs-gpio.h> #include <mach/dma.h> #include <plat/regs-iis.h> @@ -391,12 +390,9 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai) } clk_enable(s3c24xx_i2s.iis_clk); - /* Configure the I2S pins in correct mode */ - s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); - s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); - s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); writel(S3C2410_IISCON_IISEN, s3c24xx_i2s.regs + S3C2410_IISCON); diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 8eb309f23d18..48dd4dd9ee08 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -149,31 +149,41 @@ static struct snd_soc_card smdk = { .num_links = ARRAY_SIZE(smdk_dai), }; -static struct platform_device *smdk_snd_device; -static int __init smdk_audio_init(void) +static int __devinit smdk_audio_probe(struct platform_device *pdev) { int ret; + struct snd_soc_card *card = &smdk; - smdk_snd_device = platform_device_alloc("soc-audio", -1); - if (!smdk_snd_device) - return -ENOMEM; + card->dev = &pdev->dev; + ret = snd_soc_register_card(card); - platform_set_drvdata(smdk_snd_device, &smdk); - - ret = platform_device_add(smdk_snd_device); if (ret) - platform_device_put(smdk_snd_device); + dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret); return ret; } -module_init(smdk_audio_init); -static void __exit smdk_audio_exit(void) +static int __devexit smdk_audio_remove(struct platform_device *pdev) { - platform_device_unregister(smdk_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; } -module_exit(smdk_audio_exit); + +static struct platform_driver smdk_audio_driver = { + .driver = { + .name = "smdk-audio", + .owner = THIS_MODULE, + }, + .probe = smdk_audio_probe, + .remove = __devexit_p(smdk_audio_remove), +}; + +module_platform_driver(smdk_audio_driver); MODULE_DESCRIPTION("ALSA SoC SMDK WM8994"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:smdk-audio"); diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index f9ab7707a3e4..a4a9fc7e8c76 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -92,33 +92,6 @@ static int speyside_set_bias_level_post(struct snd_soc_card *card, return 0; } -static int speyside_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops speyside_ops = { - .hw_params = speyside_hw_params, -}; - static struct snd_soc_jack speyside_headset; /* Headset jack detection DAPM pins */ @@ -208,7 +181,8 @@ static struct snd_soc_dai_link speyside_dai[] = { .platform_name = "samsung-audio", .codec_name = "wm8996.1-001a", .init = speyside_wm8996_init, - .ops = &speyside_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, }, { .name = "Baseband", @@ -216,7 +190,8 @@ static struct snd_soc_dai_link speyside_dai[] = { .cpu_dai_name = "wm8996-aif2", .codec_dai_name = "wm1250-ev1", .codec_name = "wm1250-ev1.1-0027", - .ops = &speyside_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, .ignore_suspend = 1, }, }; diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index d8e06a607a22..6bcb1164d599 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -22,6 +22,7 @@ config SND_SOC_SH4_SSI config SND_SOC_SH4_FSI tristate "SH4 FSI support" + select SND_SIMPLE_CARD help This option enables FSI sound support @@ -46,29 +47,6 @@ config SND_SH7760_AC97 This option enables generic sound support for the first AC97 unit of the SH7760. -config SND_FSI_AK4642 - tristate "FSI-AK4642 sound support" - depends on SND_SOC_SH4_FSI && I2C - select SND_SOC_AK4642 - help - This option enables generic sound support for the - FSI - AK4642 unit - -config SND_FSI_DA7210 - tristate "FSI-DA7210 sound support" - depends on SND_SOC_SH4_FSI && I2C - select SND_SOC_DA7210 - help - This option enables generic sound support for the - FSI - DA7210 unit - -config SND_FSI_HDMI - tristate "FSI-HDMI sound support" - depends on SND_SOC_SH4_FSI && FB_SH_MOBILE_HDMI - help - This option enables generic sound support for the - FSI - HDMI unit - config SND_SIU_MIGOR tristate "SIU sound support on Migo-R" depends on SH_MIGOR diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index 94476d4c0fd5..849b387d17d9 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -14,13 +14,7 @@ obj-$(CONFIG_SND_SOC_SH4_SIU) += snd-soc-siu.o ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o -snd-soc-fsi-ak4642-objs := fsi-ak4642.o -snd-soc-fsi-da7210-objs := fsi-da7210.o -snd-soc-fsi-hdmi-objs := fsi-hdmi.o snd-soc-migor-objs := migor.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o -obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o -obj-$(CONFIG_SND_FSI_DA7210) += snd-soc-fsi-da7210.o -obj-$(CONFIG_SND_FSI_HDMI) += snd-soc-fsi-hdmi.o obj-$(CONFIG_SND_SIU_MIGOR) += snd-soc-migor.o diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c deleted file mode 100644 index 97f540aabbdd..000000000000 --- a/sound/soc/sh/fsi-ak4642.c +++ /dev/null @@ -1,108 +0,0 @@ -/* - * FSI-AK464x sound support for ms7724se - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Kuninori Morimoto <morimoto.kuninori@renesas.com> - * - * This file is subject to the terms and conditions of the GNU General Public - * License. See the file "COPYING" in the main directory of this archive - * for more details. - */ - -#include <linux/platform_device.h> -#include <linux/module.h> -#include <sound/sh_fsi.h> - -struct fsi_ak4642_data { - const char *name; - const char *card; - const char *cpu_dai; - const char *codec; - const char *platform; - int id; -}; - -static int fsi_ak4642_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *codec = rtd->codec_dai; - struct snd_soc_dai *cpu = rtd->cpu_dai; - int ret; - - ret = snd_soc_dai_set_fmt(codec, SND_SOC_DAIFMT_LEFT_J | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(codec, 0, 11289600, 0); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_LEFT_J | - SND_SOC_DAIFMT_CBS_CFS); - - return ret; -} - -static struct snd_soc_dai_link fsi_dai_link = { - .codec_dai_name = "ak4642-hifi", - .init = fsi_ak4642_dai_init, -}; - -static struct snd_soc_card fsi_soc_card = { - .owner = THIS_MODULE, - .dai_link = &fsi_dai_link, - .num_links = 1, -}; - -static struct platform_device *fsi_snd_device; - -static int fsi_ak4642_probe(struct platform_device *pdev) -{ - int ret = -ENOMEM; - struct fsi_ak4642_info *pinfo = pdev->dev.platform_data; - - if (!pinfo) { - dev_err(&pdev->dev, "no info for fsi ak4642\n"); - goto out; - } - - fsi_snd_device = platform_device_alloc("soc-audio", pinfo->id); - if (!fsi_snd_device) - goto out; - - fsi_dai_link.name = pinfo->name; - fsi_dai_link.stream_name = pinfo->name; - fsi_dai_link.cpu_dai_name = pinfo->cpu_dai; - fsi_dai_link.platform_name = pinfo->platform; - fsi_dai_link.codec_name = pinfo->codec; - fsi_soc_card.name = pinfo->card; - - platform_set_drvdata(fsi_snd_device, &fsi_soc_card); - ret = platform_device_add(fsi_snd_device); - - if (ret) - platform_device_put(fsi_snd_device); - -out: - return ret; -} - -static int fsi_ak4642_remove(struct platform_device *pdev) -{ - platform_device_unregister(fsi_snd_device); - return 0; -} - -static struct platform_driver fsi_ak4642 = { - .driver = { - .name = "fsi-ak4642-audio", - }, - .probe = fsi_ak4642_probe, - .remove = fsi_ak4642_remove, -}; - -module_platform_driver(fsi_ak4642); - -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("Generic SH4 FSI-AK4642 sound card"); -MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>"); diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c deleted file mode 100644 index 1dd3354c7411..000000000000 --- a/sound/soc/sh/fsi-da7210.c +++ /dev/null @@ -1,81 +0,0 @@ -/* - * fsi-da7210.c - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Kuninori Morimoto <morimoto.kuninori@renesas.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/platform_device.h> -#include <linux/module.h> -#include <sound/sh_fsi.h> - -static int fsi_da7210_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *codec = rtd->codec_dai; - struct snd_soc_dai *cpu = rtd->cpu_dai; - int ret; - - ret = snd_soc_dai_set_fmt(codec, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_CBS_CFS); - - return ret; -} - -static struct snd_soc_dai_link fsi_da7210_dai = { - .name = "DA7210", - .stream_name = "DA7210", - .cpu_dai_name = "fsib-dai", /* FSI B */ - .codec_dai_name = "da7210-hifi", - .platform_name = "sh_fsi.0", - .codec_name = "da7210-codec.0-001a", - .init = fsi_da7210_init, -}; - -static struct snd_soc_card fsi_soc_card = { - .name = "FSI-DA7210", - .owner = THIS_MODULE, - .dai_link = &fsi_da7210_dai, - .num_links = 1, -}; - -static struct platform_device *fsi_da7210_snd_device; - -static int __init fsi_da7210_sound_init(void) -{ - int ret; - - fsi_da7210_snd_device = platform_device_alloc("soc-audio", FSI_PORT_B); - if (!fsi_da7210_snd_device) - return -ENOMEM; - - platform_set_drvdata(fsi_da7210_snd_device, &fsi_soc_card); - ret = platform_device_add(fsi_da7210_snd_device); - if (ret) - platform_device_put(fsi_da7210_snd_device); - - return ret; -} - -static void __exit fsi_da7210_sound_exit(void) -{ - platform_device_unregister(fsi_da7210_snd_device); -} - -module_init(fsi_da7210_sound_init); -module_exit(fsi_da7210_sound_exit); - -/* Module information */ -MODULE_DESCRIPTION("ALSA SoC FSI DA2710"); -MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/sh/fsi-hdmi.c b/sound/soc/sh/fsi-hdmi.c deleted file mode 100644 index 6e41908323e8..000000000000 --- a/sound/soc/sh/fsi-hdmi.c +++ /dev/null @@ -1,118 +0,0 @@ -/* - * FSI - HDMI sound support - * - * Copyright (C) 2010 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> - * - * This file is subject to the terms and conditions of the GNU General Public - * License. See the file "COPYING" in the main directory of this archive - * for more details. - */ - -#include <linux/platform_device.h> -#include <linux/module.h> -#include <sound/sh_fsi.h> - -struct fsi_hdmi_data { - const char *cpu_dai; - const char *card; - int id; -}; - -static int fsi_hdmi_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *cpu = rtd->cpu_dai; - int ret; - - ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_CBM_CFM); - - return ret; -} - -static struct snd_soc_dai_link fsi_dai_link = { - .name = "HDMI", - .stream_name = "HDMI", - .codec_dai_name = "sh_mobile_hdmi-hifi", - .platform_name = "sh_fsi2", - .codec_name = "sh-mobile-hdmi", - .init = fsi_hdmi_dai_init, -}; - -static struct snd_soc_card fsi_soc_card = { - .owner = THIS_MODULE, - .dai_link = &fsi_dai_link, - .num_links = 1, -}; - -static struct platform_device *fsi_snd_device; - -static int fsi_hdmi_probe(struct platform_device *pdev) -{ - int ret = -ENOMEM; - const struct platform_device_id *id_entry; - struct fsi_hdmi_data *pdata; - - id_entry = pdev->id_entry; - if (!id_entry) { - dev_err(&pdev->dev, "unknown fsi hdmi\n"); - return -ENODEV; - } - - pdata = (struct fsi_hdmi_data *)id_entry->driver_data; - - fsi_snd_device = platform_device_alloc("soc-audio", pdata->id); - if (!fsi_snd_device) - goto out; - - fsi_dai_link.cpu_dai_name = pdata->cpu_dai; - fsi_soc_card.name = pdata->card; - - platform_set_drvdata(fsi_snd_device, &fsi_soc_card); - ret = platform_device_add(fsi_snd_device); - - if (ret) - platform_device_put(fsi_snd_device); - -out: - return ret; -} - -static int fsi_hdmi_remove(struct platform_device *pdev) -{ - platform_device_unregister(fsi_snd_device); - return 0; -} - -static struct fsi_hdmi_data fsi2_a_hdmi = { - .cpu_dai = "fsia-dai", - .card = "FSI2A-HDMI", - .id = FSI_PORT_A, -}; - -static struct fsi_hdmi_data fsi2_b_hdmi = { - .cpu_dai = "fsib-dai", - .card = "FSI2B-HDMI", - .id = FSI_PORT_B, -}; - -static struct platform_device_id fsi_id_table[] = { - /* FSI 2 */ - { "sh_fsi2_a_hdmi", (kernel_ulong_t)&fsi2_a_hdmi }, - { "sh_fsi2_b_hdmi", (kernel_ulong_t)&fsi2_b_hdmi }, - {}, -}; - -static struct platform_driver fsi_hdmi = { - .driver = { - .name = "fsi-hdmi-audio", - }, - .probe = fsi_hdmi_probe, - .remove = fsi_hdmi_remove, - .id_table = fsi_id_table, -}; - -module_platform_driver(fsi_hdmi); - -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("Generic SH4 FSI-HDMI sound card"); -MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>"); diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 74ed2dffbffd..0540408a9fa9 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -132,6 +132,25 @@ typedef int (*set_rate_func)(struct device *dev, int rate, int enable); /* + * bus options + * + * 0x000000BA + * + * A : sample widtht 16bit setting + * B : sample widtht 24bit setting + */ + +#define SHIFT_16DATA 0 +#define SHIFT_24DATA 4 + +#define PACKAGE_24BITBUS_BACK 0 +#define PACKAGE_24BITBUS_FRONT 1 +#define PACKAGE_16BITBUS_STREAM 2 + +#define BUSOP_SET(s, a) ((a) << SHIFT_ ## s ## DATA) +#define BUSOP_GET(s, a) (((a) >> SHIFT_ ## s ## DATA) & 0xF) + +/* * FSI driver use below type name for variable * * xxx_num : number of data @@ -189,6 +208,11 @@ struct fsi_stream { int oerr_num; /* + * bus options + */ + u32 bus_option; + + /* * thse are initialized by fsi_handler_init() */ struct fsi_stream_handler *handler; @@ -211,8 +235,7 @@ struct fsi_priv { struct fsi_stream playback; struct fsi_stream capture; - u32 do_fmt; - u32 di_fmt; + u32 fmt; int chan_num:16; int clk_master:1; @@ -224,7 +247,7 @@ struct fsi_priv { struct fsi_stream_handler { int (*init)(struct fsi_priv *fsi, struct fsi_stream *io); int (*quit)(struct fsi_priv *fsi, struct fsi_stream *io); - int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io); + int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev); int (*transfer)(struct fsi_priv *fsi, struct fsi_stream *io); int (*remove)(struct fsi_priv *fsi, struct fsi_stream *io); void (*start_stop)(struct fsi_priv *fsi, struct fsi_stream *io, @@ -321,6 +344,10 @@ static void _fsi_master_mask_set(struct fsi_master *master, /* * basic function */ +static int fsi_version(struct fsi_master *master) +{ + return master->core->ver; +} static struct fsi_master *fsi_get_master(struct fsi_priv *fsi) { @@ -495,6 +522,7 @@ static void fsi_stream_init(struct fsi_priv *fsi, io->period_samples = fsi_frame2sample(fsi, runtime->period_size); io->period_pos = 0; io->sample_width = samples_to_bytes(runtime, 1); + io->bus_option = 0; io->oerr_num = -1; /* ignore 1st err */ io->uerr_num = -1; /* ignore 1st err */ fsi_stream_handler_call(io, init, fsi, io); @@ -522,6 +550,7 @@ static void fsi_stream_quit(struct fsi_priv *fsi, struct fsi_stream *io) io->period_samples = 0; io->period_pos = 0; io->sample_width = 0; + io->bus_option = 0; io->oerr_num = 0; io->uerr_num = 0; spin_unlock_irqrestore(&master->lock, flags); @@ -542,16 +571,16 @@ static int fsi_stream_transfer(struct fsi_stream *io) #define fsi_stream_stop(fsi, io)\ fsi_stream_handler_call(io, start_stop, fsi, io, 0) -static int fsi_stream_probe(struct fsi_priv *fsi) +static int fsi_stream_probe(struct fsi_priv *fsi, struct device *dev) { struct fsi_stream *io; int ret1, ret2; io = &fsi->playback; - ret1 = fsi_stream_handler_call(io, probe, fsi, io); + ret1 = fsi_stream_handler_call(io, probe, fsi, io, dev); io = &fsi->capture; - ret2 = fsi_stream_handler_call(io, probe, fsi, io); + ret2 = fsi_stream_handler_call(io, probe, fsi, io, dev); if (ret1 < 0) return ret1; @@ -581,6 +610,53 @@ static int fsi_stream_remove(struct fsi_priv *fsi) } /* + * format/bus/dma setting + */ +static void fsi_format_bus_setup(struct fsi_priv *fsi, struct fsi_stream *io, + u32 bus, struct device *dev) +{ + struct fsi_master *master = fsi_get_master(fsi); + int is_play = fsi_stream_is_play(fsi, io); + u32 fmt = fsi->fmt; + + if (fsi_version(master) >= 2) { + u32 dma = 0; + + /* + * FSI2 needs DMA/Bus setting + */ + switch (bus) { + case PACKAGE_24BITBUS_FRONT: + fmt |= CR_BWS_24; + dma |= VDMD_FRONT; + dev_dbg(dev, "24bit bus / package in front\n"); + break; + case PACKAGE_16BITBUS_STREAM: + fmt |= CR_BWS_16; + dma |= VDMD_STREAM; + dev_dbg(dev, "16bit bus / stream mode\n"); + break; + case PACKAGE_24BITBUS_BACK: + default: + fmt |= CR_BWS_24; + dma |= VDMD_BACK; + dev_dbg(dev, "24bit bus / package in back\n"); + break; + } + + if (is_play) + fsi_reg_write(fsi, OUT_DMAC, dma); + else + fsi_reg_write(fsi, IN_DMAC, dma); + } + + if (is_play) + fsi_reg_write(fsi, DO_FMT, fmt); + else + fsi_reg_write(fsi, DI_FMT, fmt); +} + +/* * irq function */ @@ -629,11 +705,6 @@ static void fsi_spdif_clk_ctrl(struct fsi_priv *fsi, int enable) struct fsi_master *master = fsi_get_master(fsi); u32 mask, val; - if (master->core->ver < 2) { - pr_err("fsi: register access err (%s)\n", __func__); - return; - } - mask = BP | SE; val = enable ? mask : 0; @@ -648,9 +719,7 @@ static void fsi_spdif_clk_ctrl(struct fsi_priv *fsi, int enable) static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi, long rate, int enable) { - struct fsi_master *master = fsi_get_master(fsi); set_rate_func set_rate = fsi_get_info_set_rate(fsi); - int fsi_ver = master->core->ver; int ret; if (!set_rate) @@ -682,10 +751,7 @@ static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi, data |= (0x3 << 12); break; case SH_FSI_ACKMD_32: - if (fsi_ver < 2) - dev_err(dev, "unsupported ACKMD\n"); - else - data |= (0x4 << 12); + data |= (0x4 << 12); break; } @@ -708,10 +774,7 @@ static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi, data |= (0x4 << 8); break; case SH_FSI_BPFMD_16: - if (fsi_ver < 2) - dev_err(dev, "unsupported ACKMD\n"); - else - data |= (0x7 << 8); + data |= (0x7 << 8); break; } @@ -728,11 +791,26 @@ static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi, */ static void fsi_pio_push16(struct fsi_priv *fsi, u8 *_buf, int samples) { - u16 *buf = (u16 *)_buf; + u32 enable_stream = fsi_get_info_flags(fsi) & SH_FSI_ENABLE_STREAM_MODE; int i; - for (i = 0; i < samples; i++) - fsi_reg_write(fsi, DODT, ((u32)*(buf + i) << 8)); + if (enable_stream) { + /* + * stream mode + * see + * fsi_pio_push_init() + */ + u32 *buf = (u32 *)_buf; + + for (i = 0; i < samples / 2; i++) + fsi_reg_write(fsi, DODT, buf[i]); + } else { + /* normal mode */ + u16 *buf = (u16 *)_buf; + + for (i = 0; i < samples; i++) + fsi_reg_write(fsi, DODT, ((u32)*(buf + i) << 8)); + } } static void fsi_pio_pop16(struct fsi_priv *fsi, u8 *_buf, int samples) @@ -872,12 +950,44 @@ static void fsi_pio_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); } +static int fsi_pio_push_init(struct fsi_priv *fsi, struct fsi_stream *io) +{ + u32 enable_stream = fsi_get_info_flags(fsi) & SH_FSI_ENABLE_STREAM_MODE; + + /* + * we can use 16bit stream mode + * when "playback" and "16bit data" + * and platform allows "stream mode" + * see + * fsi_pio_push16() + */ + if (enable_stream) + io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) | + BUSOP_SET(16, PACKAGE_16BITBUS_STREAM); + else + io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) | + BUSOP_SET(16, PACKAGE_24BITBUS_BACK); + return 0; +} + +static int fsi_pio_pop_init(struct fsi_priv *fsi, struct fsi_stream *io) +{ + /* + * always 24bit bus, package back when "capture" + */ + io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) | + BUSOP_SET(16, PACKAGE_24BITBUS_BACK); + return 0; +} + static struct fsi_stream_handler fsi_pio_push_handler = { + .init = fsi_pio_push_init, .transfer = fsi_pio_push, .start_stop = fsi_pio_start_stop, }; static struct fsi_stream_handler fsi_pio_pop_handler = { + .init = fsi_pio_pop_init, .transfer = fsi_pio_pop, .start_stop = fsi_pio_start_stop, }; @@ -919,6 +1029,13 @@ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io) enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ? DMA_TO_DEVICE : DMA_FROM_DEVICE; + /* + * 24bit data : 24bit bus / package in back + * 16bit data : 16bit bus / stream mode + */ + io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) | + BUSOP_SET(16, PACKAGE_16BITBUS_STREAM); + io->dma = dma_map_single(dai->dev, runtime->dma_area, snd_pcm_lib_buffer_bytes(io->substream), dir); return 0; @@ -935,6 +1052,13 @@ static int fsi_dma_quit(struct fsi_priv *fsi, struct fsi_stream *io) return 0; } +static dma_addr_t fsi_dma_get_area(struct fsi_stream *io) +{ + struct snd_pcm_runtime *runtime = io->substream->runtime; + + return io->dma + samples_to_bytes(runtime, io->buff_sample_pos); +} + static void fsi_dma_complete(void *data) { struct fsi_stream *io = (struct fsi_stream *)data; @@ -944,7 +1068,7 @@ static void fsi_dma_complete(void *data) enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ? DMA_TO_DEVICE : DMA_FROM_DEVICE; - dma_sync_single_for_cpu(dai->dev, io->dma, + dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io), samples_to_bytes(runtime, io->period_samples), dir); io->buff_sample_pos += io->period_samples; @@ -961,24 +1085,14 @@ static void fsi_dma_complete(void *data) snd_pcm_period_elapsed(io->substream); } -static dma_addr_t fsi_dma_get_area(struct fsi_stream *io) -{ - struct snd_pcm_runtime *runtime = io->substream->runtime; - - return io->dma + samples_to_bytes(runtime, io->buff_sample_pos); -} - static void fsi_dma_do_tasklet(unsigned long data) { struct fsi_stream *io = (struct fsi_stream *)data; struct fsi_priv *fsi = fsi_stream_to_priv(io); - struct dma_chan *chan; struct snd_soc_dai *dai; struct dma_async_tx_descriptor *desc; - struct scatterlist sg; struct snd_pcm_runtime *runtime; enum dma_data_direction dir; - dma_cookie_t cookie; int is_play = fsi_stream_is_play(fsi, io); int len; dma_addr_t buf; @@ -987,22 +1101,15 @@ static void fsi_dma_do_tasklet(unsigned long data) return; dai = fsi_get_dai(io->substream); - chan = io->chan; runtime = io->substream->runtime; dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; len = samples_to_bytes(runtime, io->period_samples); buf = fsi_dma_get_area(io); - dma_sync_single_for_device(dai->dev, io->dma, len, dir); - - sg_init_table(&sg, 1); - sg_set_page(&sg, pfn_to_page(PFN_DOWN(buf)), - len , offset_in_page(buf)); - sg_dma_address(&sg) = buf; - sg_dma_len(&sg) = len; + dma_sync_single_for_device(dai->dev, buf, len, dir); - desc = dmaengine_prep_slave_sg(chan, &sg, 1, dir, - DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + desc = dmaengine_prep_slave_single(io->chan, buf, len, dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); if (!desc) { dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); return; @@ -1011,13 +1118,12 @@ static void fsi_dma_do_tasklet(unsigned long data) desc->callback = fsi_dma_complete; desc->callback_param = io; - cookie = desc->tx_submit(desc); - if (cookie < 0) { + if (dmaengine_submit(desc) < 0) { dev_err(dai->dev, "tx_submit() fail\n"); return; } - dma_async_issue_pending(chan); + dma_async_issue_pending(io->chan); /* * FIXME @@ -1055,28 +1161,19 @@ static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, int start) { - u32 bws; - u32 dma; + struct fsi_master *master = fsi_get_master(fsi); + u32 clk = fsi_is_port_a(fsi) ? CRA : CRB; + u32 enable = start ? DMA_ON : 0; - switch (io->sample_width * start) { - case 2: - bws = CR_BWS_16; - dma = VDMD_STREAM | DMA_ON; - break; - case 4: - bws = CR_BWS_24; - dma = VDMD_BACK | DMA_ON; - break; - default: - bws = 0; - dma = 0; - } + fsi_reg_mask_set(fsi, OUT_DMAC, DMA_ON, enable); + + dmaengine_terminate_all(io->chan); - fsi_reg_mask_set(fsi, DO_FMT, CR_BWS_MASK, bws); - fsi_reg_write(fsi, OUT_DMAC, dma); + if (fsi_is_clk_master(fsi)) + fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); } -static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io) +static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev) { dma_cap_mask_t mask; @@ -1084,8 +1181,19 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io) dma_cap_set(DMA_SLAVE, mask); io->chan = dma_request_channel(mask, fsi_dma_filter, &io->slave); - if (!io->chan) - return -EIO; + if (!io->chan) { + + /* switch to PIO handler */ + if (fsi_stream_is_play(fsi, io)) + fsi->playback.handler = &fsi_pio_push_handler; + else + fsi->capture.handler = &fsi_pio_pop_handler; + + dev_info(dev, "switch handler (dma => pio)\n"); + + /* probe again */ + return fsi_stream_probe(fsi, dev); + } tasklet_init(&io->tasklet, fsi_dma_do_tasklet, (unsigned long)io); @@ -1176,8 +1284,6 @@ static int fsi_hw_startup(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev) { - struct fsi_master *master = fsi_get_master(fsi); - int fsi_ver = master->core->ver; u32 flags = fsi_get_info_flags(fsi); u32 data = 0; @@ -1200,10 +1306,6 @@ static int fsi_hw_startup(struct fsi_priv *fsi, fsi_reg_write(fsi, CKG2, data); - /* set format */ - fsi_reg_write(fsi, DO_FMT, fsi->do_fmt); - fsi_reg_write(fsi, DI_FMT, fsi->di_fmt); - /* spdif ? */ if (fsi_is_spdif(fsi)) { fsi_spdif_clk_ctrl(fsi, 1); @@ -1211,15 +1313,18 @@ static int fsi_hw_startup(struct fsi_priv *fsi, } /* - * FIXME - * - * FSI driver assumed that data package is in-back. - * FSI2 chip can select it. + * get bus settings */ - if (fsi_ver >= 2) { - fsi_reg_write(fsi, OUT_DMAC, (1 << 4)); - fsi_reg_write(fsi, IN_DMAC, (1 << 4)); + data = 0; + switch (io->sample_width) { + case 2: + data = BUSOP_GET(16, io->bus_option); + break; + case 4: + data = BUSOP_GET(24, io->bus_option); + break; } + fsi_format_bus_setup(fsi, io, data, dev); /* irq clear */ fsi_irq_disable(fsi, io); @@ -1243,7 +1348,9 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, { struct fsi_priv *fsi = fsi_get_priv(substream); - return fsi_hw_startup(fsi, fsi_stream_get(fsi, substream), dai->dev); + fsi->rate = 0; + + return 0; } static void fsi_dai_shutdown(struct snd_pcm_substream *substream, @@ -1251,7 +1358,6 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream, { struct fsi_priv *fsi = fsi_get_priv(substream); - fsi_hw_shutdown(fsi, dai->dev); fsi->rate = 0; } @@ -1265,11 +1371,13 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: fsi_stream_init(fsi, io, substream); + fsi_hw_startup(fsi, io, dai->dev); ret = fsi_stream_transfer(io); if (0 == ret) fsi_stream_start(fsi, io); break; case SNDRV_PCM_TRIGGER_STOP: + fsi_hw_shutdown(fsi, dai->dev); fsi_stream_stop(fsi, io); fsi_stream_quit(fsi, io); break; @@ -1280,42 +1388,33 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, static int fsi_set_fmt_dai(struct fsi_priv *fsi, unsigned int fmt) { - u32 data = 0; - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - data = CR_I2S; + fsi->fmt = CR_I2S; fsi->chan_num = 2; break; case SND_SOC_DAIFMT_LEFT_J: - data = CR_PCM; + fsi->fmt = CR_PCM; fsi->chan_num = 2; break; default: return -EINVAL; } - fsi->do_fmt = data; - fsi->di_fmt = data; - return 0; } static int fsi_set_fmt_spdif(struct fsi_priv *fsi) { struct fsi_master *master = fsi_get_master(fsi); - u32 data = 0; - if (master->core->ver < 2) + if (fsi_version(master) < 2) return -EINVAL; - data = CR_BWS_16 | CR_DTMD_SPDIF_PCM | CR_PCM; + fsi->fmt = CR_DTMD_SPDIF_PCM | CR_PCM; fsi->chan_num = 2; fsi->spdif = 1; - fsi->do_fmt = data; - fsi->di_fmt = data; - return 0; } @@ -1532,8 +1631,8 @@ static void fsi_handler_init(struct fsi_priv *fsi) fsi->capture.priv = fsi; if (fsi->info->tx_id) { - fsi->playback.slave.slave_id = fsi->info->tx_id; - fsi->playback.handler = &fsi_dma_push_handler; + fsi->playback.slave.shdma_slave.slave_id = fsi->info->tx_id; + fsi->playback.handler = &fsi_dma_push_handler; } } @@ -1584,7 +1683,7 @@ static int fsi_probe(struct platform_device *pdev) master->fsia.master = master; master->fsia.info = &info->port_a; fsi_handler_init(&master->fsia); - ret = fsi_stream_probe(&master->fsia); + ret = fsi_stream_probe(&master->fsia, &pdev->dev); if (ret < 0) { dev_err(&pdev->dev, "FSIA stream probe failed\n"); goto exit_iounmap; @@ -1595,7 +1694,7 @@ static int fsi_probe(struct platform_device *pdev) master->fsib.master = master; master->fsib.info = &info->port_b; fsi_handler_init(&master->fsib); - ret = fsi_stream_probe(&master->fsib); + ret = fsi_stream_probe(&master->fsib, &pdev->dev); if (ret < 0) { dev_err(&pdev->dev, "FSIB stream probe failed\n"); goto exit_fsia; diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index 5cfcc655e95f..488f9becb44f 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -330,12 +330,9 @@ static bool filter(struct dma_chan *chan, void *slave) { struct sh_dmae_slave *param = slave; - pr_debug("%s: slave ID %d\n", __func__, param->slave_id); + pr_debug("%s: slave ID %d\n", __func__, param->shdma_slave.slave_id); - if (unlikely(param->dma_dev != chan->device->dev)) - return false; - - chan->private = param; + chan->private = ¶m->shdma_slave; return true; } @@ -360,16 +357,15 @@ static int siu_pcm_open(struct snd_pcm_substream *ss) if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) { siu_stream = &port_info->playback; param = &siu_stream->param; - param->slave_id = port ? pdata->dma_slave_tx_b : + param->shdma_slave.slave_id = port ? pdata->dma_slave_tx_b : pdata->dma_slave_tx_a; } else { siu_stream = &port_info->capture; param = &siu_stream->param; - param->slave_id = port ? pdata->dma_slave_rx_b : + param->shdma_slave.slave_id = port ? pdata->dma_slave_rx_b : pdata->dma_slave_rx_a; } - param->dma_dev = pdata->dma_dev; /* Get DMA channel */ siu_stream->chan = dma_request_channel(mask, filter, param); if (!siu_stream->chan) { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c88d9741b9e7..f219b2f7ee68 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -39,6 +39,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-dpcm.h> #include <sound/initval.h> #define CREATE_TRACE_POINTS @@ -54,7 +55,6 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); #endif static DEFINE_MUTEX(client_mutex); -static LIST_HEAD(card_list); static LIST_HEAD(dai_list); static LIST_HEAD(platform_list); static LIST_HEAD(codec_list); @@ -465,6 +465,35 @@ static inline void soc_cleanup_card_debugfs(struct snd_soc_card *card) } #endif +struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, + const char *dai_link, int stream) +{ + int i; + + for (i = 0; i < card->num_links; i++) { + if (card->rtd[i].dai_link->no_pcm && + !strcmp(card->rtd[i].dai_link->name, dai_link)) + return card->rtd[i].pcm->streams[stream].substream; + } + dev_dbg(card->dev, "failed to find dai link %s\n", dai_link); + return NULL; +} +EXPORT_SYMBOL_GPL(snd_soc_get_dai_substream); + +struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card, + const char *dai_link) +{ + int i; + + for (i = 0; i < card->num_links; i++) { + if (!strcmp(card->rtd[i].dai_link->name, dai_link)) + return &card->rtd[i]; + } + dev_dbg(card->dev, "failed to find rtd %s\n", dai_link); + return NULL; +} +EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime); + #ifdef CONFIG_SND_SOC_AC97_BUS /* unregister ac97 codec */ static int soc_ac97_dev_unregister(struct snd_soc_codec *codec) @@ -567,19 +596,16 @@ int snd_soc_suspend(struct device *dev) } for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; if (card->rtd[i].dai_link->ignore_suspend) continue; snd_soc_dapm_stream_event(&card->rtd[i], SNDRV_PCM_STREAM_PLAYBACK, - codec_dai, SND_SOC_DAPM_STREAM_SUSPEND); snd_soc_dapm_stream_event(&card->rtd[i], SNDRV_PCM_STREAM_CAPTURE, - codec_dai, SND_SOC_DAPM_STREAM_SUSPEND); } @@ -683,17 +709,16 @@ static void soc_resume_deferred(struct work_struct *work) } for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; if (card->rtd[i].dai_link->ignore_suspend) continue; snd_soc_dapm_stream_event(&card->rtd[i], - SNDRV_PCM_STREAM_PLAYBACK, codec_dai, + SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_RESUME); snd_soc_dapm_stream_event(&card->rtd[i], - SNDRV_PCM_STREAM_CAPTURE, codec_dai, + SNDRV_PCM_STREAM_CAPTURE, SND_SOC_DAPM_STREAM_RESUME); } @@ -783,37 +808,30 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) struct snd_soc_dai *codec_dai, *cpu_dai; const char *platform_name; - if (rtd->complete) - return 1; dev_dbg(card->dev, "binding %s at idx %d\n", dai_link->name, num); - /* do we already have the CPU DAI for this link ? */ - if (rtd->cpu_dai) { - goto find_codec; - } - /* no, then find CPU DAI from registered DAIs*/ + /* Find CPU DAI from registered DAIs*/ list_for_each_entry(cpu_dai, &dai_list, list) { - if (dai_link->cpu_dai_of_node) { - if (cpu_dai->dev->of_node != dai_link->cpu_dai_of_node) - continue; - } else { - if (strcmp(cpu_dai->name, dai_link->cpu_dai_name)) - continue; - } + if (dai_link->cpu_of_node && + (cpu_dai->dev->of_node != dai_link->cpu_of_node)) + continue; + if (dai_link->cpu_name && + strcmp(dev_name(cpu_dai->dev), dai_link->cpu_name)) + continue; + if (dai_link->cpu_dai_name && + strcmp(cpu_dai->name, dai_link->cpu_dai_name)) + continue; rtd->cpu_dai = cpu_dai; - goto find_codec; } - dev_dbg(card->dev, "CPU DAI %s not registered\n", - dai_link->cpu_dai_name); -find_codec: - /* do we already have the CODEC for this link ? */ - if (rtd->codec) { - goto find_platform; + if (!rtd->cpu_dai) { + dev_dbg(card->dev, "CPU DAI %s not registered\n", + dai_link->cpu_dai_name); + return -EPROBE_DEFER; } - /* no, then find CODEC from registered CODECs*/ + /* Find CODEC from registered CODECs */ list_for_each_entry(codec, &codec_list, list) { if (dai_link->codec_of_node) { if (codec->dev->of_node != dai_link->codec_of_node) @@ -835,28 +853,28 @@ find_codec: dai_link->codec_dai_name)) { rtd->codec_dai = codec_dai; - goto find_platform; } } - dev_dbg(card->dev, "CODEC DAI %s not registered\n", - dai_link->codec_dai_name); - goto find_platform; + if (!rtd->codec_dai) { + dev_dbg(card->dev, "CODEC DAI %s not registered\n", + dai_link->codec_dai_name); + return -EPROBE_DEFER; + } } - dev_dbg(card->dev, "CODEC %s not registered\n", - dai_link->codec_name); -find_platform: - /* do we need a platform? */ - if (rtd->platform) - goto out; + if (!rtd->codec) { + dev_dbg(card->dev, "CODEC %s not registered\n", + dai_link->codec_name); + return -EPROBE_DEFER; + } /* if there's no platform we match on the empty platform */ platform_name = dai_link->platform_name; if (!platform_name && !dai_link->platform_of_node) platform_name = "snd-soc-dummy"; - /* no, then find one from the set of registered platforms */ + /* find one from the set of registered platforms */ list_for_each_entry(platform, &platform_list, list) { if (dai_link->platform_of_node) { if (platform->dev->of_node != @@ -868,20 +886,38 @@ find_platform: } rtd->platform = platform; - goto out; } - - dev_dbg(card->dev, "platform %s not registered\n", + if (!rtd->platform) { + dev_dbg(card->dev, "platform %s not registered\n", dai_link->platform_name); + return -EPROBE_DEFER; + } + + card->num_rtd++; + return 0; +} -out: - /* mark rtd as complete if we found all 4 of our client devices */ - if (rtd->codec && rtd->codec_dai && rtd->platform && rtd->cpu_dai) { - rtd->complete = 1; - card->num_rtd++; +static int soc_remove_platform(struct snd_soc_platform *platform) +{ + int ret; + + if (platform->driver->remove) { + ret = platform->driver->remove(platform); + if (ret < 0) + pr_err("asoc: failed to remove %s: %d\n", + platform->name, ret); } - return 1; + + /* Make sure all DAPM widgets are freed */ + snd_soc_dapm_free(&platform->dapm); + + soc_cleanup_platform_debugfs(platform); + platform->probed = 0; + list_del(&platform->card_list); + module_put(platform->dev->driver->owner); + + return 0; } static void soc_remove_codec(struct snd_soc_codec *codec) @@ -905,11 +941,9 @@ static void soc_remove_codec(struct snd_soc_codec *codec) module_put(codec->dev->driver->owner); } -static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) +static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; int err; @@ -934,30 +968,6 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) list_del(&codec_dai->card_list); } - /* remove the platform */ - if (platform && platform->probed && - platform->driver->remove_order == order) { - if (platform->driver->remove) { - err = platform->driver->remove(platform); - if (err < 0) - pr_err("asoc: failed to remove %s: %d\n", - platform->name, err); - } - - /* Make sure all DAPM widgets are freed */ - snd_soc_dapm_free(&platform->dapm); - - soc_cleanup_platform_debugfs(platform); - platform->probed = 0; - list_del(&platform->card_list); - module_put(platform->dev->driver->owner); - } - - /* remove the CODEC */ - if (codec && codec->probed && - codec->driver->remove_order == order) - soc_remove_codec(codec); - /* remove the cpu_dai */ if (cpu_dai && cpu_dai->probed && cpu_dai->driver->remove_order == order) { @@ -969,7 +979,43 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) } cpu_dai->probed = 0; list_del(&cpu_dai->card_list); - module_put(cpu_dai->dev->driver->owner); + + if (!cpu_dai->codec) { + snd_soc_dapm_free(&cpu_dai->dapm); + module_put(cpu_dai->dev->driver->owner); + } + } +} + +static void soc_remove_link_components(struct snd_soc_card *card, int num, + int order) +{ + struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_codec *codec; + + /* remove the platform */ + if (platform && platform->probed && + platform->driver->remove_order == order) { + soc_remove_platform(platform); + } + + /* remove the CODEC-side CODEC */ + if (codec_dai) { + codec = codec_dai->codec; + if (codec && codec->probed && + codec->driver->remove_order == order) + soc_remove_codec(codec); + } + + /* remove any CPU-side CODEC */ + if (cpu_dai) { + codec = cpu_dai->codec; + if (codec && codec->probed && + codec->driver->remove_order == order) + soc_remove_codec(codec); } } @@ -980,8 +1026,15 @@ static void soc_remove_dai_links(struct snd_soc_card *card) for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { for (dai = 0; dai < card->num_rtd; dai++) - soc_remove_dai_link(card, dai, order); + soc_remove_link_dais(card, dai, order); + } + + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + for (dai = 0; dai < card->num_rtd; dai++) + soc_remove_link_components(card, dai, order); } + card->num_rtd = 0; } @@ -1042,6 +1095,10 @@ static int soc_probe_codec(struct snd_soc_card *card, } } + /* If the driver didn't set I/O up try regmap */ + if (!codec->control_data) + snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + if (driver->controls) snd_soc_add_codec_controls(codec, driver->controls, driver->num_controls); @@ -1068,6 +1125,7 @@ static int soc_probe_platform(struct snd_soc_card *card, { int ret = 0; const struct snd_soc_platform_driver *driver = platform->driver; + struct snd_soc_dai *dai; platform->card = card; platform->dapm.card = card; @@ -1081,6 +1139,14 @@ static int soc_probe_platform(struct snd_soc_card *card, snd_soc_dapm_new_controls(&platform->dapm, driver->dapm_widgets, driver->num_dapm_widgets); + /* Create DAPM widgets for each DAI stream */ + list_for_each_entry(dai, &dai_list, list) { + if (dai->dev != platform->dev) + continue; + + snd_soc_dapm_new_dai_widgets(&platform->dapm, dai); + } + platform->dapm.idle_bias_off = 1; if (driver->probe) { @@ -1170,6 +1236,10 @@ static int soc_post_component_init(struct snd_soc_card *card, rtd->dev->init_name = name; dev_set_drvdata(rtd->dev, rtd); mutex_init(&rtd->pcm_mutex); + INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients); + INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients); + INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients); + INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients); ret = device_add(rtd->dev); if (ret < 0) { dev_err(card->dev, @@ -1191,23 +1261,72 @@ static int soc_post_component_init(struct snd_soc_card *card, dev_err(codec->dev, "asoc: failed to add codec sysfs files: %d\n", ret); +#ifdef CONFIG_DEBUG_FS + /* add DPCM sysfs entries */ + if (!dailess && !dai_link->dynamic) + goto out; + + ret = soc_dpcm_debugfs_add(rtd); + if (ret < 0) + dev_err(rtd->dev, "asoc: failed to add dpcm sysfs entries: %d\n", ret); + +out: +#endif return 0; } -static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) +static int soc_probe_link_components(struct snd_soc_card *card, int num, + int order) +{ + struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_platform *platform = rtd->platform; + int ret; + + /* probe the CPU-side component, if it is a CODEC */ + if (cpu_dai->codec && + !cpu_dai->codec->probed && + cpu_dai->codec->driver->probe_order == order) { + ret = soc_probe_codec(card, cpu_dai->codec); + if (ret < 0) + return ret; + } + + /* probe the CODEC-side component */ + if (!codec_dai->codec->probed && + codec_dai->codec->driver->probe_order == order) { + ret = soc_probe_codec(card, codec_dai->codec); + if (ret < 0) + return ret; + } + + /* probe the platform */ + if (!platform->probed && + platform->driver->probe_order == order) { + ret = soc_probe_platform(card, platform); + if (ret < 0) + return ret; + } + + return 0; +} + +static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_codec *codec = rtd->codec; struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dapm_widget *play_w, *capture_w; int ret; dev_dbg(card->dev, "probe %s dai link %d late %d\n", card->name, num, order); /* config components */ - codec_dai->codec = codec; cpu_dai->platform = platform; codec_dai->card = card; cpu_dai->card = card; @@ -1218,8 +1337,14 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) /* probe the cpu_dai */ if (!cpu_dai->probed && cpu_dai->driver->probe_order == order) { - if (!try_module_get(cpu_dai->dev->driver->owner)) - return -ENODEV; + if (!cpu_dai->codec) { + cpu_dai->dapm.card = card; + if (!try_module_get(cpu_dai->dev->driver->owner)) + return -ENODEV; + + list_add(&cpu_dai->dapm.list, &card->dapm_list); + snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai); + } if (cpu_dai->driver->probe) { ret = cpu_dai->driver->probe(cpu_dai); @@ -1235,22 +1360,6 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) list_add(&cpu_dai->card_list, &card->dai_dev_list); } - /* probe the CODEC */ - if (!codec->probed && - codec->driver->probe_order == order) { - ret = soc_probe_codec(card, codec); - if (ret < 0) - return ret; - } - - /* probe the platform */ - if (!platform->probed && - platform->driver->probe_order == order) { - ret = soc_probe_platform(card, platform); - if (ret < 0) - return ret; - } - /* probe the CODEC DAI */ if (!codec_dai->probed && codec_dai->driver->probe_order == order) { if (codec_dai->driver->probe) { @@ -1279,12 +1388,39 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) if (ret < 0) pr_warn("asoc: failed to add pmdown_time sysfs:%d\n", ret); - /* create the pcm */ - ret = soc_new_pcm(rtd, num); - if (ret < 0) { - pr_err("asoc: can't create pcm %s :%d\n", - dai_link->stream_name, ret); - return ret; + if (!dai_link->params) { + /* create the pcm */ + ret = soc_new_pcm(rtd, num); + if (ret < 0) { + pr_err("asoc: can't create pcm %s :%d\n", + dai_link->stream_name, ret); + return ret; + } + } else { + /* link the DAI widgets */ + play_w = codec_dai->playback_widget; + capture_w = cpu_dai->capture_widget; + if (play_w && capture_w) { + ret = snd_soc_dapm_new_pcm(card, dai_link->params, + capture_w, play_w); + if (ret != 0) { + dev_err(card->dev, "Can't link %s to %s: %d\n", + play_w->name, capture_w->name, ret); + return ret; + } + } + + play_w = cpu_dai->playback_widget; + capture_w = codec_dai->capture_widget; + if (play_w && capture_w) { + ret = snd_soc_dapm_new_pcm(card, dai_link->params, + capture_w, play_w); + if (ret != 0) { + dev_err(card->dev, "Can't link %s to %s: %d\n", + play_w->name, capture_w->name, ret); + return ret; + } + } } /* add platform data for AC97 devices */ @@ -1334,6 +1470,20 @@ static void soc_unregister_ac97_dai_link(struct snd_soc_codec *codec) } #endif +static int soc_check_aux_dev(struct snd_soc_card *card, int num) +{ + struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; + struct snd_soc_codec *codec; + + /* find CODEC from registered CODECs*/ + list_for_each_entry(codec, &codec_list, list) { + if (!strcmp(codec->name, aux_dev->codec_name)) + return 0; + } + + return -EPROBE_DEFER; +} + static int soc_probe_aux_dev(struct snd_soc_card *card, int num) { struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; @@ -1354,7 +1504,7 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) } /* codec not found */ dev_err(card->dev, "asoc: codec %s not found", aux_dev->codec_name); - goto out; + return -EPROBE_DEFER; found: ret = soc_probe_codec(card, codec); @@ -1404,29 +1554,28 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec, return 0; } -static void snd_soc_instantiate_card(struct snd_soc_card *card) +static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_codec *codec; struct snd_soc_codec_conf *codec_conf; enum snd_soc_compress_type compress_type; struct snd_soc_dai_link *dai_link; - int ret, i, order; + int ret, i, order, dai_fmt; - mutex_lock(&card->mutex); - - if (card->instantiated) { - mutex_unlock(&card->mutex); - return; - } + mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); /* bind DAIs */ - for (i = 0; i < card->num_links; i++) - soc_bind_dai_link(card, i); + for (i = 0; i < card->num_links; i++) { + ret = soc_bind_dai_link(card, i); + if (ret != 0) + goto base_error; + } - /* bind completed ? */ - if (card->num_rtd != card->num_links) { - mutex_unlock(&card->mutex); - return; + /* check aux_devs too */ + for (i = 0; i < card->num_aux_devs; i++) { + ret = soc_check_aux_dev(card, i); + if (ret != 0) + goto base_error; } /* initialize the register cache for each available codec */ @@ -1446,10 +1595,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } } ret = snd_soc_init_codec_cache(codec, compress_type); - if (ret < 0) { - mutex_unlock(&card->mutex); - return; - } + if (ret < 0) + goto base_error; } /* card bind complete so register a sound card */ @@ -1458,8 +1605,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) if (ret < 0) { pr_err("asoc: can't create sound card for card %s: %d\n", card->name, ret); - mutex_unlock(&card->mutex); - return; + goto base_error; } card->snd_card->dev = card->dev; @@ -1488,14 +1634,27 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) goto card_probe_error; } - /* early DAI link probe */ + /* probe all components used by DAI links on this card */ for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { for (i = 0; i < card->num_links; i++) { - ret = soc_probe_dai_link(card, i, order); + ret = soc_probe_link_components(card, i, order); if (ret < 0) { pr_err("asoc: failed to instantiate card %s: %d\n", - card->name, ret); + card->name, ret); + goto probe_dai_err; + } + } + } + + /* probe all DAI links on this card */ + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + for (i = 0; i < card->num_links; i++) { + ret = soc_probe_link_dais(card, i, order); + if (ret < 0) { + pr_err("asoc: failed to instantiate card %s: %d\n", + card->name, ret); goto probe_dai_err; } } @@ -1523,17 +1682,47 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) for (i = 0; i < card->num_links; i++) { dai_link = &card->dai_link[i]; + dai_fmt = dai_link->dai_fmt; - if (dai_link->dai_fmt) { + if (dai_fmt) { ret = snd_soc_dai_set_fmt(card->rtd[i].codec_dai, - dai_link->dai_fmt); + dai_fmt); if (ret != 0 && ret != -ENOTSUPP) dev_warn(card->rtd[i].codec_dai->dev, "Failed to set DAI format: %d\n", ret); + } + + /* If this is a regular CPU link there will be a platform */ + if (dai_fmt && + (dai_link->platform_name || dai_link->platform_of_node)) { + ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai, + dai_fmt); + if (ret != 0 && ret != -ENOTSUPP) + dev_warn(card->rtd[i].cpu_dai->dev, + "Failed to set DAI format: %d\n", + ret); + } else if (dai_fmt) { + /* Flip the polarity for the "CPU" end */ + dai_fmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + switch (dai_link->dai_fmt & + SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + break; + case SND_SOC_DAIFMT_CBM_CFS: + dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; + break; + case SND_SOC_DAIFMT_CBS_CFM: + dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; + break; + case SND_SOC_DAIFMT_CBS_CFS: + dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + break; + } ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai, - dai_link->dai_fmt); + dai_fmt); if (ret != 0 && ret != -ENOTSUPP) dev_warn(card->rtd[i].cpu_dai->dev, "Failed to set DAI format: %d\n", @@ -1599,7 +1788,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) card->instantiated = 1; snd_soc_dapm_sync(&card->dapm); mutex_unlock(&card->mutex); - return; + + return 0; probe_aux_dev_err: for (i = 0; i < card->num_aux_devs; i++) @@ -1614,18 +1804,10 @@ card_probe_error: snd_card_free(card->snd_card); +base_error: mutex_unlock(&card->mutex); -} -/* - * Attempt to initialise any uninitialised cards. Must be called with - * client_mutex. - */ -static void snd_soc_instantiate_cards(void) -{ - struct snd_soc_card *card; - list_for_each_entry(card, &card_list, list) - snd_soc_instantiate_card(card); + return ret; } /* probes a new socdev */ @@ -2527,6 +2709,87 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_volsw); /** + * snd_soc_get_volsw_sx - single mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a single mixer control, or a double mixer + * control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int mask = (1 << (fls(min + max) - 1)) - 1; + + ucontrol->value.integer.value[0] = + ((snd_soc_read(codec, reg) >> shift) - min) & mask; + + if (snd_soc_volsw_is_stereo(mc)) + ucontrol->value.integer.value[1] = + ((snd_soc_read(codec, reg2) >> rshift) - min) & mask; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); + +/** + * snd_soc_put_volsw_sx - double mixer set callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to set the value of a double mixer control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int mask = (1 << (fls(min + max) - 1)) - 1; + int err = 0; + unsigned short val, val_mask, val2 = 0; + + val_mask = mask << shift; + val = (ucontrol->value.integer.value[0] + min) & mask; + val = val << shift; + + if (snd_soc_update_bits_locked(codec, reg, val_mask, val)) + return err; + + if (snd_soc_volsw_is_stereo(mc)) { + val_mask = mask << rshift; + val2 = (ucontrol->value.integer.value[1] + min) & mask; + val2 = val2 << rshift; + + if (snd_soc_update_bits_locked(codec, reg2, val_mask, val2)) + return err; + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_sx); + +/** * snd_soc_info_volsw_s8 - signed mixer info callback * @kcontrol: mixer control * @uinfo: control element information @@ -2609,136 +2872,141 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); /** - * snd_soc_limit_volume - Set new limit to an existing volume control. - * - * @codec: where to look for the control - * @name: Name of the control - * @max: new maximum limit - * - * Return 0 for success, else error. - */ -int snd_soc_limit_volume(struct snd_soc_codec *codec, - const char *name, int max) -{ - struct snd_card *card = codec->card->snd_card; - struct snd_kcontrol *kctl; - struct soc_mixer_control *mc; - int found = 0; - int ret = -EINVAL; - - /* Sanity check for name and max */ - if (unlikely(!name || max <= 0)) - return -EINVAL; - - list_for_each_entry(kctl, &card->controls, list) { - if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) { - found = 1; - break; - } - } - if (found) { - mc = (struct soc_mixer_control *)kctl->private_value; - if (max <= mc->max) { - mc->platform_max = max; - ret = 0; - } - } - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_limit_volume); - -/** - * snd_soc_info_volsw_2r_sx - double with tlv and variable data size - * mixer info callback + * snd_soc_info_volsw_range - single mixer info callback with range. * @kcontrol: mixer control * @uinfo: control element information * - * Returns 0 for success. + * Callback to provide information, within a range, about a single + * mixer control. + * + * returns 0 for success. */ -int snd_soc_info_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) +int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int max = mc->max; + int platform_max; int min = mc->min; + if (!mc->platform_max) + mc->platform_max = mc->max; + platform_max = mc->platform_max; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; + uinfo->count = 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = max-min; + uinfo->value.integer.max = platform_max - min; return 0; } -EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r_sx); +EXPORT_SYMBOL_GPL(snd_soc_info_volsw_range); /** - * snd_soc_get_volsw_2r_sx - double with tlv and variable data size - * mixer get callback + * snd_soc_put_volsw_range - single mixer put value callback with range. * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information + * + * Callback to set the value, within a range, for a single mixer control. * * Returns 0 for success. */ -int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int mask = (1<<mc->shift)-1; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; int min = mc->min; - int val = snd_soc_read(codec, mc->reg) & mask; - int valr = snd_soc_read(codec, mc->rreg) & mask; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + unsigned int val, val_mask; - ucontrol->value.integer.value[0] = ((val & 0xff)-min) & mask; - ucontrol->value.integer.value[1] = ((valr & 0xff)-min) & mask; - return 0; + val = ((ucontrol->value.integer.value[0] + min) & mask); + if (invert) + val = max - val; + val_mask = mask << shift; + val = val << shift; + + return snd_soc_update_bits_locked(codec, reg, val_mask, val); } -EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r_sx); +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range); /** - * snd_soc_put_volsw_2r_sx - double with tlv and variable data size - * mixer put callback + * snd_soc_get_volsw_range - single mixer get callback with range * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information + * + * Callback to get the value, within a range, of a single mixer control. * * Returns 0 for success. */ -int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int mask = (1<<mc->shift)-1; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; int min = mc->min; - int ret; - unsigned int val, valr, oval, ovalr; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; - val = ((ucontrol->value.integer.value[0]+min) & 0xff); - val &= mask; - valr = ((ucontrol->value.integer.value[1]+min) & 0xff); - valr &= mask; + ucontrol->value.integer.value[0] = + (snd_soc_read(codec, reg) >> shift) & mask; + if (invert) + ucontrol->value.integer.value[0] = + max - ucontrol->value.integer.value[0]; + ucontrol->value.integer.value[0] = + ucontrol->value.integer.value[0] - min; - oval = snd_soc_read(codec, mc->reg) & mask; - ovalr = snd_soc_read(codec, mc->rreg) & mask; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); - ret = 0; - if (oval != val) { - ret = snd_soc_write(codec, mc->reg, val); - if (ret < 0) - return ret; +/** + * snd_soc_limit_volume - Set new limit to an existing volume control. + * + * @codec: where to look for the control + * @name: Name of the control + * @max: new maximum limit + * + * Return 0 for success, else error. + */ +int snd_soc_limit_volume(struct snd_soc_codec *codec, + const char *name, int max) +{ + struct snd_card *card = codec->card->snd_card; + struct snd_kcontrol *kctl; + struct soc_mixer_control *mc; + int found = 0; + int ret = -EINVAL; + + /* Sanity check for name and max */ + if (unlikely(!name || max <= 0)) + return -EINVAL; + + list_for_each_entry(kctl, &card->controls, list) { + if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) { + found = 1; + break; + } } - if (ovalr != valr) { - ret = snd_soc_write(codec, mc->rreg, valr); - if (ret < 0) - return ret; + if (found) { + mc = (struct soc_mixer_control *)kctl->private_value; + if (max <= mc->max) { + mc->platform_max = max; + ret = 0; + } } - - return 0; + return ret; } -EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r_sx); +EXPORT_SYMBOL_GPL(snd_soc_limit_volume); int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -2850,6 +3118,186 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_bytes_put); /** + * snd_soc_info_xr_sx - signed multi register info callback + * @kcontrol: mreg control + * @uinfo: control element information + * + * Callback to provide information of a control that can + * span multiple codec registers which together + * forms a single signed value in a MSB/LSB manner. + * + * Returns 0 for success. + */ +int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = mc->min; + uinfo->value.integer.max = mc->max; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_xr_sx); + +/** + * snd_soc_get_xr_sx - signed multi register get callback + * @kcontrol: mreg control + * @ucontrol: control element information + * + * Callback to get the value of a control that can span + * multiple codec registers which together forms a single + * signed value in a MSB/LSB manner. The control supports + * specifying total no of bits used to allow for bitfields + * across the multiple codec registers. + * + * Returns 0 for success. + */ +int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int regbase = mc->regbase; + unsigned int regcount = mc->regcount; + unsigned int regwshift = codec->driver->reg_word_size * BITS_PER_BYTE; + unsigned int regwmask = (1<<regwshift)-1; + unsigned int invert = mc->invert; + unsigned long mask = (1UL<<mc->nbits)-1; + long min = mc->min; + long max = mc->max; + long val = 0; + unsigned long regval; + unsigned int i; + + for (i = 0; i < regcount; i++) { + regval = snd_soc_read(codec, regbase+i) & regwmask; + val |= regval << (regwshift*(regcount-i-1)); + } + val &= mask; + if (min < 0 && val > max) + val |= ~mask; + if (invert) + val = max - val; + ucontrol->value.integer.value[0] = val; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_xr_sx); + +/** + * snd_soc_put_xr_sx - signed multi register get callback + * @kcontrol: mreg control + * @ucontrol: control element information + * + * Callback to set the value of a control that can span + * multiple codec registers which together forms a single + * signed value in a MSB/LSB manner. The control supports + * specifying total no of bits used to allow for bitfields + * across the multiple codec registers. + * + * Returns 0 for success. + */ +int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int regbase = mc->regbase; + unsigned int regcount = mc->regcount; + unsigned int regwshift = codec->driver->reg_word_size * BITS_PER_BYTE; + unsigned int regwmask = (1<<regwshift)-1; + unsigned int invert = mc->invert; + unsigned long mask = (1UL<<mc->nbits)-1; + long max = mc->max; + long val = ucontrol->value.integer.value[0]; + unsigned int i, regval, regmask; + int err; + + if (invert) + val = max - val; + val &= mask; + for (i = 0; i < regcount; i++) { + regval = (val >> (regwshift*(regcount-i-1))) & regwmask; + regmask = (mask >> (regwshift*(regcount-i-1))) & regwmask; + err = snd_soc_update_bits_locked(codec, regbase+i, + regmask, regval); + if (err < 0) + return err; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_put_xr_sx); + +/** + * snd_soc_get_strobe - strobe get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback get the value of a strobe mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = 1 << shift; + unsigned int invert = mc->invert != 0; + unsigned int val = snd_soc_read(codec, reg) & mask; + + if (shift != 0 && val != 0) + val = val >> shift; + ucontrol->value.enumerated.item[0] = val ^ invert; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_strobe); + +/** + * snd_soc_put_strobe - strobe put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback strobe a register bit to high then low (or the inverse) + * in one pass of a single mixer enum control. + * + * Returns 1 for success. + */ +int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = 1 << shift; + unsigned int invert = mc->invert != 0; + unsigned int strobe = ucontrol->value.enumerated.item[0] != 0; + unsigned int val1 = (strobe ^ invert) ? mask : 0; + unsigned int val2 = (strobe ^ invert) ? 0 : mask; + int err; + + err = snd_soc_update_bits_locked(codec, reg, mask, val1); + if (err < 0) + return err; + + err = snd_soc_update_bits_locked(codec, reg, mask, val2); + return err; +} +EXPORT_SYMBOL_GPL(snd_soc_put_strobe); + +/** * snd_soc_dai_set_sysclk - configure DAI system or master clock. * @dai: DAI * @clk_id: DAI specific clock ID @@ -3048,7 +3496,7 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) if (dai->driver && dai->driver->ops->digital_mute) return dai->driver->ops->digital_mute(dai, mute); else - return -EINVAL; + return -ENOTSUPP; } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); @@ -3060,7 +3508,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); */ int snd_soc_register_card(struct snd_soc_card *card) { - int i; + int i, ret; if (!card->name || !card->dev) return -EINVAL; @@ -3078,6 +3526,12 @@ int snd_soc_register_card(struct snd_soc_card *card) link->name); return -EINVAL; } + /* Codec DAI name must be specified */ + if (!link->codec_dai_name) { + dev_err(card->dev, "codec_dai_name not set for %s\n", + link->name); + return -EINVAL; + } /* * Platform may be specified by either name or OF node, but @@ -3090,12 +3544,24 @@ int snd_soc_register_card(struct snd_soc_card *card) } /* - * CPU DAI must be specified by 1 of name or OF node, - * not both or neither. + * CPU device may be specified by either name or OF node, but + * can be left unspecified, and will be matched based on DAI + * name alone.. + */ + if (link->cpu_name && link->cpu_of_node) { + dev_err(card->dev, + "Neither/both cpu name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + /* + * At least one of CPU DAI name or CPU device name/node must be + * specified */ - if (!!link->cpu_dai_name == !!link->cpu_dai_of_node) { + if (!link->cpu_dai_name && + !(link->cpu_name || link->cpu_of_node)) { dev_err(card->dev, - "Neither/both cpu_dai name/of_node are set for %s\n", + "Neither cpu_dai_name nor cpu_name/of_node are set for %s\n", link->name); return -EINVAL; } @@ -3123,15 +3589,13 @@ int snd_soc_register_card(struct snd_soc_card *card) INIT_LIST_HEAD(&card->dapm_dirty); card->instantiated = 0; mutex_init(&card->mutex); + mutex_init(&card->dapm_mutex); - mutex_lock(&client_mutex); - list_add(&card->list, &card_list); - snd_soc_instantiate_cards(); - mutex_unlock(&client_mutex); + ret = snd_soc_instantiate_card(card); + if (ret != 0) + soc_cleanup_card_debugfs(card); - dev_dbg(card->dev, "Registered card '%s'\n", card->name); - - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_register_card); @@ -3145,9 +3609,6 @@ int snd_soc_unregister_card(struct snd_soc_card *card) { if (card->instantiated) soc_cleanup_card_resources(card); - mutex_lock(&client_mutex); - list_del(&card->list); - mutex_unlock(&client_mutex); dev_dbg(card->dev, "Unregistered card '%s'\n", card->name); return 0; @@ -3221,6 +3682,7 @@ static inline char *fmt_multiple_name(struct device *dev, int snd_soc_register_dai(struct device *dev, struct snd_soc_dai_driver *dai_drv) { + struct snd_soc_codec *codec; struct snd_soc_dai *dai; dev_dbg(dev, "dai register %s\n", dev_name(dev)); @@ -3238,12 +3700,23 @@ int snd_soc_register_dai(struct device *dev, dai->dev = dev; dai->driver = dai_drv; + dai->dapm.dev = dev; if (!dai->driver->ops) dai->driver->ops = &null_dai_ops; mutex_lock(&client_mutex); + + list_for_each_entry(codec, &codec_list, list) { + if (codec->dev == dev) { + dev_dbg(dev, "Mapped DAI %s to CODEC %s\n", + dai->name, codec->name); + dai->codec = codec; + break; + } + } + list_add(&dai->list, &dai_list); - snd_soc_instantiate_cards(); + mutex_unlock(&client_mutex); pr_debug("Registered DAI '%s'\n", dai->name); @@ -3287,6 +3760,7 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_dai); int snd_soc_register_dais(struct device *dev, struct snd_soc_dai_driver *dai_drv, size_t count) { + struct snd_soc_codec *codec; struct snd_soc_dai *dai; int i, ret = 0; @@ -3314,19 +3788,28 @@ int snd_soc_register_dais(struct device *dev, dai->id = dai->driver->id; else dai->id = i; + dai->dapm.dev = dev; if (!dai->driver->ops) dai->driver->ops = &null_dai_ops; mutex_lock(&client_mutex); + + list_for_each_entry(codec, &codec_list, list) { + if (codec->dev == dev) { + dev_dbg(dev, "Mapped DAI %s to CODEC %s\n", + dai->name, codec->name); + dai->codec = codec; + break; + } + } + list_add(&dai->list, &dai_list); + mutex_unlock(&client_mutex); pr_debug("Registered DAI '%s'\n", dai->name); } - mutex_lock(&client_mutex); - snd_soc_instantiate_cards(); - mutex_unlock(&client_mutex); return 0; err: @@ -3384,7 +3867,6 @@ int snd_soc_register_platform(struct device *dev, mutex_lock(&client_mutex); list_add(&platform->list, &platform_list); - snd_soc_instantiate_cards(); mutex_unlock(&client_mutex); pr_debug("Registered platform '%s'\n", platform->name); @@ -3534,18 +4016,18 @@ int snd_soc_register_codec(struct device *dev, fixup_codec_formats(&dai_drv[i].capture); } + mutex_lock(&client_mutex); + list_add(&codec->list, &codec_list); + mutex_unlock(&client_mutex); + /* register any DAIs */ if (num_dai) { ret = snd_soc_register_dais(dev, dai_drv, num_dai); if (ret < 0) - goto fail; + dev_err(codec->dev, "Failed to regster DAIs: %d\n", + ret); } - mutex_lock(&client_mutex); - list_add(&codec->list, &codec_list); - snd_soc_instantiate_cards(); - mutex_unlock(&client_mutex); - pr_debug("Registered codec '%s'\n", codec->name); return 0; @@ -3654,6 +4136,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "Property '%s' index %d could not be read: %d\n", propname, 2 * i, ret); + kfree(routes); return -EINVAL; } ret = of_property_read_string_index(np, propname, @@ -3662,6 +4145,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "Property '%s' index %d could not be read: %d\n", propname, (2 * i) + 1, ret); + kfree(routes); return -EINVAL; } } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1bb6d4a63cd8..dd7c49fafd75 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -35,6 +35,7 @@ #include <linux/debugfs.h> #include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> +#include <linux/clk.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -51,7 +52,9 @@ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, [snd_soc_dapm_supply] = 1, [snd_soc_dapm_regulator_supply] = 1, + [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_micbias] = 2, + [snd_soc_dapm_dai_link] = 2, [snd_soc_dapm_dai] = 3, [snd_soc_dapm_aif_in] = 3, [snd_soc_dapm_aif_out] = 3, @@ -90,9 +93,11 @@ static int dapm_down_seq[] = { [snd_soc_dapm_aif_in] = 10, [snd_soc_dapm_aif_out] = 10, [snd_soc_dapm_dai] = 10, - [snd_soc_dapm_regulator_supply] = 11, - [snd_soc_dapm_supply] = 11, - [snd_soc_dapm_post] = 12, + [snd_soc_dapm_dai_link] = 11, + [snd_soc_dapm_clock_supply] = 12, + [snd_soc_dapm_regulator_supply] = 12, + [snd_soc_dapm_supply] = 12, + [snd_soc_dapm_post] = 13, }; static void pop_wait(u32 pop_time) @@ -208,7 +213,23 @@ static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) return -1; } -static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, +static inline void soc_widget_lock(struct snd_soc_dapm_widget *w) +{ + if (w->codec && !w->codec->using_regmap) + mutex_lock(&w->codec->mutex); + else if (w->platform) + mutex_lock(&w->platform->mutex); +} + +static inline void soc_widget_unlock(struct snd_soc_dapm_widget *w) +{ + if (w->codec && !w->codec->using_regmap) + mutex_unlock(&w->codec->mutex); + else if (w->platform) + mutex_unlock(&w->platform->mutex); +} + +static int soc_widget_update_bits_locked(struct snd_soc_dapm_widget *w, unsigned short reg, unsigned int mask, unsigned int value) { bool change; @@ -221,18 +242,24 @@ static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, if (ret != 0) return ret; } else { + soc_widget_lock(w); ret = soc_widget_read(w, reg); - if (ret < 0) + if (ret < 0) { + soc_widget_unlock(w); return ret; + } old = ret; new = (old & ~mask) | (value & mask); change = old != new; if (change) { ret = soc_widget_write(w, reg, new); - if (ret < 0) + if (ret < 0) { + soc_widget_unlock(w); return ret; + } } + soc_widget_unlock(w); } return change; @@ -264,9 +291,9 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, if (dapm->codec->driver->set_bias_level) ret = dapm->codec->driver->set_bias_level(dapm->codec, level); - else - dapm->bias_level = level; - } + } else + dapm->bias_level = level; + if (ret != 0) goto out; @@ -297,11 +324,10 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, val = soc_widget_read(w, reg); val = (val >> shift) & mask; + if (invert) + val = max - val; - if ((invert && !val) || (!invert && val)) - p->connect = 1; - else - p->connect = 0; + p->connect = !!val; } break; case snd_soc_dapm_mux: { @@ -367,6 +393,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_vmid: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: case snd_soc_dapm_dai: @@ -374,6 +401,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_mic: case snd_soc_dapm_spk: case snd_soc_dapm_line: + case snd_soc_dapm_dai_link: p->connect = 1; break; /* does affect routing - dynamically connected */ @@ -682,11 +710,51 @@ static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget) } } +/* add widget to list if it's not already in the list */ +static int dapm_list_add_widget(struct snd_soc_dapm_widget_list **list, + struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_widget_list *wlist; + int wlistsize, wlistentries, i; + + if (*list == NULL) + return -EINVAL; + + wlist = *list; + + /* is this widget already in the list */ + for (i = 0; i < wlist->num_widgets; i++) { + if (wlist->widgets[i] == w) + return 0; + } + + /* allocate some new space */ + wlistentries = wlist->num_widgets + 1; + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + wlistentries * sizeof(struct snd_soc_dapm_widget *); + *list = krealloc(wlist, wlistsize, GFP_KERNEL); + if (*list == NULL) { + dev_err(w->dapm->dev, "can't allocate widget list for %s\n", + w->name); + return -ENOMEM; + } + wlist = *list; + + /* insert the widget */ + dev_dbg(w->dapm->dev, "added %s in widget list pos %d\n", + w->name, wlist->num_widgets); + + wlist->widgets[wlist->num_widgets] = w; + wlist->num_widgets++; + return 1; +} + /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. */ -static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -699,6 +767,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) switch (widget->id) { case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: return 0; default: break; @@ -742,9 +811,23 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) if (path->walked) continue; + trace_snd_soc_dapm_output_path(widget, path); + if (path->sink && path->connect) { path->walked = 1; - con += is_connected_output_ep(path->sink); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + err = dapm_list_add_widget(list, path->sink); + if (err < 0) { + dev_err(widget->dapm->dev, "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_output_ep(path->sink, list); } } @@ -757,7 +840,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. */ -static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -770,6 +854,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) switch (widget->id) { case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: return 0; default: break; @@ -825,9 +910,23 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) if (path->walked) continue; + trace_snd_soc_dapm_input_path(widget, path); + if (path->source && path->connect) { path->walked = 1; - con += is_connected_input_ep(path->source); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + err = dapm_list_add_widget(list, path->source); + if (err < 0) { + dev_err(widget->dapm->dev, "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_input_ep(path->source, list); } } @@ -836,6 +935,39 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return con; } +/** + * snd_soc_dapm_get_connected_widgets - query audio path and it's widgets. + * @dai: the soc DAI. + * @stream: stream direction. + * @list: list of active widgets for this stream. + * + * Queries DAPM graph as to whether an valid audio stream path exists for + * the initial stream specified by name. This takes into account + * current mixer and mux kcontrol settings. Creates list of valid widgets. + * + * Returns the number of valid paths or negative error. + */ +int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, + struct snd_soc_dapm_widget_list **list) +{ + struct snd_soc_card *card = dai->card; + int paths; + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + dapm_reset(card); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + paths = is_connected_output_ep(dai->playback_widget, list); + else + paths = is_connected_input_ep(dai->capture_widget, list); + + trace_snd_soc_dapm_connected(paths, stream); + dapm_clear_walk(&card->dapm); + mutex_unlock(&card->dapm_mutex); + + return paths; +} + /* * Handler for generic register modifier widget. */ @@ -849,7 +981,7 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, else val = w->off_val; - soc_widget_update_bits(w, -(w->reg + 1), + soc_widget_update_bits_locked(w, -(w->reg + 1), w->mask << w->shift, val << w->shift); return 0; @@ -863,12 +995,33 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - return regulator_enable(w->priv); + return regulator_enable(w->regulator); else - return regulator_disable_deferred(w->priv, w->shift); + return regulator_disable_deferred(w->regulator, w->shift); } EXPORT_SYMBOL_GPL(dapm_regulator_event); +/* + * Handler for clock supply widget. + */ +int dapm_clock_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (!w->clk) + return -EIO; + +#ifdef CONFIG_HAVE_CLK + if (SND_SOC_DAPM_EVENT_ON(event)) { + return clk_enable(w->clk); + } else { + clk_disable(w->clk); + return 0; + } +#endif + return 0; +} +EXPORT_SYMBOL_GPL(dapm_clock_event); + static int dapm_widget_power_check(struct snd_soc_dapm_widget *w) { if (w->power_checked) @@ -892,9 +1045,9 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0 && in != 0; } @@ -903,7 +1056,10 @@ static int dapm_dai_check_power(struct snd_soc_dapm_widget *w) { DAPM_UPDATE_STAT(w, power_checks); - return w->active; + if (w->active) + return w->active; + + return dapm_generic_check_power(w); } /* Check to see if an ADC has power */ @@ -914,7 +1070,7 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); if (w->active) { - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); return in != 0; } else { @@ -930,7 +1086,7 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); if (w->active) { - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0; } else { @@ -1107,7 +1263,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, "pop test : Applying 0x%x/0x%x to %x in %dms\n", value, mask, reg, card->pop_time); pop_wait(card->pop_time); - soc_widget_update_bits(w, reg, mask, value); + soc_widget_update_bits_locked(w, reg, mask, value); } list_for_each_entry(w, pending, power_list) { @@ -1237,7 +1393,7 @@ static void dapm_widget_update(struct snd_soc_dapm_context *dapm) w->name, ret); } - ret = snd_soc_update_bits(w->codec, update->reg, update->mask, + ret = soc_widget_update_bits_locked(w, update->reg, update->mask, update->val); if (ret < 0) pr_err("%s DAPM update failed: %d\n", w->name, ret); @@ -1357,6 +1513,7 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, switch (w->id) { case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: /* Supplies can't affect their outputs, only their inputs */ break; default: @@ -1415,18 +1572,16 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) struct snd_soc_dapm_context *d; LIST_HEAD(up_list); LIST_HEAD(down_list); - LIST_HEAD(async_domain); + ASYNC_DOMAIN_EXCLUSIVE(async_domain); enum snd_soc_bias_level bias; trace_snd_soc_dapm_start(card); list_for_each_entry(d, &card->dapm_list, list) { - if (d->n_widgets || d->codec == NULL) { - if (d->idle_bias_off) - d->target_bias_level = SND_SOC_BIAS_OFF; - else - d->target_bias_level = SND_SOC_BIAS_STANDBY; - } + if (d->idle_bias_off) + d->target_bias_level = SND_SOC_BIAS_OFF; + else + d->target_bias_level = SND_SOC_BIAS_STANDBY; } dapm_reset(card); @@ -1442,7 +1597,15 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } list_for_each_entry(w, &card->widgets, list) { - list_del_init(&w->dirty); + switch (w->id) { + case snd_soc_dapm_pre: + case snd_soc_dapm_post: + /* These widgets always need to be powered */ + break; + default: + list_del_init(&w->dirty); + break; + } if (w->power) { d = w->dapm; @@ -1459,6 +1622,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: case snd_soc_dapm_micbias: if (d->target_bias_level < SND_SOC_BIAS_STANDBY) d->target_bias_level = SND_SOC_BIAS_STANDBY; @@ -1471,32 +1635,6 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } - /* If there are no DAPM widgets then try to figure out power from the - * event type. - */ - if (!dapm->n_widgets) { - switch (event) { - case SND_SOC_DAPM_STREAM_START: - case SND_SOC_DAPM_STREAM_RESUME: - dapm->target_bias_level = SND_SOC_BIAS_ON; - break; - case SND_SOC_DAPM_STREAM_STOP: - if (dapm->codec && dapm->codec->active) - dapm->target_bias_level = SND_SOC_BIAS_ON; - else - dapm->target_bias_level = SND_SOC_BIAS_STANDBY; - break; - case SND_SOC_DAPM_STREAM_SUSPEND: - dapm->target_bias_level = SND_SOC_BIAS_STANDBY; - break; - case SND_SOC_DAPM_STREAM_NOP: - dapm->target_bias_level = dapm->bias_level; - break; - default: - break; - } - } - /* Force all contexts in the card to the same bias state if * they're not ground referenced. */ @@ -1560,9 +1698,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!buf) return -ENOMEM; - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", @@ -1709,7 +1847,7 @@ static inline void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm) #endif /* test and update the power status of a mux widget */ -int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, +static int soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e) { struct snd_soc_dapm_path *path; @@ -1746,12 +1884,26 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); } - return 0; + return found; +} + +int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e) +{ + struct snd_soc_card *card = widget->dapm->card; + int ret; + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + ret = soc_dapm_mux_update_power(widget, kcontrol, mux, e); + mutex_unlock(&card->dapm_mutex); + if (ret > 0) + soc_dpcm_runtime_update(widget); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_mux_update_power); /* test and update the power status of a mixer or switch widget */ -int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, +static int soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int connect) { struct snd_soc_dapm_path *path; @@ -1778,7 +1930,21 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); } - return 0; + return found; +} + +int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kcontrol, int connect) +{ + struct snd_soc_card *card = widget->dapm->card; + int ret; + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + ret = soc_dapm_mixer_update_power(widget, kcontrol, connect); + mutex_unlock(&card->dapm_mutex); + if (ret > 0) + soc_dpcm_runtime_update(widget); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power); @@ -1811,6 +1977,7 @@ static ssize_t dapm_widget_show(struct device *dev, case snd_soc_dapm_mixer_named_ctl: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: if (w->name) count += sprintf(buf + count, "%s: %s\n", w->name, w->power ? "On":"Off"); @@ -1939,6 +2106,8 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, */ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) { + int ret; + /* * Suppress early reports (eg, jacks syncing their state) to avoid * silly DAPM runs during card startup. @@ -1946,7 +2115,10 @@ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) if (!dapm->card || !dapm->card->instantiated) return 0; - return dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + ret = dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); + mutex_unlock(&dapm->card->dapm_mutex); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); @@ -2052,9 +2224,11 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_post: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: case snd_soc_dapm_dai: + case snd_soc_dapm_dai_link: list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &wsink->sources); list_add(&path->list_source, &wsource->sinks); @@ -2085,6 +2259,10 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, path->connect = 0; return 0; } + + dapm_mark_dirty(wsource, "Route added"); + dapm_mark_dirty(wsink, "Route added"); + return 0; err: @@ -2094,6 +2272,59 @@ err: return ret; } +static int snd_soc_dapm_del_route(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_route *route) +{ + struct snd_soc_dapm_path *path, *p; + const char *sink; + const char *source; + char prefixed_sink[80]; + char prefixed_source[80]; + + if (route->control) { + dev_err(dapm->dev, + "Removal of routes with controls not supported\n"); + return -EINVAL; + } + + if (dapm->codec && dapm->codec->name_prefix) { + snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s", + dapm->codec->name_prefix, route->sink); + sink = prefixed_sink; + snprintf(prefixed_source, sizeof(prefixed_source), "%s %s", + dapm->codec->name_prefix, route->source); + source = prefixed_source; + } else { + sink = route->sink; + source = route->source; + } + + path = NULL; + list_for_each_entry(p, &dapm->card->paths, list) { + if (strcmp(p->source->name, source) != 0) + continue; + if (strcmp(p->sink->name, sink) != 0) + continue; + path = p; + break; + } + + if (path) { + dapm_mark_dirty(path->source, "Route removed"); + dapm_mark_dirty(path->sink, "Route removed"); + + list_del(&path->list); + list_del(&path->list_sink); + list_del(&path->list_source); + kfree(path); + } else { + dev_warn(dapm->dev, "Route %s->%s does not exist\n", + source, sink); + } + + return 0; +} + /** * snd_soc_dapm_add_routes - Add routes between DAPM widgets * @dapm: DAPM context @@ -2110,22 +2341,48 @@ err: int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num) { - int i, ret; + int i, r, ret = 0; + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { - ret = snd_soc_dapm_add_route(dapm, route); - if (ret < 0) { + r = snd_soc_dapm_add_route(dapm, route); + if (r < 0) { dev_err(dapm->dev, "Failed to add route %s->%s\n", route->source, route->sink); - return ret; + ret = r; } route++; } + mutex_unlock(&dapm->card->dapm_mutex); - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes); +/** + * snd_soc_dapm_del_routes - Remove routes between DAPM widgets + * @dapm: DAPM context + * @route: audio routes + * @num: number of routes + * + * Removes routes from the DAPM context. + */ +int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_route *route, int num) +{ + int i, ret = 0; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); + for (i = 0; i < num; i++) { + snd_soc_dapm_del_route(dapm, route); + route++; + } + mutex_unlock(&dapm->card->dapm_mutex); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_del_routes); + static int snd_soc_dapm_weak_route(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route) { @@ -2193,12 +2450,14 @@ int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm, int i, err; int ret = 0; + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { err = snd_soc_dapm_weak_route(dapm, route); if (err) ret = err; route++; } + mutex_unlock(&dapm->card->dapm_mutex); return ret; } @@ -2217,6 +2476,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) struct snd_soc_dapm_widget *w; unsigned int val; + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); + list_for_each_entry(w, &dapm->card->widgets, list) { if (w->new) @@ -2226,8 +2487,10 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) w->kcontrols = kzalloc(w->num_kcontrols * sizeof(struct snd_kcontrol *), GFP_KERNEL); - if (!w->kcontrols) + if (!w->kcontrols) { + mutex_unlock(&dapm->card->dapm_mutex); return -ENOMEM; + } } switch(w->id) { @@ -2267,6 +2530,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) } dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); + mutex_unlock(&dapm->card->dapm_mutex); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); @@ -2289,23 +2553,20 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; unsigned int shift = mc->shift; - unsigned int rshift = mc->rshift; int max = mc->max; - unsigned int invert = mc->invert; unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + + if (snd_soc_volsw_is_stereo(mc)) + dev_warn(widget->dapm->dev, + "Control '%s' is stereo, which is not supported\n", + kcontrol->id.name); ucontrol->value.integer.value[0] = (snd_soc_read(widget->codec, reg) >> shift) & mask; - if (shift != rshift) - ucontrol->value.integer.value[1] = - (snd_soc_read(widget->codec, reg) >> rshift) & mask; - if (invert) { + if (invert) ucontrol->value.integer.value[0] = max - ucontrol->value.integer.value[0]; - if (shift != rshift) - ucontrol->value.integer.value[1] = - max - ucontrol->value.integer.value[1]; - } return 0; } @@ -2326,6 +2587,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; + struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; @@ -2338,21 +2600,20 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_update update; int wi; + if (snd_soc_volsw_is_stereo(mc)) + dev_warn(widget->dapm->dev, + "Control '%s' is stereo, which is not supported\n", + kcontrol->id.name); + val = (ucontrol->value.integer.value[0] & mask); + connect = !!val; if (invert) val = max - val; mask = mask << shift; val = val << shift; - if (val) - /* new connection */ - connect = invert ? 0 : 1; - else - /* old connection must be powered down */ - connect = invert ? 1 : 0; - - mutex_lock(&codec->mutex); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = snd_soc_test_bits(widget->codec, reg, mask, val); if (change) { @@ -2368,13 +2629,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; widget->dapm->update = &update; - snd_soc_dapm_mixer_update_power(widget, kcontrol, connect); + soc_dapm_mixer_update_power(widget, kcontrol, connect); widget->dapm->update = NULL; } } - mutex_unlock(&codec->mutex); + mutex_unlock(&card->dapm_mutex); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); @@ -2423,6 +2684,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; + struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val, mux, change; unsigned int mask, bitmask; @@ -2443,7 +2705,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mask |= (bitmask - 1) << e->shift_r; } - mutex_lock(&codec->mutex); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = snd_soc_test_bits(widget->codec, e->reg, mask, val); if (change) { @@ -2459,13 +2721,13 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; widget->dapm->update = &update; - snd_soc_dapm_mux_update_power(widget, kcontrol, mux, e); + soc_dapm_mux_update_power(widget, kcontrol, mux, e); widget->dapm->update = NULL; } } - mutex_unlock(&codec->mutex); + mutex_unlock(&card->dapm_mutex); return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); @@ -2502,6 +2764,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; + struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int change; @@ -2511,7 +2774,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, if (ucontrol->value.enumerated.item[0] >= e->max) return -EINVAL; - mutex_lock(&codec->mutex); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = widget->value != ucontrol->value.enumerated.item[0]; if (change) { @@ -2520,11 +2783,11 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, widget->value = ucontrol->value.enumerated.item[0]; - snd_soc_dapm_mux_update_power(widget, kcontrol, widget->value, e); + soc_dapm_mux_update_power(widget, kcontrol, widget->value, e); } } - mutex_unlock(&codec->mutex); + mutex_unlock(&card->dapm_mutex); return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt); @@ -2589,6 +2852,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; + struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val, mux, change; unsigned int mask; @@ -2607,7 +2871,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, mask |= e->mask << e->shift_r; } - mutex_lock(&codec->mutex); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = snd_soc_test_bits(widget->codec, e->reg, mask, val); if (change) { @@ -2623,13 +2887,13 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, update.val = val; widget->dapm->update = &update; - snd_soc_dapm_mux_update_power(widget, kcontrol, mux, e); + soc_dapm_mux_update_power(widget, kcontrol, mux, e); widget->dapm->update = NULL; } } - mutex_unlock(&codec->mutex); + mutex_unlock(&card->dapm_mutex); return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_value_enum_double); @@ -2666,12 +2930,12 @@ int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; - mutex_lock(&card->mutex); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); ucontrol->value.integer.value[0] = snd_soc_dapm_get_pin_status(&card->dapm, pin); - mutex_unlock(&card->mutex); + mutex_unlock(&card->dapm_mutex); return 0; } @@ -2689,17 +2953,16 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; - mutex_lock(&card->mutex); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); if (ucontrol->value.integer.value[0]) snd_soc_dapm_enable_pin(&card->dapm, pin); else snd_soc_dapm_disable_pin(&card->dapm, pin); - snd_soc_dapm_sync(&card->dapm); - - mutex_unlock(&card->mutex); + mutex_unlock(&card->dapm_mutex); + snd_soc_dapm_sync(&card->dapm); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); @@ -2717,14 +2980,27 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, switch (w->id) { case snd_soc_dapm_regulator_supply: - w->priv = devm_regulator_get(dapm->dev, w->name); - if (IS_ERR(w->priv)) { - ret = PTR_ERR(w->priv); + w->regulator = devm_regulator_get(dapm->dev, w->name); + if (IS_ERR(w->regulator)) { + ret = PTR_ERR(w->regulator); dev_err(dapm->dev, "Failed to request %s: %d\n", w->name, ret); return NULL; } break; + case snd_soc_dapm_clock_supply: +#ifdef CONFIG_CLKDEV_LOOKUP + w->clk = devm_clk_get(dapm->dev, w->name); + if (IS_ERR(w->clk)) { + ret = PTR_ERR(w->clk); + dev_err(dapm->dev, "Failed to request %s: %d\n", + w->name, ret); + return NULL; + } +#else + return NULL; +#endif + break; default: break; } @@ -2771,10 +3047,12 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_hp: case snd_soc_dapm_mic: case snd_soc_dapm_line: + case snd_soc_dapm_dai_link: w->power_check = dapm_generic_check_power; break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: w->power_check = dapm_supply_check_power; break; case snd_soc_dapm_dai: @@ -2816,21 +3094,177 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, { struct snd_soc_dapm_widget *w; int i; + int ret = 0; + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { w = snd_soc_dapm_new_control(dapm, widget); if (!w) { dev_err(dapm->dev, "ASoC: Failed to create DAPM control %s\n", widget->name); - return -ENOMEM; + ret = -ENOMEM; + break; } widget++; } - return 0; + mutex_unlock(&dapm->card->dapm_mutex); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls); +static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_dapm_path *source_p, *sink_p; + struct snd_soc_dai *source, *sink; + const struct snd_soc_pcm_stream *config = w->params; + struct snd_pcm_substream substream; + struct snd_pcm_hw_params *params = NULL; + u64 fmt; + int ret; + + BUG_ON(!config); + BUG_ON(list_empty(&w->sources) || list_empty(&w->sinks)); + + /* We only support a single source and sink, pick the first */ + source_p = list_first_entry(&w->sources, struct snd_soc_dapm_path, + list_sink); + sink_p = list_first_entry(&w->sinks, struct snd_soc_dapm_path, + list_source); + + BUG_ON(!source_p || !sink_p); + BUG_ON(!sink_p->source || !source_p->sink); + BUG_ON(!source_p->source || !sink_p->sink); + + source = source_p->source->priv; + sink = sink_p->sink->priv; + + /* Be a little careful as we don't want to overflow the mask array */ + if (config->formats) { + fmt = ffs(config->formats) - 1; + } else { + dev_warn(w->dapm->dev, "Invalid format %llx specified\n", + config->formats); + fmt = 0; + } + + /* Currently very limited parameter selection */ + params = kzalloc(sizeof(*params), GFP_KERNEL); + if (!params) { + ret = -ENOMEM; + goto out; + } + snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), fmt); + + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min = + config->rate_min; + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->max = + config->rate_max; + + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->min + = config->channels_min; + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max + = config->channels_max; + + memset(&substream, 0, sizeof(substream)); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (source->driver->ops && source->driver->ops->hw_params) { + substream.stream = SNDRV_PCM_STREAM_CAPTURE; + ret = source->driver->ops->hw_params(&substream, + params, source); + if (ret != 0) { + dev_err(source->dev, + "hw_params() failed: %d\n", ret); + goto out; + } + } + + if (sink->driver->ops && sink->driver->ops->hw_params) { + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; + ret = sink->driver->ops->hw_params(&substream, params, + sink); + if (ret != 0) { + dev_err(sink->dev, + "hw_params() failed: %d\n", ret); + goto out; + } + } + break; + + case SND_SOC_DAPM_POST_PMU: + ret = snd_soc_dai_digital_mute(sink, 0); + if (ret != 0 && ret != -ENOTSUPP) + dev_warn(sink->dev, "Failed to unmute: %d\n", ret); + ret = 0; + break; + + case SND_SOC_DAPM_PRE_PMD: + ret = snd_soc_dai_digital_mute(sink, 1); + if (ret != 0 && ret != -ENOTSUPP) + dev_warn(sink->dev, "Failed to mute: %d\n", ret); + ret = 0; + break; + + default: + BUG(); + return -EINVAL; + } + +out: + kfree(params); + return ret; +} + +int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_route routes[2]; + struct snd_soc_dapm_widget template; + struct snd_soc_dapm_widget *w; + size_t len; + char *link_name; + + len = strlen(source->name) + strlen(sink->name) + 2; + link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); + if (!link_name) + return -ENOMEM; + snprintf(link_name, len, "%s-%s", source->name, sink->name); + + memset(&template, 0, sizeof(template)); + template.reg = SND_SOC_NOPM; + template.id = snd_soc_dapm_dai_link; + template.name = link_name; + template.event = snd_soc_dai_link_event; + template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD; + + dev_dbg(card->dev, "adding %s widget\n", link_name); + + w = snd_soc_dapm_new_control(&card->dapm, &template); + if (!w) { + dev_err(card->dev, "Failed to create %s widget\n", + link_name); + return -ENOMEM; + } + + w->params = params; + + memset(&routes, 0, sizeof(routes)); + + routes[0].source = source->name; + routes[0].sink = link_name; + routes[1].source = link_name; + routes[1].sink = sink->name; + + return snd_soc_dapm_add_routes(&card->dapm, routes, + ARRAY_SIZE(routes)); +} + int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, struct snd_soc_dai *dai) { @@ -2934,37 +3368,61 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) return 0; } -static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, - int stream, struct snd_soc_dai *dai, - int event) +static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, + int event) { - struct snd_soc_dapm_widget *w; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - w = dai->playback_widget; - else - w = dai->capture_widget; + struct snd_soc_dapm_widget *w_cpu, *w_codec; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; - if (!w) - return; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + w_cpu = cpu_dai->playback_widget; + w_codec = codec_dai->playback_widget; + } else { + w_cpu = cpu_dai->capture_widget; + w_codec = codec_dai->capture_widget; + } - dapm_mark_dirty(w, "stream event"); + if (w_cpu) { - switch (event) { - case SND_SOC_DAPM_STREAM_START: - w->active = 1; - break; - case SND_SOC_DAPM_STREAM_STOP: - w->active = 0; - break; - case SND_SOC_DAPM_STREAM_SUSPEND: - case SND_SOC_DAPM_STREAM_RESUME: - case SND_SOC_DAPM_STREAM_PAUSE_PUSH: - case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: - break; + dapm_mark_dirty(w_cpu, "stream event"); + + switch (event) { + case SND_SOC_DAPM_STREAM_START: + w_cpu->active = 1; + break; + case SND_SOC_DAPM_STREAM_STOP: + w_cpu->active = 0; + break; + case SND_SOC_DAPM_STREAM_SUSPEND: + case SND_SOC_DAPM_STREAM_RESUME: + case SND_SOC_DAPM_STREAM_PAUSE_PUSH: + case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: + break; + } } - dapm_power_widgets(dapm, event); + if (w_codec) { + + dapm_mark_dirty(w_codec, "stream event"); + + switch (event) { + case SND_SOC_DAPM_STREAM_START: + w_codec->active = 1; + break; + case SND_SOC_DAPM_STREAM_STOP: + w_codec->active = 0; + break; + case SND_SOC_DAPM_STREAM_SUSPEND: + case SND_SOC_DAPM_STREAM_RESUME: + case SND_SOC_DAPM_STREAM_PAUSE_PUSH: + case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: + break; + } + } + + dapm_power_widgets(&rtd->card->dapm, event); } /** @@ -2978,15 +3436,14 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, * * Returns 0 for success else error. */ -int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, - struct snd_soc_dai *dai, int event) +void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, + int event) { - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = rtd->card; - mutex_lock(&codec->mutex); - soc_dapm_stream_event(&codec->dapm, stream, dai, event); - mutex_unlock(&codec->mutex); - return 0; + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + soc_dapm_stream_event(rtd, stream, event); + mutex_unlock(&card->dapm_mutex); } /** @@ -3210,10 +3667,13 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_free); static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) { + struct snd_soc_card *card = dapm->card; struct snd_soc_dapm_widget *w; LIST_HEAD(down_list); int powerdown = 0; + mutex_lock(&card->dapm_mutex); + list_for_each_entry(w, &dapm->card->widgets, list) { if (w->dapm != dapm) continue; @@ -3236,6 +3696,8 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY); } + + mutex_unlock(&card->dapm_mutex); } /* diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 475695234b3d..5df529eda251 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -30,6 +30,7 @@ struct dmaengine_pcm_runtime_data { struct dma_chan *dma_chan; + dma_cookie_t cookie; unsigned int pos; @@ -153,7 +154,7 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) desc->callback = dmaengine_pcm_dma_complete; desc->callback_param = substream; - dmaengine_submit(desc); + prtd->cookie = dmaengine_submit(desc); return 0; } @@ -200,6 +201,20 @@ int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd) EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_trigger); /** + * snd_dmaengine_pcm_pointer_no_residue - dmaengine based PCM pointer implementation + * @substream: PCM substream + * + * This function is deprecated and should not be used by new drivers, as its + * results may be unreliable. + */ +snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream) +{ + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + return bytes_to_frames(substream->runtime, prtd->pos); +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer_no_residue); + +/** * snd_dmaengine_pcm_pointer - dmaengine based PCM pointer implementation * @substream: PCM substream * @@ -209,7 +224,19 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_trigger); snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - return bytes_to_frames(substream->runtime, prtd->pos); + struct dma_tx_state state; + enum dma_status status; + unsigned int buf_size; + unsigned int pos = 0; + + status = dmaengine_tx_status(prtd->dma_chan, prtd->cookie, &state); + if (status == DMA_IN_PROGRESS || status == DMA_PAUSED) { + buf_size = snd_pcm_lib_buffer_bytes(substream); + if (state.residue > 0 && state.residue <= buf_size) + pos = buf_size - state.residue; + } + + return bytes_to_frames(substream->runtime, pos); } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer); @@ -243,7 +270,7 @@ static int dmaengine_pcm_request_channel(struct dmaengine_pcm_runtime_data *prtd * Note that this function will use private_data field of the substream's * runtime. So it is not availabe to your pcm driver implementation. If you need * to keep additional data attached to a substream use - * snd_dmaeinge_pcm_{set,get}_data. + * snd_dmaengine_pcm_{set,get}_data. */ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, dma_filter_fn filter_fn, void *filter_data) diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 4d8dc6a27d4d..29183ef2b93d 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -142,11 +142,16 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, case SND_SOC_REGMAP: /* Device has made its own regmap arrangements */ codec->using_regmap = true; - - ret = regmap_get_val_bytes(codec->control_data); - /* Errors are legitimate for non-integer byte multiples */ - if (ret > 0) - codec->val_bytes = ret; + if (!codec->control_data) + codec->control_data = dev_get_regmap(codec->dev, NULL); + + if (codec->control_data) { + ret = regmap_get_val_bytes(codec->control_data); + /* Errors are legitimate for non-integer byte + * multiples */ + if (ret > 0) + codec->val_bytes = ret; + } break; default: diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ee4353f843ea..7f8b3b7428bb 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -36,6 +36,7 @@ int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, struct snd_soc_jack *jack) { + mutex_init(&jack->mutex); jack->codec = codec; INIT_LIST_HEAD(&jack->pins); INIT_LIST_HEAD(&jack->jack_zones); @@ -75,7 +76,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) codec = jack->codec; dapm = &codec->dapm; - mutex_lock(&codec->mutex); + mutex_lock(&jack->mutex); oldstatus = jack->status; @@ -109,7 +110,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) snd_jack_report(jack->jack, jack->status); out: - mutex_unlock(&codec->mutex); + mutex_unlock(&jack->mutex); } EXPORT_SYMBOL_GPL(snd_soc_jack_report); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 0ad8dcacd2f3..ef22d0bd9e9e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -22,12 +22,38 @@ #include <linux/pm_runtime.h> #include <linux/slab.h> #include <linux/workqueue.h> +#include <linux/export.h> +#include <linux/debugfs.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-dpcm.h> #include <sound/initval.h> +#define DPCM_MAX_BE_USERS 8 + +/* DPCM stream event, send event to FE and all active BEs. */ +static int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, + int event) +{ + struct snd_soc_dpcm *dpcm; + + list_for_each_entry(dpcm, &fe->dpcm[dir].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + + dev_dbg(be->dev, "pm: BE %s event %d dir %d\n", + be->dai_link->name, event, dir); + + snd_soc_dapm_stream_event(be, dir, event); + } + + snd_soc_dapm_stream_event(fe, dir, event); + + return 0; +} + static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, struct snd_soc_dai *soc_dai) { @@ -156,6 +182,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } + /* Dynamic PCM DAI links compat checks use dynamic capabilities */ + if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) + goto dynamic; + /* Check that the codec and cpu DAIs are compatible */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { runtime->hw.rate_min = @@ -248,6 +278,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, runtime->hw.rate_max); +dynamic: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active++; codec_dai->playback_active++; @@ -308,7 +339,7 @@ static void close_delayed_work(struct work_struct *work) if (codec_dai->pop_wait == 1) { codec_dai->pop_wait = 0; snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - codec_dai, SND_SOC_DAPM_STREAM_STOP); + SND_SOC_DAPM_STREAM_STOP); } mutex_unlock(&rtd->pcm_mutex); @@ -373,7 +404,6 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - codec_dai, SND_SOC_DAPM_STREAM_STOP); } else { /* start delayed pop wq here for playback streams */ @@ -384,7 +414,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) } else { /* capture streams can be powered down now */ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, - codec_dai, SND_SOC_DAPM_STREAM_STOP); + SND_SOC_DAPM_STREAM_STOP); } mutex_unlock(&rtd->pcm_mutex); @@ -453,8 +483,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) cancel_delayed_work(&rtd->delayed_work); } - snd_soc_dapm_stream_event(rtd, substream->stream, codec_dai, - SND_SOC_DAPM_STREAM_START); + snd_soc_dapm_stream_event(rtd, substream->stream, + SND_SOC_DAPM_STREAM_START); snd_soc_dai_digital_mute(codec_dai, 0); @@ -602,6 +632,34 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } +static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + if (codec_dai->driver->ops->bespoke_trigger) { + ret = codec_dai->driver->ops->bespoke_trigger(substream, cmd, codec_dai); + if (ret < 0) + return ret; + } + + if (platform->driver->bespoke_trigger) { + ret = platform->driver->bespoke_trigger(substream, cmd); + if (ret < 0) + return ret; + } + + if (cpu_dai->driver->ops->bespoke_trigger) { + ret = cpu_dai->driver->ops->bespoke_trigger(substream, cmd, cpu_dai); + if (ret < 0) + return ret; + } + return 0; +} /* * soc level wrapper for pointer callback * If cpu_dai, codec_dai, platform driver has the delay callback, than @@ -634,74 +692,1668 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) return offset; } +/* connect a FE and BE */ +static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + struct snd_soc_dpcm *dpcm; + + /* only add new dpcms */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + if (dpcm->be == be && dpcm->fe == fe) + return 0; + } + + dpcm = kzalloc(sizeof(struct snd_soc_dpcm), GFP_KERNEL); + if (!dpcm) + return -ENOMEM; + + dpcm->be = be; + dpcm->fe = fe; + be->dpcm[stream].runtime = fe->dpcm[stream].runtime; + dpcm->state = SND_SOC_DPCM_LINK_STATE_NEW; + list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); + list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); + + dev_dbg(fe->dev, " connected new DPCM %s path %s %s %s\n", + stream ? "capture" : "playback", fe->dai_link->name, + stream ? "<-" : "->", be->dai_link->name); + +#ifdef CONFIG_DEBUG_FS + dpcm->debugfs_state = debugfs_create_u32(be->dai_link->name, 0644, + fe->debugfs_dpcm_root, &dpcm->state); +#endif + return 1; +} + +/* reparent a BE onto another FE */ +static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + struct snd_soc_dpcm *dpcm; + struct snd_pcm_substream *fe_substream, *be_substream; + + /* reparent if BE is connected to other FEs */ + if (!be->dpcm[stream].users) + return; + + be_substream = snd_soc_dpcm_get_substream(be, stream); + + list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) { + if (dpcm->fe == fe) + continue; + + dev_dbg(fe->dev, " reparent %s path %s %s %s\n", + stream ? "capture" : "playback", + dpcm->fe->dai_link->name, + stream ? "<-" : "->", dpcm->be->dai_link->name); + + fe_substream = snd_soc_dpcm_get_substream(dpcm->fe, stream); + be_substream->runtime = fe_substream->runtime; + break; + } +} + +/* disconnect a BE and FE */ +static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm, *d; + + list_for_each_entry_safe(dpcm, d, &fe->dpcm[stream].be_clients, list_be) { + dev_dbg(fe->dev, "BE %s disconnect check for %s\n", + stream ? "capture" : "playback", + dpcm->be->dai_link->name); + + if (dpcm->state != SND_SOC_DPCM_LINK_STATE_FREE) + continue; + + dev_dbg(fe->dev, " freed DSP %s path %s %s %s\n", + stream ? "capture" : "playback", fe->dai_link->name, + stream ? "<-" : "->", dpcm->be->dai_link->name); + + /* BEs still alive need new FE */ + dpcm_be_reparent(fe, dpcm->be, stream); + +#ifdef CONFIG_DEBUG_FS + debugfs_remove(dpcm->debugfs_state); +#endif + list_del(&dpcm->list_be); + list_del(&dpcm->list_fe); + kfree(dpcm); + } +} + +/* get BE for DAI widget and stream */ +static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, + struct snd_soc_dapm_widget *widget, int stream) +{ + struct snd_soc_pcm_runtime *be; + int i; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < card->num_links; i++) { + be = &card->rtd[i]; + + if (!be->dai_link->no_pcm) + continue; + + if (be->cpu_dai->playback_widget == widget || + be->codec_dai->playback_widget == widget) + return be; + } + } else { + + for (i = 0; i < card->num_links; i++) { + be = &card->rtd[i]; + + if (!be->dai_link->no_pcm) + continue; + + if (be->cpu_dai->capture_widget == widget || + be->codec_dai->capture_widget == widget) + return be; + } + } + + dev_err(card->dev, "can't get %s BE for %s\n", + stream ? "capture" : "playback", widget->name); + return NULL; +} + +static inline struct snd_soc_dapm_widget * + rtd_get_cpu_widget(struct snd_soc_pcm_runtime *rtd, int stream) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + return rtd->cpu_dai->playback_widget; + else + return rtd->cpu_dai->capture_widget; +} + +static inline struct snd_soc_dapm_widget * + rtd_get_codec_widget(struct snd_soc_pcm_runtime *rtd, int stream) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + return rtd->codec_dai->playback_widget; + else + return rtd->codec_dai->capture_widget; +} + +static int widget_in_list(struct snd_soc_dapm_widget_list *list, + struct snd_soc_dapm_widget *widget) +{ + int i; + + for (i = 0; i < list->num_widgets; i++) { + if (widget == list->widgets[i]) + return 1; + } + + return 0; +} + +static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list_) +{ + struct snd_soc_dai *cpu_dai = fe->cpu_dai; + struct snd_soc_dapm_widget_list *list; + int paths; + + list = kzalloc(sizeof(struct snd_soc_dapm_widget_list) + + sizeof(struct snd_soc_dapm_widget *), GFP_KERNEL); + if (list == NULL) + return -ENOMEM; + + /* get number of valid DAI paths and their widgets */ + paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, &list); + + dev_dbg(fe->dev, "found %d audio %s paths\n", paths, + stream ? "capture" : "playback"); + + *list_ = list; + return paths; +} + +static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) +{ + kfree(*list); +} + +static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, + struct snd_soc_dapm_widget_list **list_) +{ + struct snd_soc_dpcm *dpcm; + struct snd_soc_dapm_widget_list *list = *list_; + struct snd_soc_dapm_widget *widget; + int prune = 0; + + /* Destroy any old FE <--> BE connections */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + /* is there a valid CPU DAI widget for this BE */ + widget = rtd_get_cpu_widget(dpcm->be, stream); + + /* prune the BE if it's no longer in our active list */ + if (widget && widget_in_list(list, widget)) + continue; + + /* is there a valid CODEC DAI widget for this BE */ + widget = rtd_get_codec_widget(dpcm->be, stream); + + /* prune the BE if it's no longer in our active list */ + if (widget && widget_in_list(list, widget)) + continue; + + dev_dbg(fe->dev, "pruning %s BE %s for %s\n", + stream ? "capture" : "playback", + dpcm->be->dai_link->name, fe->dai_link->name); + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + dpcm->be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE; + prune++; + } + + dev_dbg(fe->dev, "found %d old BE paths for pruning\n", prune); + return prune; +} + +static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, + struct snd_soc_dapm_widget_list **list_) +{ + struct snd_soc_card *card = fe->card; + struct snd_soc_dapm_widget_list *list = *list_; + struct snd_soc_pcm_runtime *be; + int i, new = 0, err; + + /* Create any new FE <--> BE connections */ + for (i = 0; i < list->num_widgets; i++) { + + if (list->widgets[i]->id != snd_soc_dapm_dai) + continue; + + /* is there a valid BE rtd for this widget */ + be = dpcm_get_be(card, list->widgets[i], stream); + if (!be) { + dev_err(fe->dev, "no BE found for %s\n", + list->widgets[i]->name); + continue; + } + + /* make sure BE is a real BE */ + if (!be->dai_link->no_pcm) + continue; + + /* don't connect if FE is not running */ + if (!fe->dpcm[stream].runtime) + continue; + + /* newly connected FE and BE */ + err = dpcm_be_connect(fe, be, stream); + if (err < 0) { + dev_err(fe->dev, "can't connect %s\n", + list->widgets[i]->name); + break; + } else if (err == 0) /* already connected */ + continue; + + /* new */ + be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE; + new++; + } + + dev_dbg(fe->dev, "found %d new BE paths\n", new); + return new; +} + +/* + * Find the corresponding BE DAIs that source or sink audio to this + * FE substream. + */ +static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list, int new) +{ + if (new) + return dpcm_add_paths(fe, stream, list); + else + return dpcm_prune_paths(fe, stream, list); +} + +static void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm; + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + dpcm->be->dpcm[stream].runtime_update = + SND_SOC_DPCM_UPDATE_NO; +} + +static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, + int stream) +{ + struct snd_soc_dpcm *dpcm; + + /* disable any enabled and non active backends */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + if (be->dpcm[stream].users == 0) + dev_err(be->dev, "no users %s at close - state %d\n", + stream ? "capture" : "playback", + be->dpcm[stream].state); + + if (--be->dpcm[stream].users != 0) + continue; + + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN) + continue; + + soc_pcm_close(be_substream); + be_substream->runtime = NULL; + be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; + } +} + +static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm; + int err, count = 0; + + /* only startup BE DAIs that are either sinks or sources to this FE DAI */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + /* first time the dpcm is open ? */ + if (be->dpcm[stream].users == DPCM_MAX_BE_USERS) + dev_err(be->dev, "too many users %s at open %d\n", + stream ? "capture" : "playback", + be->dpcm[stream].state); + + if (be->dpcm[stream].users++ != 0) + continue; + + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_NEW) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_CLOSE)) + continue; + + dev_dbg(be->dev, "dpcm: open BE %s\n", be->dai_link->name); + + be_substream->runtime = be->dpcm[stream].runtime; + err = soc_pcm_open(be_substream); + if (err < 0) { + dev_err(be->dev, "BE open failed %d\n", err); + be->dpcm[stream].users--; + if (be->dpcm[stream].users < 0) + dev_err(be->dev, "no users %s at unwind %d\n", + stream ? "capture" : "playback", + be->dpcm[stream].state); + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; + goto unwind; + } + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; + count++; + } + + return count; + +unwind: + /* disable any enabled and non active backends */ + list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + if (be->dpcm[stream].users == 0) + dev_err(be->dev, "no users %s at close %d\n", + stream ? "capture" : "playback", + be->dpcm[stream].state); + + if (--be->dpcm[stream].users != 0) + continue; + + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN) + continue; + + soc_pcm_close(be_substream); + be_substream->runtime = NULL; + be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; + } + + return err; +} + +static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + runtime->hw.rate_min = cpu_dai_drv->playback.rate_min; + runtime->hw.rate_max = cpu_dai_drv->playback.rate_max; + runtime->hw.channels_min = cpu_dai_drv->playback.channels_min; + runtime->hw.channels_max = cpu_dai_drv->playback.channels_max; + runtime->hw.formats &= cpu_dai_drv->playback.formats; + runtime->hw.rates = cpu_dai_drv->playback.rates; + } else { + runtime->hw.rate_min = cpu_dai_drv->capture.rate_min; + runtime->hw.rate_max = cpu_dai_drv->capture.rate_max; + runtime->hw.channels_min = cpu_dai_drv->capture.channels_min; + runtime->hw.channels_max = cpu_dai_drv->capture.channels_max; + runtime->hw.formats &= cpu_dai_drv->capture.formats; + runtime->hw.rates = cpu_dai_drv->capture.rates; + } +} + +static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) +{ + struct snd_soc_pcm_runtime *fe = fe_substream->private_data; + struct snd_pcm_runtime *runtime = fe_substream->runtime; + int stream = fe_substream->stream, ret = 0; + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + ret = dpcm_be_dai_startup(fe, fe_substream->stream); + if (ret < 0) { + dev_err(fe->dev,"dpcm: failed to start some BEs %d\n", ret); + goto be_err; + } + + dev_dbg(fe->dev, "dpcm: open FE %s\n", fe->dai_link->name); + + /* start the DAI frontend */ + ret = soc_pcm_open(fe_substream); + if (ret < 0) { + dev_err(fe->dev,"dpcm: failed to start FE %d\n", ret); + goto unwind; + } + + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; + + dpcm_set_fe_runtime(fe_substream); + snd_pcm_limit_hw_rates(runtime); + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + return 0; + +unwind: + dpcm_be_dai_startup_unwind(fe, fe_substream->stream); +be_err: + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + return ret; +} + +static int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm; + + /* only shutdown BEs that are either sinks or sources to this FE DAI */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + if (be->dpcm[stream].users == 0) + dev_err(be->dev, "no users %s at close - state %d\n", + stream ? "capture" : "playback", + be->dpcm[stream].state); + + if (--be->dpcm[stream].users != 0) + continue; + + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN)) + continue; + + dev_dbg(be->dev, "dpcm: close BE %s\n", + dpcm->fe->dai_link->name); + + soc_pcm_close(be_substream); + be_substream->runtime = NULL; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; + } + return 0; +} + +static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + int stream = substream->stream; + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + /* shutdown the BEs */ + dpcm_be_dai_shutdown(fe, substream->stream); + + dev_dbg(fe->dev, "dpcm: close FE %s\n", fe->dai_link->name); + + /* now shutdown the frontend */ + soc_pcm_close(substream); + + /* run the stream event for each BE */ + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); + + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + return 0; +} + +static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm; + + /* only hw_params backends that are either sinks or sources + * to this frontend DAI */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + /* only free hw when no longer used - check all FEs */ + if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + continue; + + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + continue; + + dev_dbg(be->dev, "dpcm: hw_free BE %s\n", + dpcm->fe->dai_link->name); + + soc_pcm_hw_free(be_substream); + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE; + } + + return 0; +} + +static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + int err, stream = substream->stream; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + dev_dbg(fe->dev, "dpcm: hw_free FE %s\n", fe->dai_link->name); + + /* call hw_free on the frontend */ + err = soc_pcm_hw_free(substream); + if (err < 0) + dev_err(fe->dev,"dpcm: hw_free FE %s failed\n", + fe->dai_link->name); + + /* only hw_params backends that are either sinks or sources + * to this frontend DAI */ + err = dpcm_be_dai_hw_free(fe, stream); + + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE; + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + + mutex_unlock(&fe->card->mutex); + return 0; +} + +static int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm; + int ret; + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + /* only allow hw_params() if no connected FEs are running */ + if (!snd_soc_dpcm_can_be_params(fe, be, stream)) + continue; + + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE)) + continue; + + dev_dbg(be->dev, "dpcm: hw_params BE %s\n", + dpcm->fe->dai_link->name); + + /* copy params for each dpcm */ + memcpy(&dpcm->hw_params, &fe->dpcm[stream].hw_params, + sizeof(struct snd_pcm_hw_params)); + + /* perform any hw_params fixups */ + if (be->dai_link->be_hw_params_fixup) { + ret = be->dai_link->be_hw_params_fixup(be, + &dpcm->hw_params); + if (ret < 0) { + dev_err(be->dev, + "dpcm: hw_params BE fixup failed %d\n", + ret); + goto unwind; + } + } + + ret = soc_pcm_hw_params(be_substream, &dpcm->hw_params); + if (ret < 0) { + dev_err(dpcm->be->dev, + "dpcm: hw_params BE failed %d\n", ret); + goto unwind; + } + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_PARAMS; + } + return 0; + +unwind: + /* disable any enabled and non active backends */ + list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + /* only allow hw_free() if no connected FEs are running */ + if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + continue; + + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + continue; + + soc_pcm_hw_free(be_substream); + } + + return ret; +} + +static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + int ret, stream = substream->stream; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + memcpy(&fe->dpcm[substream->stream].hw_params, params, + sizeof(struct snd_pcm_hw_params)); + ret = dpcm_be_dai_hw_params(fe, substream->stream); + if (ret < 0) { + dev_err(fe->dev,"dpcm: hw_params BE failed %d\n", ret); + goto out; + } + + dev_dbg(fe->dev, "dpcm: hw_params FE %s rate %d chan %x fmt %d\n", + fe->dai_link->name, params_rate(params), + params_channels(params), params_format(params)); + + /* call hw_params on the frontend */ + ret = soc_pcm_hw_params(substream, params); + if (ret < 0) { + dev_err(fe->dev,"dpcm: hw_params FE failed %d\n", ret); + dpcm_be_dai_hw_free(fe, stream); + } else + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_PARAMS; + +out: + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + mutex_unlock(&fe->card->mutex); + return ret; +} + +static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, + struct snd_pcm_substream *substream, int cmd) +{ + int ret; + + dev_dbg(dpcm->be->dev, "dpcm: trigger BE %s cmd %d\n", + dpcm->fe->dai_link->name, cmd); + + ret = soc_pcm_trigger(substream, cmd); + if (ret < 0) + dev_err(dpcm->be->dev,"dpcm: trigger BE failed %d\n", ret); + + return ret; +} + +static int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, + int cmd) +{ + struct snd_soc_dpcm *dpcm; + int ret = 0; + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + continue; + + ret = dpcm_do_trigger(dpcm, be_substream, cmd); + if (ret) + return ret; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; + break; + case SNDRV_PCM_TRIGGER_RESUME: + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND)) + continue; + + ret = dpcm_do_trigger(dpcm, be_substream, cmd); + if (ret) + return ret; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) + continue; + + ret = dpcm_do_trigger(dpcm, be_substream, cmd); + if (ret) + return ret; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; + break; + case SNDRV_PCM_TRIGGER_STOP: + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + continue; + + if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + continue; + + ret = dpcm_do_trigger(dpcm, be_substream, cmd); + if (ret) + return ret; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) + continue; + + if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + continue; + + ret = dpcm_do_trigger(dpcm, be_substream, cmd); + if (ret) + return ret; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + continue; + + if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + continue; + + ret = dpcm_do_trigger(dpcm, be_substream, cmd); + if (ret) + return ret; + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; + break; + } + } + + return ret; +} +EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger); + +static int dpcm_fe_dai_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + int stream = substream->stream, ret; + enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + switch (trigger) { + case SND_SOC_DPCM_TRIGGER_PRE: + /* call trigger on the frontend before the backend. */ + + dev_dbg(fe->dev, "dpcm: pre trigger FE %s cmd %d\n", + fe->dai_link->name, cmd); + + ret = soc_pcm_trigger(substream, cmd); + if (ret < 0) { + dev_err(fe->dev,"dpcm: trigger FE failed %d\n", ret); + goto out; + } + + ret = dpcm_be_dai_trigger(fe, substream->stream, cmd); + break; + case SND_SOC_DPCM_TRIGGER_POST: + /* call trigger on the frontend after the backend. */ + + ret = dpcm_be_dai_trigger(fe, substream->stream, cmd); + if (ret < 0) { + dev_err(fe->dev,"dpcm: trigger FE failed %d\n", ret); + goto out; + } + + dev_dbg(fe->dev, "dpcm: post trigger FE %s cmd %d\n", + fe->dai_link->name, cmd); + + ret = soc_pcm_trigger(substream, cmd); + break; + case SND_SOC_DPCM_TRIGGER_BESPOKE: + /* bespoke trigger() - handles both FE and BEs */ + + dev_dbg(fe->dev, "dpcm: bespoke trigger FE %s cmd %d\n", + fe->dai_link->name, cmd); + + ret = soc_pcm_bespoke_trigger(substream, cmd); + if (ret < 0) { + dev_err(fe->dev,"dpcm: trigger FE failed %d\n", ret); + goto out; + } + break; + default: + dev_err(fe->dev, "dpcm: invalid trigger cmd %d for %s\n", cmd, + fe->dai_link->name); + ret = -EINVAL; + goto out; + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_START; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; + break; + } + +out: + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + return ret; +} + +static int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_soc_dpcm *dpcm; + int ret = 0; + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *be_substream = + snd_soc_dpcm_get_substream(be, stream); + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + continue; + + dev_dbg(be->dev, "dpcm: prepare BE %s\n", + dpcm->fe->dai_link->name); + + ret = soc_pcm_prepare(be_substream); + if (ret < 0) { + dev_err(be->dev, "dpcm: backend prepare failed %d\n", + ret); + break; + } + + be->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; + } + return ret; +} + +static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + int stream = substream->stream, ret = 0; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + + dev_dbg(fe->dev, "dpcm: prepare FE %s\n", fe->dai_link->name); + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + /* there is no point preparing this FE if there are no BEs */ + if (list_empty(&fe->dpcm[stream].be_clients)) { + dev_err(fe->dev, "dpcm: no backend DAIs enabled for %s\n", + fe->dai_link->name); + ret = -EINVAL; + goto out; + } + + ret = dpcm_be_dai_prepare(fe, substream->stream); + if (ret < 0) + goto out; + + /* call prepare on the frontend */ + ret = soc_pcm_prepare(substream); + if (ret < 0) { + dev_err(fe->dev,"dpcm: prepare FE %s failed\n", + fe->dai_link->name); + goto out; + } + + /* run the stream event for each BE */ + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START); + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; + +out: + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + mutex_unlock(&fe->card->mutex); + + return ret; +} + +static int soc_pcm_ioctl(struct snd_pcm_substream *substream, + unsigned int cmd, void *arg) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + + if (platform->driver->ops->ioctl) + return platform->driver->ops->ioctl(substream, cmd, arg); + return snd_pcm_lib_ioctl(substream, cmd, arg); +} + +static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_pcm_substream *substream = + snd_soc_dpcm_get_substream(fe, stream); + enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; + int err; + + dev_dbg(fe->dev, "runtime %s close on FE %s\n", + stream ? "capture" : "playback", fe->dai_link->name); + + if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { + /* call bespoke trigger - FE takes care of all BE triggers */ + dev_dbg(fe->dev, "dpcm: bespoke trigger FE %s cmd stop\n", + fe->dai_link->name); + + err = soc_pcm_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_STOP); + if (err < 0) + dev_err(fe->dev,"dpcm: trigger FE failed %d\n", err); + } else { + dev_dbg(fe->dev, "dpcm: trigger FE %s cmd stop\n", + fe->dai_link->name); + + err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); + if (err < 0) + dev_err(fe->dev,"dpcm: trigger FE failed %d\n", err); + } + + err = dpcm_be_dai_hw_free(fe, stream); + if (err < 0) + dev_err(fe->dev,"dpcm: hw_free FE failed %d\n", err); + + err = dpcm_be_dai_shutdown(fe, stream); + if (err < 0) + dev_err(fe->dev,"dpcm: shutdown FE failed %d\n", err); + + /* run the stream event for each BE */ + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_NOP); + + return 0; +} + +static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) +{ + struct snd_pcm_substream *substream = + snd_soc_dpcm_get_substream(fe, stream); + struct snd_soc_dpcm *dpcm; + enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; + int ret; + + dev_dbg(fe->dev, "runtime %s open on FE %s\n", + stream ? "capture" : "playback", fe->dai_link->name); + + /* Only start the BE if the FE is ready */ + if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_HW_FREE || + fe->dpcm[stream].state == SND_SOC_DPCM_STATE_CLOSE) + return -EINVAL; + + /* startup must always be called for new BEs */ + ret = dpcm_be_dai_startup(fe, stream); + if (ret < 0) { + goto disconnect; + return ret; + } + + /* keep going if FE state is > open */ + if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_OPEN) + return 0; + + ret = dpcm_be_dai_hw_params(fe, stream); + if (ret < 0) { + goto close; + return ret; + } + + /* keep going if FE state is > hw_params */ + if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_HW_PARAMS) + return 0; + + + ret = dpcm_be_dai_prepare(fe, stream); + if (ret < 0) { + goto hw_free; + return ret; + } + + /* run the stream event for each BE */ + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_NOP); + + /* keep going if FE state is > prepare */ + if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_PREPARE || + fe->dpcm[stream].state == SND_SOC_DPCM_STATE_STOP) + return 0; + + if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { + /* call trigger on the frontend - FE takes care of all BE triggers */ + dev_dbg(fe->dev, "dpcm: bespoke trigger FE %s cmd start\n", + fe->dai_link->name); + + ret = soc_pcm_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_START); + if (ret < 0) { + dev_err(fe->dev,"dpcm: bespoke trigger FE failed %d\n", ret); + goto hw_free; + } + } else { + dev_dbg(fe->dev, "dpcm: trigger FE %s cmd start\n", + fe->dai_link->name); + + ret = dpcm_be_dai_trigger(fe, stream, + SNDRV_PCM_TRIGGER_START); + if (ret < 0) { + dev_err(fe->dev,"dpcm: trigger FE failed %d\n", ret); + goto hw_free; + } + } + + return 0; + +hw_free: + dpcm_be_dai_hw_free(fe, stream); +close: + dpcm_be_dai_shutdown(fe, stream); +disconnect: + /* disconnect any non started BEs */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + } + + return ret; +} + +static int dpcm_run_new_update(struct snd_soc_pcm_runtime *fe, int stream) +{ + int ret; + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE; + ret = dpcm_run_update_startup(fe, stream); + if (ret < 0) + dev_err(fe->dev, "failed to startup some BEs\n"); + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + + return ret; +} + +static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) +{ + int ret; + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE; + ret = dpcm_run_update_shutdown(fe, stream); + if (ret < 0) + dev_err(fe->dev, "failed to shutdown some BEs\n"); + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + + return ret; +} + +/* Called by DAPM mixer/mux changes to update audio routing between PCMs and + * any DAI links. + */ +int soc_dpcm_runtime_update(struct snd_soc_dapm_widget *widget) +{ + struct snd_soc_card *card; + int i, old, new, paths; + + if (widget->codec) + card = widget->codec->card; + else if (widget->platform) + card = widget->platform->card; + else + return -EINVAL; + + mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dapm_widget_list *list; + struct snd_soc_pcm_runtime *fe = &card->rtd[i]; + + /* make sure link is FE */ + if (!fe->dai_link->dynamic) + continue; + + /* only check active links */ + if (!fe->cpu_dai->active) + continue; + + /* DAPM sync will call this to update DSP paths */ + dev_dbg(fe->dev, "DPCM runtime update for FE %s\n", + fe->dai_link->name); + + /* skip if FE doesn't have playback capability */ + if (!fe->cpu_dai->driver->playback.channels_min) + goto capture; + + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); + if (paths < 0) { + dev_warn(fe->dev, "%s no valid %s path\n", + fe->dai_link->name, "playback"); + mutex_unlock(&card->mutex); + return paths; + } + + /* update any new playback paths */ + new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 1); + if (new) { + dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); + } + + /* update any old playback paths */ + old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 0); + if (old) { + dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); + } + +capture: + /* skip if FE doesn't have capture capability */ + if (!fe->cpu_dai->driver->capture.channels_min) + continue; + + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); + if (paths < 0) { + dev_warn(fe->dev, "%s no valid %s path\n", + fe->dai_link->name, "capture"); + mutex_unlock(&card->mutex); + return paths; + } + + /* update any new capture paths */ + new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 1); + if (new) { + dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); + } + + /* update any old capture paths */ + old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 0); + if (old) { + dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); + } + + dpcm_path_put(&list); + } + + mutex_unlock(&card->mutex); + return 0; +} +int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) +{ + struct snd_soc_dpcm *dpcm; + struct list_head *clients = + &fe->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients; + + list_for_each_entry(dpcm, clients, list_be) { + + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_soc_dai *dai = be->codec_dai; + struct snd_soc_dai_driver *drv = dai->driver; + + if (be->dai_link->ignore_suspend) + continue; + + dev_dbg(be->dev, "BE digital mute %s\n", be->dai_link->name); + + if (drv->ops->digital_mute && dai->playback_active) + drv->ops->digital_mute(dai, mute); + } + + return 0; +} + +static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) +{ + struct snd_soc_pcm_runtime *fe = fe_substream->private_data; + struct snd_soc_dpcm *dpcm; + struct snd_soc_dapm_widget_list *list; + int ret; + int stream = fe_substream->stream; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + fe->dpcm[stream].runtime = fe_substream->runtime; + + if (dpcm_path_get(fe, stream, &list) <= 0) { + dev_dbg(fe->dev, "asoc: %s no valid %s route\n", + fe->dai_link->name, stream ? "capture" : "playback"); + } + + /* calculate valid and active FE <-> BE dpcms */ + dpcm_process_paths(fe, stream, &list, 1); + + ret = dpcm_fe_dai_startup(fe_substream); + if (ret < 0) { + /* clean up all links */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + + dpcm_be_disconnect(fe, stream); + fe->dpcm[stream].runtime = NULL; + } + + dpcm_clear_pending_state(fe, stream); + dpcm_path_put(&list); + mutex_unlock(&fe->card->mutex); + return ret; +} + +static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) +{ + struct snd_soc_pcm_runtime *fe = fe_substream->private_data; + struct snd_soc_dpcm *dpcm; + int stream = fe_substream->stream, ret; + + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + ret = dpcm_fe_dai_shutdown(fe_substream); + + /* mark FE's links ready to prune */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + + dpcm_be_disconnect(fe, stream); + + fe->dpcm[stream].runtime = NULL; + mutex_unlock(&fe->card->mutex); + return ret; +} + /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { - struct snd_soc_codec *codec = rtd->codec; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_pcm_ops *soc_pcm_ops = &rtd->ops; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; - soc_pcm_ops->open = soc_pcm_open; - soc_pcm_ops->close = soc_pcm_close; - soc_pcm_ops->hw_params = soc_pcm_hw_params; - soc_pcm_ops->hw_free = soc_pcm_hw_free; - soc_pcm_ops->prepare = soc_pcm_prepare; - soc_pcm_ops->trigger = soc_pcm_trigger; - soc_pcm_ops->pointer = soc_pcm_pointer; - - /* check client and interface hw capabilities */ - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); - - if (codec_dai->driver->playback.channels_min) - playback = 1; - if (codec_dai->driver->capture.channels_min) - capture = 1; - - dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name); - ret = snd_pcm_new(rtd->card->snd_card, new_name, - num, playback, capture, &pcm); + if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { + if (cpu_dai->driver->playback.channels_min) + playback = 1; + if (cpu_dai->driver->capture.channels_min) + capture = 1; + } else { + if (codec_dai->driver->playback.channels_min) + playback = 1; + if (codec_dai->driver->capture.channels_min) + capture = 1; + } + + /* create the PCM */ + if (rtd->dai_link->no_pcm) { + snprintf(new_name, sizeof(new_name), "(%s)", + rtd->dai_link->stream_name); + + ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, + playback, capture, &pcm); + } else { + if (rtd->dai_link->dynamic) + snprintf(new_name, sizeof(new_name), "%s (*)", + rtd->dai_link->stream_name); + else + snprintf(new_name, sizeof(new_name), "%s %s-%d", + rtd->dai_link->stream_name, codec_dai->name, num); + + ret = snd_pcm_new(rtd->card->snd_card, new_name, num, playback, + capture, &pcm); + } if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); + dev_err(rtd->card->dev, "can't create pcm for %s\n", + rtd->dai_link->name); return ret; } + dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num, new_name); /* DAPM dai link stream work */ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); rtd->pcm = pcm; pcm->private_data = rtd; + + if (rtd->dai_link->no_pcm) { + if (playback) + pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; + if (capture) + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; + goto out; + } + + /* ASoC PCM operations */ + if (rtd->dai_link->dynamic) { + rtd->ops.open = dpcm_fe_dai_open; + rtd->ops.hw_params = dpcm_fe_dai_hw_params; + rtd->ops.prepare = dpcm_fe_dai_prepare; + rtd->ops.trigger = dpcm_fe_dai_trigger; + rtd->ops.hw_free = dpcm_fe_dai_hw_free; + rtd->ops.close = dpcm_fe_dai_close; + rtd->ops.pointer = soc_pcm_pointer; + rtd->ops.ioctl = soc_pcm_ioctl; + } else { + rtd->ops.open = soc_pcm_open; + rtd->ops.hw_params = soc_pcm_hw_params; + rtd->ops.prepare = soc_pcm_prepare; + rtd->ops.trigger = soc_pcm_trigger; + rtd->ops.hw_free = soc_pcm_hw_free; + rtd->ops.close = soc_pcm_close; + rtd->ops.pointer = soc_pcm_pointer; + rtd->ops.ioctl = soc_pcm_ioctl; + } + if (platform->driver->ops) { - soc_pcm_ops->mmap = platform->driver->ops->mmap; - soc_pcm_ops->pointer = platform->driver->ops->pointer; - soc_pcm_ops->ioctl = platform->driver->ops->ioctl; - soc_pcm_ops->copy = platform->driver->ops->copy; - soc_pcm_ops->silence = platform->driver->ops->silence; - soc_pcm_ops->ack = platform->driver->ops->ack; - soc_pcm_ops->page = platform->driver->ops->page; + rtd->ops.ack = platform->driver->ops->ack; + rtd->ops.copy = platform->driver->ops->copy; + rtd->ops.silence = platform->driver->ops->silence; + rtd->ops.page = platform->driver->ops->page; + rtd->ops.mmap = platform->driver->ops->mmap; } if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &rtd->ops); if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &rtd->ops); if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); if (ret < 0) { - pr_err("asoc: platform pcm constructor failed\n"); + dev_err(platform->dev, "pcm constructor failed\n"); return ret; } } pcm->private_free = platform->driver->pcm_free; - printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, +out: + dev_info(rtd->card->dev, " %s <-> %s mapping ok\n", codec_dai->name, cpu_dai->name); return ret; } + +/* is the current PCM operation for this FE ? */ +int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream) +{ + if (fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_FE) + return 1; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_fe_can_update); + +/* is the current PCM operation for this BE ? */ +int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + if ((fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_FE) || + ((fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_BE) && + be->dpcm[stream].runtime_update)) + return 1; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_can_update); + +/* get the substream for this BE */ +struct snd_pcm_substream * + snd_soc_dpcm_get_substream(struct snd_soc_pcm_runtime *be, int stream) +{ + return be->pcm->streams[stream].substream; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_get_substream); + +/* get the BE runtime state */ +enum snd_soc_dpcm_state + snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream) +{ + return be->dpcm[stream].state; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_get_state); + +/* set the BE runtime state */ +void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, + int stream, enum snd_soc_dpcm_state state) +{ + be->dpcm[stream].state = state; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_set_state); + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + struct snd_soc_dpcm *dpcm; + int state; + + list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) { + + if (dpcm->fe == fe) + continue; + + state = dpcm->fe->dpcm[stream].state; + if (state == SND_SOC_DPCM_STATE_START || + state == SND_SOC_DPCM_STATE_PAUSED || + state == SND_SOC_DPCM_STATE_SUSPEND) + return 0; + } + + /* it's safe to free/stop this BE DAI */ + return 1; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); + +/* + * We can only change hw params a BE DAI if any of it's FE are not prepared, + * running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + struct snd_soc_dpcm *dpcm; + int state; + + list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) { + + if (dpcm->fe == fe) + continue; + + state = dpcm->fe->dpcm[stream].state; + if (state == SND_SOC_DPCM_STATE_START || + state == SND_SOC_DPCM_STATE_PAUSED || + state == SND_SOC_DPCM_STATE_SUSPEND || + state == SND_SOC_DPCM_STATE_PREPARE) + return 0; + } + + /* it's safe to change hw_params */ + return 1; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); + +int snd_soc_platform_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_platform *platform) +{ + if (platform->driver->ops->trigger) + return platform->driver->ops->trigger(substream, cmd); + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_platform_trigger); + +#ifdef CONFIG_DEBUG_FS +static char *dpcm_state_string(enum snd_soc_dpcm_state state) +{ + switch (state) { + case SND_SOC_DPCM_STATE_NEW: + return "new"; + case SND_SOC_DPCM_STATE_OPEN: + return "open"; + case SND_SOC_DPCM_STATE_HW_PARAMS: + return "hw_params"; + case SND_SOC_DPCM_STATE_PREPARE: + return "prepare"; + case SND_SOC_DPCM_STATE_START: + return "start"; + case SND_SOC_DPCM_STATE_STOP: + return "stop"; + case SND_SOC_DPCM_STATE_SUSPEND: + return "suspend"; + case SND_SOC_DPCM_STATE_PAUSED: + return "paused"; + case SND_SOC_DPCM_STATE_HW_FREE: + return "hw_free"; + case SND_SOC_DPCM_STATE_CLOSE: + return "close"; + } + + return "unknown"; +} + +static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, + int stream, char *buf, size_t size) +{ + struct snd_pcm_hw_params *params = &fe->dpcm[stream].hw_params; + struct snd_soc_dpcm *dpcm; + ssize_t offset = 0; + + /* FE state */ + offset += snprintf(buf + offset, size - offset, + "[%s - %s]\n", fe->dai_link->name, + stream ? "Capture" : "Playback"); + + offset += snprintf(buf + offset, size - offset, "State: %s\n", + dpcm_state_string(fe->dpcm[stream].state)); + + if ((fe->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && + (fe->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) + offset += snprintf(buf + offset, size - offset, + "Hardware Params: " + "Format = %s, Channels = %d, Rate = %d\n", + snd_pcm_format_name(params_format(params)), + params_channels(params), + params_rate(params)); + + /* BEs state */ + offset += snprintf(buf + offset, size - offset, "Backends:\n"); + + if (list_empty(&fe->dpcm[stream].be_clients)) { + offset += snprintf(buf + offset, size - offset, + " No active DSP links\n"); + goto out; + } + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + params = &dpcm->hw_params; + + offset += snprintf(buf + offset, size - offset, + "- %s\n", be->dai_link->name); + + offset += snprintf(buf + offset, size - offset, + " State: %s\n", + dpcm_state_string(be->dpcm[stream].state)); + + if ((be->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && + (be->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) + offset += snprintf(buf + offset, size - offset, + " Hardware Params: " + "Format = %s, Channels = %d, Rate = %d\n", + snd_pcm_format_name(params_format(params)), + params_channels(params), + params_rate(params)); + } + +out: + return offset; +} + +static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + struct snd_soc_pcm_runtime *fe = file->private_data; + ssize_t out_count = PAGE_SIZE, offset = 0, ret = 0; + char *buf; + + buf = kmalloc(out_count, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + if (fe->cpu_dai->driver->playback.channels_min) + offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_PLAYBACK, + buf + offset, out_count - offset); + + if (fe->cpu_dai->driver->capture.channels_min) + offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_CAPTURE, + buf + offset, out_count - offset); + + ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset); + + kfree(buf); + return ret; +} + +static const struct file_operations dpcm_state_fops = { + .open = simple_open, + .read = dpcm_state_read_file, + .llseek = default_llseek, +}; + +int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd) +{ + if (!rtd->dai_link) + return 0; + + rtd->debugfs_dpcm_root = debugfs_create_dir(rtd->dai_link->name, + rtd->card->debugfs_card_root); + if (!rtd->debugfs_dpcm_root) { + dev_dbg(rtd->dev, + "ASoC: Failed to create dpcm debugfs directory %s\n", + rtd->dai_link->name); + return -EINVAL; + } + + rtd->debugfs_dpcm_state = debugfs_create_file("state", 0444, + rtd->debugfs_dpcm_root, + rtd, &dpcm_state_fops); + + return 0; +} +#endif diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c new file mode 100644 index 000000000000..c7c4b20395bb --- /dev/null +++ b/sound/soc/spear/spdif_in.c @@ -0,0 +1,297 @@ +/* + * ALSA SoC SPDIF In Audio Layer for spear processors + * + * Copyright (C) 2012 ST Microelectronics + * Vipin Kumar <vipin.kumar@st.com> + * + * This file is licensed under the terms of the GNU General Public + * License version 2. This program is licensed "as is" without any + * warranty of any kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/ioport.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/spear_dma.h> +#include <sound/spear_spdif.h> +#include "spdif_in_regs.h" + +struct spdif_in_params { + u32 format; +}; + +struct spdif_in_dev { + struct clk *clk; + struct spear_dma_data dma_params; + struct spdif_in_params saved_params; + void *io_base; + struct device *dev; + void (*reset_perip)(void); + int irq; +}; + +static void spdif_in_configure(struct spdif_in_dev *host) +{ + u32 ctrl = SPDIF_IN_PRTYEN | SPDIF_IN_STATEN | SPDIF_IN_USREN | + SPDIF_IN_VALEN | SPDIF_IN_BLKEN; + ctrl |= SPDIF_MODE_16BIT | SPDIF_FIFO_THRES_16; + + writel(ctrl, host->io_base + SPDIF_IN_CTRL); + writel(0xF, host->io_base + SPDIF_IN_IRQ_MASK); +} + +static int spdif_in_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct spdif_in_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)&host->dma_params); + return 0; +} + +static void spdif_in_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return; + + writel(0x0, host->io_base + SPDIF_IN_IRQ_MASK); + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static void spdif_in_format(struct spdif_in_dev *host, u32 format) +{ + u32 ctrl = readl(host->io_base + SPDIF_IN_CTRL); + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + ctrl |= SPDIF_XTRACT_16BIT; + break; + + case SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE: + ctrl &= ~SPDIF_XTRACT_16BIT; + break; + } + + writel(ctrl, host->io_base + SPDIF_IN_CTRL); +} + +static int spdif_in_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai); + u32 format; + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + format = params_format(params); + host->saved_params.format = format; + + return 0; +} + +static int spdif_in_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai); + u32 ctrl; + int ret = 0; + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + clk_enable(host->clk); + spdif_in_configure(host); + spdif_in_format(host, host->saved_params.format); + + ctrl = readl(host->io_base + SPDIF_IN_CTRL); + ctrl |= SPDIF_IN_SAMPLE | SPDIF_IN_ENB; + writel(ctrl, host->io_base + SPDIF_IN_CTRL); + writel(0xF, host->io_base + SPDIF_IN_IRQ_MASK); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ctrl = readl(host->io_base + SPDIF_IN_CTRL); + ctrl &= ~(SPDIF_IN_SAMPLE | SPDIF_IN_ENB); + writel(ctrl, host->io_base + SPDIF_IN_CTRL); + writel(0x0, host->io_base + SPDIF_IN_IRQ_MASK); + + if (host->reset_perip) + host->reset_perip(); + clk_disable(host->clk); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static struct snd_soc_dai_ops spdif_in_dai_ops = { + .startup = spdif_in_startup, + .shutdown = spdif_in_shutdown, + .trigger = spdif_in_trigger, + .hw_params = spdif_in_hw_params, +}; + +struct snd_soc_dai_driver spdif_in_dai = { + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000), + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE, + }, + .ops = &spdif_in_dai_ops, +}; + +static irqreturn_t spdif_in_irq(int irq, void *arg) +{ + struct spdif_in_dev *host = (struct spdif_in_dev *)arg; + + u32 irq_status = readl(host->io_base + SPDIF_IN_IRQ); + + if (!irq_status) + return IRQ_NONE; + + if (irq_status & SPDIF_IRQ_FIFOWRITE) + dev_err(host->dev, "spdif in: fifo write error"); + if (irq_status & SPDIF_IRQ_EMPTYFIFOREAD) + dev_err(host->dev, "spdif in: empty fifo read error"); + if (irq_status & SPDIF_IRQ_FIFOFULL) + dev_err(host->dev, "spdif in: fifo full error"); + if (irq_status & SPDIF_IRQ_OUTOFRANGE) + dev_err(host->dev, "spdif in: out of range error"); + + writel(0, host->io_base + SPDIF_IN_IRQ); + + return IRQ_HANDLED; +} + +static int spdif_in_probe(struct platform_device *pdev) +{ + struct spdif_in_dev *host; + struct spear_spdif_platform_data *pdata; + struct resource *res, *res_fifo; + int ret; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) + return -EINVAL; + + res_fifo = platform_get_resource(pdev, IORESOURCE_IO, 0); + if (!res_fifo) + return -EINVAL; + + if (!devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), pdev->name)) { + dev_warn(&pdev->dev, "Failed to get memory resourse\n"); + return -ENOENT; + } + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) { + dev_warn(&pdev->dev, "kzalloc fail\n"); + return -ENOMEM; + } + + host->io_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + if (!host->io_base) { + dev_warn(&pdev->dev, "ioremap failed\n"); + return -ENOMEM; + } + + host->irq = platform_get_irq(pdev, 0); + if (host->irq < 0) + return -EINVAL; + + host->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(host->clk)) + return PTR_ERR(host->clk); + + pdata = dev_get_platdata(&pdev->dev); + + if (!pdata) + return -EINVAL; + + host->dma_params.data = pdata->dma_params; + host->dma_params.addr = res_fifo->start; + host->dma_params.max_burst = 16; + host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + host->dma_params.filter = pdata->filter; + host->reset_perip = pdata->reset_perip; + + host->dev = &pdev->dev; + dev_set_drvdata(&pdev->dev, host); + + ret = devm_request_irq(&pdev->dev, host->irq, spdif_in_irq, 0, + "spdif-in", host); + if (ret) { + clk_put(host->clk); + dev_warn(&pdev->dev, "request_irq failed\n"); + return ret; + } + + ret = snd_soc_register_dai(&pdev->dev, &spdif_in_dai); + if (ret != 0) { + clk_put(host->clk); + return ret; + } + + return 0; +} + +static int spdif_in_remove(struct platform_device *pdev) +{ + struct spdif_in_dev *host = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_dai(&pdev->dev); + dev_set_drvdata(&pdev->dev, NULL); + + clk_put(host->clk); + + return 0; +} + + +static struct platform_driver spdif_in_driver = { + .probe = spdif_in_probe, + .remove = spdif_in_remove, + .driver = { + .name = "spdif-in", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(spdif_in_driver); + +MODULE_AUTHOR("Vipin Kumar <vipin.kumar@st.com>"); +MODULE_DESCRIPTION("SPEAr SPDIF IN SoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:spdif_in"); diff --git a/sound/soc/spear/spdif_in_regs.h b/sound/soc/spear/spdif_in_regs.h new file mode 100644 index 000000000000..37af7bc66b7f --- /dev/null +++ b/sound/soc/spear/spdif_in_regs.h @@ -0,0 +1,60 @@ +/* + * SPEAr SPDIF IN controller header file + * + * Copyright (ST) 2011 Vipin Kumar (vipin.kumar@st.com) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef SPDIF_IN_REGS_H +#define SPDIF_IN_REGS_H + +#define SPDIF_IN_CTRL 0x00 + #define SPDIF_IN_PRTYEN (1 << 20) + #define SPDIF_IN_STATEN (1 << 19) + #define SPDIF_IN_USREN (1 << 18) + #define SPDIF_IN_VALEN (1 << 17) + #define SPDIF_IN_BLKEN (1 << 16) + + #define SPDIF_MODE_24BIT (8 << 12) + #define SPDIF_MODE_23BIT (7 << 12) + #define SPDIF_MODE_22BIT (6 << 12) + #define SPDIF_MODE_21BIT (5 << 12) + #define SPDIF_MODE_20BIT (4 << 12) + #define SPDIF_MODE_19BIT (3 << 12) + #define SPDIF_MODE_18BIT (2 << 12) + #define SPDIF_MODE_17BIT (1 << 12) + #define SPDIF_MODE_16BIT (0 << 12) + #define SPDIF_MODE_MASK (0x0F << 12) + + #define SPDIF_IN_VALID (1 << 11) + #define SPDIF_IN_SAMPLE (1 << 10) + #define SPDIF_DATA_SWAP (1 << 9) + #define SPDIF_IN_ENB (1 << 8) + #define SPDIF_DATA_REVERT (1 << 7) + #define SPDIF_XTRACT_16BIT (1 << 6) + #define SPDIF_FIFO_THRES_16 (16 << 0) + +#define SPDIF_IN_IRQ_MASK 0x04 +#define SPDIF_IN_IRQ 0x08 + #define SPDIF_IRQ_FIFOWRITE (1 << 0) + #define SPDIF_IRQ_EMPTYFIFOREAD (1 << 1) + #define SPDIF_IRQ_FIFOFULL (1 << 2) + #define SPDIF_IRQ_OUTOFRANGE (1 << 3) + +#define SPDIF_IN_STA 0x0C + #define SPDIF_IN_LOCK (0x1 << 0) + +#endif /* SPDIF_IN_REGS_H */ diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c new file mode 100644 index 000000000000..5eac4cda2fd7 --- /dev/null +++ b/sound/soc/spear/spdif_out.c @@ -0,0 +1,389 @@ +/* + * ALSA SoC SPDIF Out Audio Layer for spear processors + * + * Copyright (C) 2012 ST Microelectronics + * Vipin Kumar <vipin.kumar@st.com> + * + * This file is licensed under the terms of the GNU General Public + * License version 2. This program is licensed "as is" without any + * warranty of any kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/ioport.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/spear_dma.h> +#include <sound/spear_spdif.h> +#include "spdif_out_regs.h" + +struct spdif_out_params { + u32 rate; + u32 core_freq; + u32 mute; +}; + +struct spdif_out_dev { + struct clk *clk; + struct spear_dma_data dma_params; + struct spdif_out_params saved_params; + u32 running; + void __iomem *io_base; +}; + +static void spdif_out_configure(struct spdif_out_dev *host) +{ + writel(SPDIF_OUT_RESET, host->io_base + SPDIF_OUT_SOFT_RST); + mdelay(1); + writel(readl(host->io_base + SPDIF_OUT_SOFT_RST) & ~SPDIF_OUT_RESET, + host->io_base + SPDIF_OUT_SOFT_RST); + + writel(SPDIF_OUT_FDMA_TRIG_16 | SPDIF_OUT_MEMFMT_16_16 | + SPDIF_OUT_VALID_HW | SPDIF_OUT_USER_HW | + SPDIF_OUT_CHNLSTA_HW | SPDIF_OUT_PARITY_HW, + host->io_base + SPDIF_OUT_CFG); + + writel(0x7F, host->io_base + SPDIF_OUT_INT_STA_CLR); + writel(0x7F, host->io_base + SPDIF_OUT_INT_EN_CLR); +} + +static int spdif_out_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + int ret; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -EINVAL; + + snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)&host->dma_params); + + ret = clk_enable(host->clk); + if (ret) + return ret; + + host->running = true; + spdif_out_configure(host); + + return 0; +} + +static void spdif_out_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return; + + clk_disable(host->clk); + host->running = false; + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static void spdif_out_clock(struct spdif_out_dev *host, u32 core_freq, + u32 rate) +{ + u32 divider, ctrl; + + clk_set_rate(host->clk, core_freq); + divider = DIV_ROUND_CLOSEST(clk_get_rate(host->clk), (rate * 128)); + + ctrl = readl(host->io_base + SPDIF_OUT_CTRL); + ctrl &= ~SPDIF_DIVIDER_MASK; + ctrl |= (divider << SPDIF_DIVIDER_SHIFT) & SPDIF_DIVIDER_MASK; + writel(ctrl, host->io_base + SPDIF_OUT_CTRL); +} + +static int spdif_out_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai); + u32 rate, core_freq; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -EINVAL; + + rate = params_rate(params); + + switch (rate) { + case 8000: + case 16000: + case 32000: + case 64000: + /* + * The clock is multiplied by 10 to bring it to feasible range + * of frequencies for sscg + */ + core_freq = 64000 * 128 * 10; /* 81.92 MHz */ + break; + case 5512: + case 11025: + case 22050: + case 44100: + case 88200: + case 176400: + core_freq = 176400 * 128; /* 22.5792 MHz */ + break; + case 48000: + case 96000: + case 192000: + default: + core_freq = 192000 * 128; /* 24.576 MHz */ + break; + } + + spdif_out_clock(host, core_freq, rate); + host->saved_params.core_freq = core_freq; + host->saved_params.rate = rate; + + return 0; +} + +static int spdif_out_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai); + u32 ctrl; + int ret = 0; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ctrl = readl(host->io_base + SPDIF_OUT_CTRL); + ctrl &= ~SPDIF_OPMODE_MASK; + if (!host->saved_params.mute) + ctrl |= SPDIF_OPMODE_AUD_DATA | + SPDIF_STATE_NORMAL; + else + ctrl |= SPDIF_OPMODE_MUTE_PCM; + writel(ctrl, host->io_base + SPDIF_OUT_CTRL); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ctrl = readl(host->io_base + SPDIF_OUT_CTRL); + ctrl &= ~SPDIF_OPMODE_MASK; + ctrl |= SPDIF_OPMODE_OFF; + writel(ctrl, host->io_base + SPDIF_OUT_CTRL); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int spdif_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai); + u32 val; + + host->saved_params.mute = mute; + val = readl(host->io_base + SPDIF_OUT_CTRL); + val &= ~SPDIF_OPMODE_MASK; + + if (mute) + val |= SPDIF_OPMODE_MUTE_PCM; + else { + if (host->running) + val |= SPDIF_OPMODE_AUD_DATA | SPDIF_STATE_NORMAL; + else + val |= SPDIF_OPMODE_OFF; + } + + writel(val, host->io_base + SPDIF_OUT_CTRL); + return 0; +} + +static int spdif_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = codec->card; + struct snd_soc_pcm_runtime *rtd = card->rtd; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + + ucontrol->value.integer.value[0] = host->saved_params.mute; + return 0; +} + +static int spdif_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = codec->card; + struct snd_soc_pcm_runtime *rtd = card->rtd; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + + if (host->saved_params.mute == ucontrol->value.integer.value[0]) + return 0; + + spdif_digital_mute(cpu_dai, ucontrol->value.integer.value[0]); + + return 1; +} +static const struct snd_kcontrol_new spdif_out_controls[] = { + SOC_SINGLE_BOOL_EXT("IEC958 Playback Switch", 0, + spdif_mute_get, spdif_mute_put), +}; + +int spdif_soc_dai_probe(struct snd_soc_dai *dai) +{ + return snd_soc_add_dai_controls(dai, spdif_out_controls, + ARRAY_SIZE(spdif_out_controls)); +} + +static const struct snd_soc_dai_ops spdif_out_dai_ops = { + .digital_mute = spdif_digital_mute, + .startup = spdif_out_startup, + .shutdown = spdif_out_shutdown, + .trigger = spdif_out_trigger, + .hw_params = spdif_out_hw_params, +}; + +static struct snd_soc_dai_driver spdif_out_dai = { + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .probe = spdif_soc_dai_probe, + .ops = &spdif_out_dai_ops, +}; + +static int spdif_out_probe(struct platform_device *pdev) +{ + struct spdif_out_dev *host; + struct spear_spdif_platform_data *pdata; + struct resource *res; + int ret; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) + return -EINVAL; + + if (!devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), pdev->name)) { + dev_warn(&pdev->dev, "Failed to get memory resourse\n"); + return -ENOENT; + } + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) { + dev_warn(&pdev->dev, "kzalloc fail\n"); + return -ENOMEM; + } + + host->io_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + if (!host->io_base) { + dev_warn(&pdev->dev, "ioremap failed\n"); + return -ENOMEM; + } + + host->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(host->clk)) + return PTR_ERR(host->clk); + + pdata = dev_get_platdata(&pdev->dev); + + host->dma_params.data = pdata->dma_params; + host->dma_params.addr = res->start + SPDIF_OUT_FIFO_DATA; + host->dma_params.max_burst = 16; + host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + host->dma_params.filter = pdata->filter; + + dev_set_drvdata(&pdev->dev, host); + + ret = snd_soc_register_dai(&pdev->dev, &spdif_out_dai); + if (ret != 0) { + clk_put(host->clk); + return ret; + } + + return 0; +} + +static int spdif_out_remove(struct platform_device *pdev) +{ + struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_dai(&pdev->dev); + dev_set_drvdata(&pdev->dev, NULL); + + clk_put(host->clk); + + return 0; +} + +#ifdef CONFIG_PM +static int spdif_out_suspend(struct device *dev) +{ + struct platform_device *pdev = to_platform_device(dev); + struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev); + + if (host->running) + clk_disable(host->clk); + + return 0; +} + +static int spdif_out_resume(struct device *dev) +{ + struct platform_device *pdev = to_platform_device(dev); + struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev); + + if (host->running) { + clk_enable(host->clk); + spdif_out_configure(host); + spdif_out_clock(host, host->saved_params.core_freq, + host->saved_params.rate); + } + return 0; +} + +static SIMPLE_DEV_PM_OPS(spdif_out_dev_pm_ops, spdif_out_suspend, \ + spdif_out_resume); + +#define SPDIF_OUT_DEV_PM_OPS (&spdif_out_dev_pm_ops) + +#else +#define SPDIF_OUT_DEV_PM_OPS NULL + +#endif + +static struct platform_driver spdif_out_driver = { + .probe = spdif_out_probe, + .remove = spdif_out_remove, + .driver = { + .name = "spdif-out", + .owner = THIS_MODULE, + .pm = SPDIF_OUT_DEV_PM_OPS, + }, +}; + +module_platform_driver(spdif_out_driver); + +MODULE_AUTHOR("Vipin Kumar <vipin.kumar@st.com>"); +MODULE_DESCRIPTION("SPEAr SPDIF OUT SoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:spdif_out"); diff --git a/sound/soc/spear/spdif_out_regs.h b/sound/soc/spear/spdif_out_regs.h new file mode 100644 index 000000000000..a5e53324b452 --- /dev/null +++ b/sound/soc/spear/spdif_out_regs.h @@ -0,0 +1,79 @@ +/* + * SPEAr SPDIF OUT controller header file + * + * Copyright (ST) 2011 Vipin Kumar (vipin.kumar@st.com) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef SPDIF_OUT_REGS_H +#define SPDIF_OUT_REGS_H + +#define SPDIF_OUT_SOFT_RST 0x00 + #define SPDIF_OUT_RESET (1 << 0) +#define SPDIF_OUT_FIFO_DATA 0x04 +#define SPDIF_OUT_INT_STA 0x08 +#define SPDIF_OUT_INT_STA_CLR 0x0C + #define SPDIF_INT_UNDERFLOW (1 << 0) + #define SPDIF_INT_EODATA (1 << 1) + #define SPDIF_INT_EOBLOCK (1 << 2) + #define SPDIF_INT_EOLATENCY (1 << 3) + #define SPDIF_INT_EOPD_DATA (1 << 4) + #define SPDIF_INT_MEMFULLREAD (1 << 5) + #define SPDIF_INT_EOPD_PAUSE (1 << 6) + +#define SPDIF_OUT_INT_EN 0x10 +#define SPDIF_OUT_INT_EN_SET 0x14 +#define SPDIF_OUT_INT_EN_CLR 0x18 +#define SPDIF_OUT_CTRL 0x1C + #define SPDIF_OPMODE_MASK (7 << 0) + #define SPDIF_OPMODE_OFF (0 << 0) + #define SPDIF_OPMODE_MUTE_PCM (1 << 0) + #define SPDIF_OPMODE_MUTE_PAUSE (2 << 0) + #define SPDIF_OPMODE_AUD_DATA (3 << 0) + #define SPDIF_OPMODE_ENCODE (4 << 0) + #define SPDIF_STATE_NORMAL (1 << 3) + #define SPDIF_DIVIDER_MASK (0xff << 5) + #define SPDIF_DIVIDER_SHIFT (5) + #define SPDIF_SAMPLEREAD_MASK (0x1ffff << 15) + #define SPDIF_SAMPLEREAD_SHIFT (15) +#define SPDIF_OUT_STA 0x20 +#define SPDIF_OUT_PA_PB 0x24 +#define SPDIF_OUT_PC_PD 0x28 +#define SPDIF_OUT_CL1 0x2C +#define SPDIF_OUT_CR1 0x30 +#define SPDIF_OUT_CL2_CR2_UV 0x34 +#define SPDIF_OUT_PAUSE_LAT 0x38 +#define SPDIF_OUT_FRMLEN_BRST 0x3C +#define SPDIF_OUT_CFG 0x40 + #define SPDIF_OUT_MEMFMT_16_0 (0 << 5) + #define SPDIF_OUT_MEMFMT_16_16 (1 << 5) + #define SPDIF_OUT_VALID_DMA (0 << 3) + #define SPDIF_OUT_VALID_HW (1 << 3) + #define SPDIF_OUT_USER_DMA (0 << 2) + #define SPDIF_OUT_USER_HW (1 << 2) + #define SPDIF_OUT_CHNLSTA_DMA (0 << 1) + #define SPDIF_OUT_CHNLSTA_HW (1 << 1) + #define SPDIF_OUT_PARITY_HW (0 << 0) + #define SPDIF_OUT_PARITY_DMA (1 << 0) + #define SPDIF_OUT_FDMA_TRIG_2 (2 << 8) + #define SPDIF_OUT_FDMA_TRIG_6 (6 << 8) + #define SPDIF_OUT_FDMA_TRIG_8 (8 << 8) + #define SPDIF_OUT_FDMA_TRIG_10 (10 << 8) + #define SPDIF_OUT_FDMA_TRIG_12 (12 << 8) + #define SPDIF_OUT_FDMA_TRIG_16 (16 << 8) + #define SPDIF_OUT_FDMA_TRIG_18 (18 << 8) + +#endif /* SPDIF_OUT_REGS_H */ diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c new file mode 100644 index 000000000000..97c2cac8e92c --- /dev/null +++ b/sound/soc/spear/spear_pcm.c @@ -0,0 +1,214 @@ +/* + * ALSA PCM interface for ST SPEAr Processors + * + * sound/soc/spear/spear_pcm.c + * + * Copyright (C) 2012 ST Microelectronics + * Rajeev Kumar<rajeev-dlh.kumar@st.com> + * + * This file is licensed under the terms of the GNU General Public + * License version 2. This program is licensed "as is" without any + * warranty of any kind, whether express or implied. + */ + +#include <linux/module.h> +#include <linux/dmaengine.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/scatterlist.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/spear_dma.h> + +struct snd_pcm_hardware spear_pcm_hardware = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), + .buffer_bytes_max = 16 * 1024, /* max buffer size */ + .period_bytes_min = 2 * 1024, /* 1 msec data minimum period size */ + .period_bytes_max = 2 * 1024, /* maximum period size */ + .periods_min = 1, /* min # periods */ + .periods_max = 8, /* max # of periods */ + .fifo_size = 0, /* fifo size in bytes */ +}; + +static int spear_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static int spear_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + + return 0; +} + +static int spear_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + struct spear_dma_data *dma_data = (struct spear_dma_data *) + snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + int ret; + + ret = snd_soc_set_runtime_hwparams(substream, &spear_pcm_hardware); + if (ret) + return ret; + + ret = snd_dmaengine_pcm_open(substream, dma_data->filter, dma_data); + if (ret) + return ret; + + snd_dmaengine_pcm_set_data(substream, dma_data); + + return 0; +} + +static int spear_pcm_close(struct snd_pcm_substream *substream) +{ + + snd_dmaengine_pcm_close(substream); + + return 0; +} + +static int spear_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops spear_pcm_ops = { + .open = spear_pcm_open, + .close = spear_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = spear_pcm_hw_params, + .hw_free = spear_pcm_hw_free, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, + .mmap = spear_pcm_mmap, +}; + +static int +spear_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream, + size_t size) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + + dev_info(buf->dev.dev, + " preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", + (void *)buf->area, (void *)buf->addr, size); + + buf->bytes = size; + return 0; +} + +static void spear_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf && !buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static u64 spear_pcm_dmamask = DMA_BIT_MASK(32); + +static int spear_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int ret; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &spear_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (dai->driver->playback.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK, + spear_pcm_hardware.buffer_bytes_max); + if (ret) + return ret; + } + + if (dai->driver->capture.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE, + spear_pcm_hardware.buffer_bytes_max); + if (ret) + return ret; + } + + return 0; +} + +struct snd_soc_platform_driver spear_soc_platform = { + .ops = &spear_pcm_ops, + .pcm_new = spear_pcm_new, + .pcm_free = spear_pcm_free, +}; + +static int __devinit spear_soc_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &spear_soc_platform); +} + +static int __devexit spear_soc_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + + return 0; +} + +static struct platform_driver spear_pcm_driver = { + .driver = { + .name = "spear-pcm-audio", + .owner = THIS_MODULE, + }, + + .probe = spear_soc_platform_probe, + .remove = __devexit_p(spear_soc_platform_remove), +}; + +module_platform_driver(spear_pcm_driver); + +MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>"); +MODULE_DESCRIPTION("SPEAr PCM DMA module"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:spear-pcm-audio"); diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index ce1b773c351f..02bcd308c189 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -1,38 +1,69 @@ config SND_SOC_TEGRA tristate "SoC Audio for the Tegra System-on-Chip" - depends on ARCH_TEGRA && TEGRA_SYSTEM_DMA + depends on ARCH_TEGRA && (TEGRA_SYSTEM_DMA || TEGRA20_APB_DMA) + select REGMAP_MMIO + select SND_SOC_DMAENGINE_PCM if TEGRA20_APB_DMA help Say Y or M here if you want support for SoC audio on Tegra. -config SND_SOC_TEGRA_I2S +config SND_SOC_TEGRA20_DAS tristate - depends on SND_SOC_TEGRA + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + help + Say Y or M if you want to add support for the Tegra20 DAS module. + You will also need to select the individual machine drivers to + support below. + +config SND_SOC_TEGRA20_I2S + tristate + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA20_DAS help Say Y or M if you want to add support for codecs attached to the - Tegra I2S interface. You will also need to select the individual + Tegra20 I2S interface. You will also need to select the individual machine drivers to support below. -config SND_SOC_TEGRA_SPDIF +config SND_SOC_TEGRA20_SPDIF tristate - depends on SND_SOC_TEGRA + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC default m help - Say Y or M if you want to add support for the SPDIF interface. + Say Y or M if you want to add support for the Tegra20 SPDIF interface. You will also need to select the individual machine drivers to support below. -config MACH_HAS_SND_SOC_TEGRA_WM8903 - bool +config SND_SOC_TEGRA30_AHUB + tristate + depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC help - Machines that use the SND_SOC_TEGRA_WM8903 driver should select - this config option, in order to allow the user to enable - SND_SOC_TEGRA_WM8903. + Say Y or M if you want to add support for the Tegra20 AHUB module. + You will also need to select the individual machine drivers to + support below. + +config SND_SOC_TEGRA30_I2S + tristate + depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC + select SND_SOC_TEGRA30_AHUB + help + Say Y or M if you want to add support for codecs attached to the + Tegra30 I2S interface. You will also need to select the individual + machine drivers to support below. + +config SND_SOC_TEGRA_WM8753 + tristate "SoC Audio support for Tegra boards using a WM8753 codec" + depends on SND_SOC_TEGRA && I2C + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC + select SND_SOC_WM8753 + help + Say Y or M here if you want to add support for SoC audio on Tegra + boards using the WM8753 codec, such as Whistler. config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" depends on SND_SOC_TEGRA && I2C - depends on MACH_HAS_SND_SOC_TEGRA_WM8903 - select SND_SOC_TEGRA_I2S + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_WM8903 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -41,18 +72,18 @@ config SND_SOC_TEGRA_WM8903 config SND_SOC_TEGRA_TRIMSLICE tristate "SoC Audio support for TrimSlice board" - depends on SND_SOC_TEGRA && MACH_TRIMSLICE && I2C - select SND_SOC_TEGRA_I2S + depends on SND_SOC_TEGRA && I2C + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_TLV320AIC23 help Say Y or M here if you want to add support for SoC audio on the TrimSlice platform. config SND_SOC_TEGRA_ALC5632 - tristate "SoC Audio support for Tegra boards using an ALC5632 codec" - depends on SND_SOC_TEGRA && I2C - select SND_SOC_TEGRA_I2S - select SND_SOC_ALC5632 - help - Say Y or M here if you want to add support for SoC audio on the - Toshiba AC100 netbook. + tristate "SoC Audio support for Tegra boards using an ALC5632 codec" + depends on SND_SOC_TEGRA && I2C + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_ALC5632 + help + Say Y or M here if you want to add support for SoC audio on the + Toshiba AC100 netbook. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 8e584b8fcfba..391e78a34c06 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -1,21 +1,27 @@ # Tegra platform Support -snd-soc-tegra-das-objs := tegra_das.o snd-soc-tegra-pcm-objs := tegra_pcm.o -snd-soc-tegra-i2s-objs := tegra_i2s.o -snd-soc-tegra-spdif-objs := tegra_spdif.o snd-soc-tegra-utils-objs += tegra_asoc_utils.o +snd-soc-tegra20-das-objs := tegra20_das.o +snd-soc-tegra20-i2s-objs := tegra20_i2s.o +snd-soc-tegra20-spdif-objs := tegra20_spdif.o +snd-soc-tegra30-ahub-objs := tegra30_ahub.o +snd-soc-tegra30-i2s-objs := tegra30_i2s.o -obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-utils.o -obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-das.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o -obj-$(CONFIG_SND_SOC_TEGRA_I2S) += snd-soc-tegra-i2s.o -obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o +obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-utils.o +obj-$(CONFIG_SND_SOC_TEGRA20_DAS) += snd-soc-tegra20-das.o +obj-$(CONFIG_SND_SOC_TEGRA20_I2S) += snd-soc-tegra20-i2s.o +obj-$(CONFIG_SND_SOC_TEGRA20_SPDIF) += snd-soc-tegra20-spdif.o +obj-$(CONFIG_SND_SOC_TEGRA30_AHUB) += snd-soc-tegra30-ahub.o +obj-$(CONFIG_SND_SOC_TEGRA30_I2S) += snd-soc-tegra30-i2s.o # Tegra machine Support +snd-soc-tegra-wm8753-objs := tegra_wm8753.o snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-trimslice-objs := trimslice.o snd-soc-tegra-alc5632-objs := tegra_alc5632.o +obj-$(CONFIG_SND_SOC_TEGRA_WM8753) += snd-soc-tegra-wm8753.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o obj-$(CONFIG_SND_SOC_TEGRA_ALC5632) += snd-soc-tegra-alc5632.o diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c new file mode 100644 index 000000000000..bf99296bce95 --- /dev/null +++ b/sound/soc/tegra/tegra20_das.c @@ -0,0 +1,233 @@ +/* + * tegra20_das.c - Tegra20 DAS driver + * + * Author: Stephen Warren <swarren@nvidia.com> + * Copyright (C) 2010 - NVIDIA, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/device.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/soc.h> +#include "tegra20_das.h" + +#define DRV_NAME "tegra20-das" + +static struct tegra20_das *das; + +static inline void tegra20_das_write(u32 reg, u32 val) +{ + regmap_write(das->regmap, reg, val); +} + +static inline u32 tegra20_das_read(u32 reg) +{ + u32 val; + regmap_read(das->regmap, reg, &val); + return val; +} + +int tegra20_das_connect_dap_to_dac(int dap, int dac) +{ + u32 addr; + u32 reg; + + if (!das) + return -ENODEV; + + addr = TEGRA20_DAS_DAP_CTRL_SEL + + (dap * TEGRA20_DAS_DAP_CTRL_SEL_STRIDE); + reg = dac << TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P; + + tegra20_das_write(addr, reg); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra20_das_connect_dap_to_dac); + +int tegra20_das_connect_dap_to_dap(int dap, int otherdap, int master, + int sdata1rx, int sdata2rx) +{ + u32 addr; + u32 reg; + + if (!das) + return -ENODEV; + + addr = TEGRA20_DAS_DAP_CTRL_SEL + + (dap * TEGRA20_DAS_DAP_CTRL_SEL_STRIDE); + reg = otherdap << TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P | + !!sdata2rx << TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P | + !!sdata1rx << TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P | + !!master << TEGRA20_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P; + + tegra20_das_write(addr, reg); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra20_das_connect_dap_to_dap); + +int tegra20_das_connect_dac_to_dap(int dac, int dap) +{ + u32 addr; + u32 reg; + + if (!das) + return -ENODEV; + + addr = TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL + + (dac * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE); + reg = dap << TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P | + dap << TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P | + dap << TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P; + + tegra20_das_write(addr, reg); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra20_das_connect_dac_to_dap); + +#define LAST_REG(name) \ + (TEGRA20_DAS_##name + \ + (TEGRA20_DAS_##name##_STRIDE * (TEGRA20_DAS_##name##_COUNT - 1))) + +static bool tegra20_das_wr_rd_reg(struct device *dev, unsigned int reg) +{ + if ((reg >= TEGRA20_DAS_DAP_CTRL_SEL) && + (reg <= LAST_REG(DAP_CTRL_SEL))) + return true; + if ((reg >= TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL) && + (reg <= LAST_REG(DAC_INPUT_DATA_CLK_SEL))) + return true; + + return false; +} + +static const struct regmap_config tegra20_das_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = LAST_REG(DAC_INPUT_DATA_CLK_SEL), + .writeable_reg = tegra20_das_wr_rd_reg, + .readable_reg = tegra20_das_wr_rd_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit tegra20_das_probe(struct platform_device *pdev) +{ + struct resource *res, *region; + void __iomem *regs; + int ret = 0; + + if (das) + return -ENODEV; + + das = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_das), GFP_KERNEL); + if (!das) { + dev_err(&pdev->dev, "Can't allocate tegra20_das\n"); + ret = -ENOMEM; + goto err; + } + das->dev = &pdev->dev; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "No memory resource\n"); + ret = -ENODEV; + goto err; + } + + region = devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), pdev->name); + if (!region) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + ret = -EBUSY; + goto err; + } + + regs = devm_ioremap(&pdev->dev, res->start, resource_size(res)); + if (!regs) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err; + } + + das->regmap = devm_regmap_init_mmio(&pdev->dev, regs, + &tegra20_das_regmap_config); + if (IS_ERR(das->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + ret = PTR_ERR(das->regmap); + goto err; + } + + ret = tegra20_das_connect_dap_to_dac(TEGRA20_DAS_DAP_ID_1, + TEGRA20_DAS_DAP_SEL_DAC1); + if (ret) { + dev_err(&pdev->dev, "Can't set up DAS DAP connection\n"); + goto err; + } + ret = tegra20_das_connect_dac_to_dap(TEGRA20_DAS_DAC_ID_1, + TEGRA20_DAS_DAC_SEL_DAP1); + if (ret) { + dev_err(&pdev->dev, "Can't set up DAS DAC connection\n"); + goto err; + } + + platform_set_drvdata(pdev, das); + + return 0; + +err: + das = NULL; + return ret; +} + +static int __devexit tegra20_das_remove(struct platform_device *pdev) +{ + if (!das) + return -ENODEV; + + das = NULL; + + return 0; +} + +static const struct of_device_id tegra20_das_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra20-das", }, + {}, +}; + +static struct platform_driver tegra20_das_driver = { + .probe = tegra20_das_probe, + .remove = __devexit_p(tegra20_das_remove), + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = tegra20_das_of_match, + }, +}; +module_platform_driver(tegra20_das_driver); + +MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); +MODULE_DESCRIPTION("Tegra20 DAS driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra20_das_of_match); diff --git a/sound/soc/tegra/tegra20_das.h b/sound/soc/tegra/tegra20_das.h new file mode 100644 index 000000000000..be217f3d3a75 --- /dev/null +++ b/sound/soc/tegra/tegra20_das.h @@ -0,0 +1,134 @@ +/* + * tegra20_das.h - Definitions for Tegra20 DAS driver + * + * Author: Stephen Warren <swarren@nvidia.com> + * Copyright (C) 2010,2012 - NVIDIA, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TEGRA20_DAS_H__ +#define __TEGRA20_DAS_H__ + +/* Register TEGRA20_DAS_DAP_CTRL_SEL */ +#define TEGRA20_DAS_DAP_CTRL_SEL 0x00 +#define TEGRA20_DAS_DAP_CTRL_SEL_COUNT 5 +#define TEGRA20_DAS_DAP_CTRL_SEL_STRIDE 4 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P 31 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_MS_SEL_S 1 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P 30 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_S 1 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P 29 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_S 1 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P 0 +#define TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_S 5 + +/* Values for field TEGRA20_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL */ +#define TEGRA20_DAS_DAP_SEL_DAC1 0 +#define TEGRA20_DAS_DAP_SEL_DAC2 1 +#define TEGRA20_DAS_DAP_SEL_DAC3 2 +#define TEGRA20_DAS_DAP_SEL_DAP1 16 +#define TEGRA20_DAS_DAP_SEL_DAP2 17 +#define TEGRA20_DAS_DAP_SEL_DAP3 18 +#define TEGRA20_DAS_DAP_SEL_DAP4 19 +#define TEGRA20_DAS_DAP_SEL_DAP5 20 + +/* Register TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL */ +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL 0x40 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT 3 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE 4 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P 28 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_S 4 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P 24 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_S 4 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P 0 +#define TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_S 4 + +/* + * Values for: + * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL + * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL + * TEGRA20_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL + */ +#define TEGRA20_DAS_DAC_SEL_DAP1 0 +#define TEGRA20_DAS_DAC_SEL_DAP2 1 +#define TEGRA20_DAS_DAC_SEL_DAP3 2 +#define TEGRA20_DAS_DAC_SEL_DAP4 3 +#define TEGRA20_DAS_DAC_SEL_DAP5 4 + +/* + * Names/IDs of the DACs/DAPs. + */ + +#define TEGRA20_DAS_DAP_ID_1 0 +#define TEGRA20_DAS_DAP_ID_2 1 +#define TEGRA20_DAS_DAP_ID_3 2 +#define TEGRA20_DAS_DAP_ID_4 3 +#define TEGRA20_DAS_DAP_ID_5 4 + +#define TEGRA20_DAS_DAC_ID_1 0 +#define TEGRA20_DAS_DAC_ID_2 1 +#define TEGRA20_DAS_DAC_ID_3 2 + +struct tegra20_das { + struct device *dev; + struct regmap *regmap; +}; + +/* + * Terminology: + * DAS: Digital audio switch (HW module controlled by this driver) + * DAP: Digital audio port (port/pins on Tegra device) + * DAC: Digital audio controller (e.g. I2S or AC97 controller elsewhere) + * + * The Tegra DAS is a mux/cross-bar which can connect each DAP to a specific + * DAC, or another DAP. When DAPs are connected, one must be the master and + * one the slave. Each DAC allows selection of a specific DAP for input, to + * cater for the case where N DAPs are connected to 1 DAC for broadcast + * output. + * + * This driver is dumb; no attempt is made to ensure that a valid routing + * configuration is programmed. + */ + +/* + * Connect a DAP to to a DAC + * dap_id: DAP to connect: TEGRA20_DAS_DAP_ID_* + * dac_sel: DAC to connect to: TEGRA20_DAS_DAP_SEL_DAC* + */ +extern int tegra20_das_connect_dap_to_dac(int dap_id, int dac_sel); + +/* + * Connect a DAP to to another DAP + * dap_id: DAP to connect: TEGRA20_DAS_DAP_ID_* + * other_dap_sel: DAP to connect to: TEGRA20_DAS_DAP_SEL_DAP* + * master: Is this DAP the master (1) or slave (0) + * sdata1rx: Is this DAP's SDATA1 pin RX (1) or TX (0) + * sdata2rx: Is this DAP's SDATA2 pin RX (1) or TX (0) + */ +extern int tegra20_das_connect_dap_to_dap(int dap_id, int other_dap_sel, + int master, int sdata1rx, + int sdata2rx); + +/* + * Connect a DAC's input to a DAP + * (DAC outputs are selected by the DAP) + * dac_id: DAC ID to connect: TEGRA20_DAS_DAC_ID_* + * dap_sel: DAP to receive input from: TEGRA20_DAS_DAC_SEL_DAP* + */ +extern int tegra20_das_connect_dac_to_dap(int dac_id, int dap_sel); + +#endif diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c new file mode 100644 index 000000000000..0832e8afd73c --- /dev/null +++ b/sound/soc/tegra/tegra20_i2s.c @@ -0,0 +1,492 @@ +/* + * tegra20_i2s.c - Tegra20 I2S driver + * + * Author: Stephen Warren <swarren@nvidia.com> + * Copyright (C) 2010,2012 - NVIDIA, Inc. + * + * Based on code copyright/by: + * + * Copyright (c) 2009-2010, NVIDIA Corporation. + * Scott Peterson <speterson@nvidia.com> + * + * Copyright (C) 2010 Google, Inc. + * Iliyan Malchev <malchev@google.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "tegra20_i2s.h" + +#define DRV_NAME "tegra20-i2s" + +static int tegra20_i2s_runtime_suspend(struct device *dev) +{ + struct tegra20_i2s *i2s = dev_get_drvdata(dev); + + clk_disable_unprepare(i2s->clk_i2s); + + return 0; +} + +static int tegra20_i2s_runtime_resume(struct device *dev) +{ + struct tegra20_i2s *i2s = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(i2s->clk_i2s); + if (ret) { + dev_err(dev, "clk_enable failed: %d\n", ret); + return ret; + } + + return 0; +} + +static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned int mask, val; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + mask = TEGRA20_I2S_CTRL_MASTER_ENABLE; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + val = TEGRA20_I2S_CTRL_MASTER_ENABLE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + mask |= TEGRA20_I2S_CTRL_BIT_FORMAT_MASK | + TEGRA20_I2S_CTRL_LRCK_MASK; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + val |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP; + val |= TEGRA20_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_DSP_B: + val |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP; + val |= TEGRA20_I2S_CTRL_LRCK_R_LOW; + break; + case SND_SOC_DAIFMT_I2S: + val |= TEGRA20_I2S_CTRL_BIT_FORMAT_I2S; + val |= TEGRA20_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val |= TEGRA20_I2S_CTRL_BIT_FORMAT_RJM; + val |= TEGRA20_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_LEFT_J: + val |= TEGRA20_I2S_CTRL_BIT_FORMAT_LJM; + val |= TEGRA20_I2S_CTRL_LRCK_L_LOW; + break; + default: + return -EINVAL; + } + + regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, mask, val); + + return 0; +} + +static int tegra20_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct device *dev = dai->dev; + struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned int mask, val; + int ret, sample_size, srate, i2sclock, bitcnt; + + mask = TEGRA20_I2S_CTRL_BIT_SIZE_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + val = TEGRA20_I2S_CTRL_BIT_SIZE_16; + sample_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = TEGRA20_I2S_CTRL_BIT_SIZE_24; + sample_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + val = TEGRA20_I2S_CTRL_BIT_SIZE_32; + sample_size = 32; + break; + default: + return -EINVAL; + } + + mask |= TEGRA20_I2S_CTRL_FIFO_FORMAT_MASK; + val |= TEGRA20_I2S_CTRL_FIFO_FORMAT_PACKED; + + regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, mask, val); + + srate = params_rate(params); + + /* Final "* 2" required by Tegra hardware */ + i2sclock = srate * params_channels(params) * sample_size * 2; + + ret = clk_set_rate(i2s->clk_i2s, i2sclock); + if (ret) { + dev_err(dev, "Can't set I2S clock rate: %d\n", ret); + return ret; + } + + bitcnt = (i2sclock / (2 * srate)) - 1; + if (bitcnt < 0 || bitcnt > TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US) + return -EINVAL; + val = bitcnt << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT; + + if (i2sclock % (2 * srate)) + val |= TEGRA20_I2S_TIMING_NON_SYM_ENABLE; + + regmap_write(i2s->regmap, TEGRA20_I2S_TIMING, val); + + regmap_write(i2s->regmap, TEGRA20_I2S_FIFO_SCR, + TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS | + TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS); + + return 0; +} + +static void tegra20_i2s_start_playback(struct tegra20_i2s *i2s) +{ + regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, + TEGRA20_I2S_CTRL_FIFO1_ENABLE, + TEGRA20_I2S_CTRL_FIFO1_ENABLE); +} + +static void tegra20_i2s_stop_playback(struct tegra20_i2s *i2s) +{ + regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, + TEGRA20_I2S_CTRL_FIFO1_ENABLE, 0); +} + +static void tegra20_i2s_start_capture(struct tegra20_i2s *i2s) +{ + regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, + TEGRA20_I2S_CTRL_FIFO2_ENABLE, + TEGRA20_I2S_CTRL_FIFO2_ENABLE); +} + +static void tegra20_i2s_stop_capture(struct tegra20_i2s *i2s) +{ + regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, + TEGRA20_I2S_CTRL_FIFO2_ENABLE, 0); +} + +static int tegra20_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tegra20_i2s_start_playback(i2s); + else + tegra20_i2s_start_capture(i2s); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tegra20_i2s_stop_playback(i2s); + else + tegra20_i2s_stop_capture(i2s); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int tegra20_i2s_probe(struct snd_soc_dai *dai) +{ + struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + dai->capture_dma_data = &i2s->capture_dma_data; + dai->playback_dma_data = &i2s->playback_dma_data; + + return 0; +} + +static const struct snd_soc_dai_ops tegra20_i2s_dai_ops = { + .set_fmt = tegra20_i2s_set_fmt, + .hw_params = tegra20_i2s_hw_params, + .trigger = tegra20_i2s_trigger, +}; + +static const struct snd_soc_dai_driver tegra20_i2s_dai_template = { + .probe = tegra20_i2s_probe, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &tegra20_i2s_dai_ops, + .symmetric_rates = 1, +}; + +static bool tegra20_i2s_wr_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_I2S_CTRL: + case TEGRA20_I2S_STATUS: + case TEGRA20_I2S_TIMING: + case TEGRA20_I2S_FIFO_SCR: + case TEGRA20_I2S_PCM_CTRL: + case TEGRA20_I2S_NW_CTRL: + case TEGRA20_I2S_TDM_CTRL: + case TEGRA20_I2S_TDM_TX_RX_CTRL: + case TEGRA20_I2S_FIFO1: + case TEGRA20_I2S_FIFO2: + return true; + default: + return false; + }; +} + +static bool tegra20_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_I2S_STATUS: + case TEGRA20_I2S_FIFO_SCR: + case TEGRA20_I2S_FIFO1: + case TEGRA20_I2S_FIFO2: + return true; + default: + return false; + }; +} + +static bool tegra20_i2s_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_I2S_FIFO1: + case TEGRA20_I2S_FIFO2: + return true; + default: + return false; + }; +} + +static const struct regmap_config tegra20_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = TEGRA20_I2S_FIFO2, + .writeable_reg = tegra20_i2s_wr_rd_reg, + .readable_reg = tegra20_i2s_wr_rd_reg, + .volatile_reg = tegra20_i2s_volatile_reg, + .precious_reg = tegra20_i2s_precious_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static __devinit int tegra20_i2s_platform_probe(struct platform_device *pdev) +{ + struct tegra20_i2s *i2s; + struct resource *mem, *memregion, *dmareq; + u32 of_dma[2]; + u32 dma_ch; + void __iomem *regs; + int ret; + + i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_i2s), GFP_KERNEL); + if (!i2s) { + dev_err(&pdev->dev, "Can't allocate tegra20_i2s\n"); + ret = -ENOMEM; + goto err; + } + dev_set_drvdata(&pdev->dev, i2s); + + i2s->dai = tegra20_i2s_dai_template; + i2s->dai.name = dev_name(&pdev->dev); + + i2s->clk_i2s = clk_get(&pdev->dev, NULL); + if (IS_ERR(i2s->clk_i2s)) { + dev_err(&pdev->dev, "Can't retrieve i2s clock\n"); + ret = PTR_ERR(i2s->clk_i2s); + goto err; + } + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "No memory resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmareq) { + if (of_property_read_u32_array(pdev->dev.of_node, + "nvidia,dma-request-selector", + of_dma, 2) < 0) { + dev_err(&pdev->dev, "No DMA resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + dma_ch = of_dma[1]; + } else { + dma_ch = dmareq->start; + } + + memregion = devm_request_mem_region(&pdev->dev, mem->start, + resource_size(mem), DRV_NAME); + if (!memregion) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + ret = -EBUSY; + goto err_clk_put; + } + + regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); + if (!regs) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err_clk_put; + } + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, regs, + &tegra20_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + ret = PTR_ERR(i2s->regmap); + goto err_clk_put; + } + + i2s->capture_dma_data.addr = mem->start + TEGRA20_I2S_FIFO2; + i2s->capture_dma_data.wrap = 4; + i2s->capture_dma_data.width = 32; + i2s->capture_dma_data.req_sel = dma_ch; + + i2s->playback_dma_data.addr = mem->start + TEGRA20_I2S_FIFO1; + i2s->playback_dma_data.wrap = 4; + i2s->playback_dma_data.width = 32; + i2s->playback_dma_data.req_sel = dma_ch; + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = tegra20_i2s_runtime_resume(&pdev->dev); + if (ret) + goto err_pm_disable; + } + + ret = snd_soc_register_dai(&pdev->dev, &i2s->dai); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + ret = -ENOMEM; + goto err_suspend; + } + + ret = tegra_pcm_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + goto err_unregister_dai; + } + + return 0; + +err_unregister_dai: + snd_soc_unregister_dai(&pdev->dev); +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra20_i2s_runtime_suspend(&pdev->dev); +err_pm_disable: + pm_runtime_disable(&pdev->dev); +err_clk_put: + clk_put(i2s->clk_i2s); +err: + return ret; +} + +static int __devexit tegra20_i2s_platform_remove(struct platform_device *pdev) +{ + struct tegra20_i2s *i2s = dev_get_drvdata(&pdev->dev); + + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra20_i2s_runtime_suspend(&pdev->dev); + + tegra_pcm_platform_unregister(&pdev->dev); + snd_soc_unregister_dai(&pdev->dev); + + clk_put(i2s->clk_i2s); + + return 0; +} + +static const struct of_device_id tegra20_i2s_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra20-i2s", }, + {}, +}; + +static const struct dev_pm_ops tegra20_i2s_pm_ops __devinitconst = { + SET_RUNTIME_PM_OPS(tegra20_i2s_runtime_suspend, + tegra20_i2s_runtime_resume, NULL) +}; + +static struct platform_driver tegra20_i2s_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = tegra20_i2s_of_match, + .pm = &tegra20_i2s_pm_ops, + }, + .probe = tegra20_i2s_platform_probe, + .remove = __devexit_p(tegra20_i2s_platform_remove), +}; +module_platform_driver(tegra20_i2s_driver); + +MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); +MODULE_DESCRIPTION("Tegra20 I2S ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra20_i2s_of_match); diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h new file mode 100644 index 000000000000..c27069d24d77 --- /dev/null +++ b/sound/soc/tegra/tegra20_i2s.h @@ -0,0 +1,163 @@ +/* + * tegra20_i2s.h - Definitions for Tegra20 I2S driver + * + * Author: Stephen Warren <swarren@nvidia.com> + * Copyright (C) 2010,2012 - NVIDIA, Inc. + * + * Based on code copyright/by: + * + * Copyright (c) 2009-2010, NVIDIA Corporation. + * Scott Peterson <speterson@nvidia.com> + * + * Copyright (C) 2010 Google, Inc. + * Iliyan Malchev <malchev@google.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TEGRA20_I2S_H__ +#define __TEGRA20_I2S_H__ + +#include "tegra_pcm.h" + +/* Register offsets from TEGRA20_I2S1_BASE and TEGRA20_I2S2_BASE */ + +#define TEGRA20_I2S_CTRL 0x00 +#define TEGRA20_I2S_STATUS 0x04 +#define TEGRA20_I2S_TIMING 0x08 +#define TEGRA20_I2S_FIFO_SCR 0x0c +#define TEGRA20_I2S_PCM_CTRL 0x10 +#define TEGRA20_I2S_NW_CTRL 0x14 +#define TEGRA20_I2S_TDM_CTRL 0x20 +#define TEGRA20_I2S_TDM_TX_RX_CTRL 0x24 +#define TEGRA20_I2S_FIFO1 0x40 +#define TEGRA20_I2S_FIFO2 0x80 + +/* Fields in TEGRA20_I2S_CTRL */ + +#define TEGRA20_I2S_CTRL_FIFO2_TX_ENABLE (1 << 30) +#define TEGRA20_I2S_CTRL_FIFO1_ENABLE (1 << 29) +#define TEGRA20_I2S_CTRL_FIFO2_ENABLE (1 << 28) +#define TEGRA20_I2S_CTRL_FIFO1_RX_ENABLE (1 << 27) +#define TEGRA20_I2S_CTRL_FIFO_LPBK_ENABLE (1 << 26) +#define TEGRA20_I2S_CTRL_MASTER_ENABLE (1 << 25) + +#define TEGRA20_I2S_LRCK_LEFT_LOW 0 +#define TEGRA20_I2S_LRCK_RIGHT_LOW 1 + +#define TEGRA20_I2S_CTRL_LRCK_SHIFT 24 +#define TEGRA20_I2S_CTRL_LRCK_MASK (1 << TEGRA20_I2S_CTRL_LRCK_SHIFT) +#define TEGRA20_I2S_CTRL_LRCK_L_LOW (TEGRA20_I2S_LRCK_LEFT_LOW << TEGRA20_I2S_CTRL_LRCK_SHIFT) +#define TEGRA20_I2S_CTRL_LRCK_R_LOW (TEGRA20_I2S_LRCK_RIGHT_LOW << TEGRA20_I2S_CTRL_LRCK_SHIFT) + +#define TEGRA20_I2S_BIT_FORMAT_I2S 0 +#define TEGRA20_I2S_BIT_FORMAT_RJM 1 +#define TEGRA20_I2S_BIT_FORMAT_LJM 2 +#define TEGRA20_I2S_BIT_FORMAT_DSP 3 + +#define TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT 10 +#define TEGRA20_I2S_CTRL_BIT_FORMAT_MASK (3 << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_FORMAT_I2S (TEGRA20_I2S_BIT_FORMAT_I2S << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_FORMAT_RJM (TEGRA20_I2S_BIT_FORMAT_RJM << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_FORMAT_LJM (TEGRA20_I2S_BIT_FORMAT_LJM << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_FORMAT_DSP (TEGRA20_I2S_BIT_FORMAT_DSP << TEGRA20_I2S_CTRL_BIT_FORMAT_SHIFT) + +#define TEGRA20_I2S_BIT_SIZE_16 0 +#define TEGRA20_I2S_BIT_SIZE_20 1 +#define TEGRA20_I2S_BIT_SIZE_24 2 +#define TEGRA20_I2S_BIT_SIZE_32 3 + +#define TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT 8 +#define TEGRA20_I2S_CTRL_BIT_SIZE_MASK (3 << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_SIZE_16 (TEGRA20_I2S_BIT_SIZE_16 << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_SIZE_20 (TEGRA20_I2S_BIT_SIZE_20 << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_SIZE_24 (TEGRA20_I2S_BIT_SIZE_24 << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA20_I2S_CTRL_BIT_SIZE_32 (TEGRA20_I2S_BIT_SIZE_32 << TEGRA20_I2S_CTRL_BIT_SIZE_SHIFT) + +#define TEGRA20_I2S_FIFO_16_LSB 0 +#define TEGRA20_I2S_FIFO_20_LSB 1 +#define TEGRA20_I2S_FIFO_24_LSB 2 +#define TEGRA20_I2S_FIFO_32 3 +#define TEGRA20_I2S_FIFO_PACKED 7 + +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT 4 +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_MASK (7 << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_16_LSB (TEGRA20_I2S_FIFO_16_LSB << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_20_LSB (TEGRA20_I2S_FIFO_20_LSB << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_24_LSB (TEGRA20_I2S_FIFO_24_LSB << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_32 (TEGRA20_I2S_FIFO_32 << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA20_I2S_CTRL_FIFO_FORMAT_PACKED (TEGRA20_I2S_FIFO_PACKED << TEGRA20_I2S_CTRL_FIFO_FORMAT_SHIFT) + +#define TEGRA20_I2S_CTRL_IE_FIFO1_ERR (1 << 3) +#define TEGRA20_I2S_CTRL_IE_FIFO2_ERR (1 << 2) +#define TEGRA20_I2S_CTRL_QE_FIFO1 (1 << 1) +#define TEGRA20_I2S_CTRL_QE_FIFO2 (1 << 0) + +/* Fields in TEGRA20_I2S_STATUS */ + +#define TEGRA20_I2S_STATUS_FIFO1_RDY (1 << 31) +#define TEGRA20_I2S_STATUS_FIFO2_RDY (1 << 30) +#define TEGRA20_I2S_STATUS_FIFO1_BSY (1 << 29) +#define TEGRA20_I2S_STATUS_FIFO2_BSY (1 << 28) +#define TEGRA20_I2S_STATUS_FIFO1_ERR (1 << 3) +#define TEGRA20_I2S_STATUS_FIFO2_ERR (1 << 2) +#define TEGRA20_I2S_STATUS_QS_FIFO1 (1 << 1) +#define TEGRA20_I2S_STATUS_QS_FIFO2 (1 << 0) + +/* Fields in TEGRA20_I2S_TIMING */ + +#define TEGRA20_I2S_TIMING_NON_SYM_ENABLE (1 << 12) +#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 +#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff +#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) + +/* Fields in TEGRA20_I2S_FIFO_SCR */ + +#define TEGRA20_I2S_FIFO_SCR_FIFO2_FULL_EMPTY_COUNT_SHIFT 24 +#define TEGRA20_I2S_FIFO_SCR_FIFO1_FULL_EMPTY_COUNT_SHIFT 16 +#define TEGRA20_I2S_FIFO_SCR_FIFO_FULL_EMPTY_COUNT_MASK 0x3f + +#define TEGRA20_I2S_FIFO_SCR_FIFO2_CLR (1 << 12) +#define TEGRA20_I2S_FIFO_SCR_FIFO1_CLR (1 << 8) + +#define TEGRA20_I2S_FIFO_ATN_LVL_ONE_SLOT 0 +#define TEGRA20_I2S_FIFO_ATN_LVL_FOUR_SLOTS 1 +#define TEGRA20_I2S_FIFO_ATN_LVL_EIGHT_SLOTS 2 +#define TEGRA20_I2S_FIFO_ATN_LVL_TWELVE_SLOTS 3 + +#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT 4 +#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_MASK (3 << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_ONE_SLOT (TEGRA20_I2S_FIFO_ATN_LVL_ONE_SLOT << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS (TEGRA20_I2S_FIFO_ATN_LVL_FOUR_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_EIGHT_SLOTS (TEGRA20_I2S_FIFO_ATN_LVL_EIGHT_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_TWELVE_SLOTS (TEGRA20_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) + +#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT 0 +#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_MASK (3 << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_ONE_SLOT (TEGRA20_I2S_FIFO_ATN_LVL_ONE_SLOT << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS (TEGRA20_I2S_FIFO_ATN_LVL_FOUR_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_EIGHT_SLOTS (TEGRA20_I2S_FIFO_ATN_LVL_EIGHT_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_TWELVE_SLOTS (TEGRA20_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) + +struct tegra20_i2s { + struct snd_soc_dai_driver dai; + struct clk *clk_i2s; + struct tegra_pcm_dma_params capture_dma_data; + struct tegra_pcm_dma_params playback_dma_data; + struct regmap *regmap; +}; + +#endif diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c new file mode 100644 index 000000000000..3ebc8670ba00 --- /dev/null +++ b/sound/soc/tegra/tegra20_spdif.c @@ -0,0 +1,396 @@ +/* + * tegra20_spdif.c - Tegra20 SPDIF driver + * + * Author: Stephen Warren <swarren@nvidia.com> + * Copyright (C) 2011-2012 - NVIDIA, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "tegra20_spdif.h" + +#define DRV_NAME "tegra20-spdif" + +static int tegra20_spdif_runtime_suspend(struct device *dev) +{ + struct tegra20_spdif *spdif = dev_get_drvdata(dev); + + clk_disable_unprepare(spdif->clk_spdif_out); + + return 0; +} + +static int tegra20_spdif_runtime_resume(struct device *dev) +{ + struct tegra20_spdif *spdif = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(spdif->clk_spdif_out); + if (ret) { + dev_err(dev, "clk_enable failed: %d\n", ret); + return ret; + } + + return 0; +} + +static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct device *dev = dai->dev; + struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai); + unsigned int mask, val; + int ret, spdifclock; + + mask = TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + val = TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; + break; + default: + return -EINVAL; + } + + regmap_update_bits(spdif->regmap, TEGRA20_SPDIF_CTRL, mask, val); + + switch (params_rate(params)) { + case 32000: + spdifclock = 4096000; + break; + case 44100: + spdifclock = 5644800; + break; + case 48000: + spdifclock = 6144000; + break; + case 88200: + spdifclock = 11289600; + break; + case 96000: + spdifclock = 12288000; + break; + case 176400: + spdifclock = 22579200; + break; + case 192000: + spdifclock = 24576000; + break; + default: + return -EINVAL; + } + + ret = clk_set_rate(spdif->clk_spdif_out, spdifclock); + if (ret) { + dev_err(dev, "Can't set SPDIF clock rate: %d\n", ret); + return ret; + } + + return 0; +} + +static void tegra20_spdif_start_playback(struct tegra20_spdif *spdif) +{ + regmap_update_bits(spdif->regmap, TEGRA20_SPDIF_CTRL, + TEGRA20_SPDIF_CTRL_TX_EN, + TEGRA20_SPDIF_CTRL_TX_EN); +} + +static void tegra20_spdif_stop_playback(struct tegra20_spdif *spdif) +{ + regmap_update_bits(spdif->regmap, TEGRA20_SPDIF_CTRL, + TEGRA20_SPDIF_CTRL_TX_EN, 0); +} + +static int tegra20_spdif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + tegra20_spdif_start_playback(spdif); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + tegra20_spdif_stop_playback(spdif); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int tegra20_spdif_probe(struct snd_soc_dai *dai) +{ + struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai); + + dai->capture_dma_data = NULL; + dai->playback_dma_data = &spdif->playback_dma_data; + + return 0; +} + +static const struct snd_soc_dai_ops tegra20_spdif_dai_ops = { + .hw_params = tegra20_spdif_hw_params, + .trigger = tegra20_spdif_trigger, +}; + +static struct snd_soc_dai_driver tegra20_spdif_dai = { + .name = DRV_NAME, + .probe = tegra20_spdif_probe, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &tegra20_spdif_dai_ops, +}; + +static bool tegra20_spdif_wr_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_SPDIF_CTRL: + case TEGRA20_SPDIF_STATUS: + case TEGRA20_SPDIF_STROBE_CTRL: + case TEGRA20_SPDIF_DATA_FIFO_CSR: + case TEGRA20_SPDIF_DATA_OUT: + case TEGRA20_SPDIF_DATA_IN: + case TEGRA20_SPDIF_CH_STA_RX_A: + case TEGRA20_SPDIF_CH_STA_RX_B: + case TEGRA20_SPDIF_CH_STA_RX_C: + case TEGRA20_SPDIF_CH_STA_RX_D: + case TEGRA20_SPDIF_CH_STA_RX_E: + case TEGRA20_SPDIF_CH_STA_RX_F: + case TEGRA20_SPDIF_CH_STA_TX_A: + case TEGRA20_SPDIF_CH_STA_TX_B: + case TEGRA20_SPDIF_CH_STA_TX_C: + case TEGRA20_SPDIF_CH_STA_TX_D: + case TEGRA20_SPDIF_CH_STA_TX_E: + case TEGRA20_SPDIF_CH_STA_TX_F: + case TEGRA20_SPDIF_USR_STA_RX_A: + case TEGRA20_SPDIF_USR_DAT_TX_A: + return true; + default: + return false; + }; +} + +static bool tegra20_spdif_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_SPDIF_STATUS: + case TEGRA20_SPDIF_DATA_FIFO_CSR: + case TEGRA20_SPDIF_DATA_OUT: + case TEGRA20_SPDIF_DATA_IN: + case TEGRA20_SPDIF_CH_STA_RX_A: + case TEGRA20_SPDIF_CH_STA_RX_B: + case TEGRA20_SPDIF_CH_STA_RX_C: + case TEGRA20_SPDIF_CH_STA_RX_D: + case TEGRA20_SPDIF_CH_STA_RX_E: + case TEGRA20_SPDIF_CH_STA_RX_F: + case TEGRA20_SPDIF_USR_STA_RX_A: + case TEGRA20_SPDIF_USR_DAT_TX_A: + return true; + default: + return false; + }; +} + +static bool tegra20_spdif_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA20_SPDIF_DATA_OUT: + case TEGRA20_SPDIF_DATA_IN: + case TEGRA20_SPDIF_USR_STA_RX_A: + case TEGRA20_SPDIF_USR_DAT_TX_A: + return true; + default: + return false; + }; +} + +static const struct regmap_config tegra20_spdif_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = TEGRA20_SPDIF_USR_DAT_TX_A, + .writeable_reg = tegra20_spdif_wr_rd_reg, + .readable_reg = tegra20_spdif_wr_rd_reg, + .volatile_reg = tegra20_spdif_volatile_reg, + .precious_reg = tegra20_spdif_precious_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static __devinit int tegra20_spdif_platform_probe(struct platform_device *pdev) +{ + struct tegra20_spdif *spdif; + struct resource *mem, *memregion, *dmareq; + void __iomem *regs; + int ret; + + spdif = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_spdif), + GFP_KERNEL); + if (!spdif) { + dev_err(&pdev->dev, "Can't allocate tegra20_spdif\n"); + ret = -ENOMEM; + goto err; + } + dev_set_drvdata(&pdev->dev, spdif); + + spdif->clk_spdif_out = clk_get(&pdev->dev, "spdif_out"); + if (IS_ERR(spdif->clk_spdif_out)) { + pr_err("Can't retrieve spdif clock\n"); + ret = PTR_ERR(spdif->clk_spdif_out); + goto err; + } + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "No memory resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmareq) { + dev_err(&pdev->dev, "No DMA resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + memregion = devm_request_mem_region(&pdev->dev, mem->start, + resource_size(mem), DRV_NAME); + if (!memregion) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + ret = -EBUSY; + goto err_clk_put; + } + + regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); + if (!regs) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err_clk_put; + } + + spdif->regmap = devm_regmap_init_mmio(&pdev->dev, regs, + &tegra20_spdif_regmap_config); + if (IS_ERR(spdif->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + ret = PTR_ERR(spdif->regmap); + goto err_clk_put; + } + + spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT; + spdif->playback_dma_data.wrap = 4; + spdif->playback_dma_data.width = 32; + spdif->playback_dma_data.req_sel = dmareq->start; + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = tegra20_spdif_runtime_resume(&pdev->dev); + if (ret) + goto err_pm_disable; + } + + ret = snd_soc_register_dai(&pdev->dev, &tegra20_spdif_dai); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + ret = -ENOMEM; + goto err_suspend; + } + + ret = tegra_pcm_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + goto err_unregister_dai; + } + + return 0; + +err_unregister_dai: + snd_soc_unregister_dai(&pdev->dev); +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra20_spdif_runtime_suspend(&pdev->dev); +err_pm_disable: + pm_runtime_disable(&pdev->dev); +err_clk_put: + clk_put(spdif->clk_spdif_out); +err: + return ret; +} + +static int __devexit tegra20_spdif_platform_remove(struct platform_device *pdev) +{ + struct tegra20_spdif *spdif = dev_get_drvdata(&pdev->dev); + + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra20_spdif_runtime_suspend(&pdev->dev); + + tegra_pcm_platform_unregister(&pdev->dev); + snd_soc_unregister_dai(&pdev->dev); + + clk_put(spdif->clk_spdif_out); + + return 0; +} + +static const struct dev_pm_ops tegra20_spdif_pm_ops __devinitconst = { + SET_RUNTIME_PM_OPS(tegra20_spdif_runtime_suspend, + tegra20_spdif_runtime_resume, NULL) +}; + +static struct platform_driver tegra20_spdif_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &tegra20_spdif_pm_ops, + }, + .probe = tegra20_spdif_platform_probe, + .remove = __devexit_p(tegra20_spdif_platform_remove), +}; + +module_platform_driver(tegra20_spdif_driver); + +MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); +MODULE_DESCRIPTION("Tegra20 SPDIF ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra20_spdif.h b/sound/soc/tegra/tegra20_spdif.h new file mode 100644 index 000000000000..b48d699fd583 --- /dev/null +++ b/sound/soc/tegra/tegra20_spdif.h @@ -0,0 +1,470 @@ +/* + * tegra20_spdif.h - Definitions for Tegra20 SPDIF driver + * + * Author: Stephen Warren <swarren@nvidia.com> + * Copyright (C) 2011 - NVIDIA, Inc. + * + * Based on code copyright/by: + * Copyright (c) 2008-2009, NVIDIA Corporation + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TEGRA20_SPDIF_H__ +#define __TEGRA20_SPDIF_H__ + +#include "tegra_pcm.h" + +/* Offsets from TEGRA20_SPDIF_BASE */ + +#define TEGRA20_SPDIF_CTRL 0x0 +#define TEGRA20_SPDIF_STATUS 0x4 +#define TEGRA20_SPDIF_STROBE_CTRL 0x8 +#define TEGRA20_SPDIF_DATA_FIFO_CSR 0x0C +#define TEGRA20_SPDIF_DATA_OUT 0x40 +#define TEGRA20_SPDIF_DATA_IN 0x80 +#define TEGRA20_SPDIF_CH_STA_RX_A 0x100 +#define TEGRA20_SPDIF_CH_STA_RX_B 0x104 +#define TEGRA20_SPDIF_CH_STA_RX_C 0x108 +#define TEGRA20_SPDIF_CH_STA_RX_D 0x10C +#define TEGRA20_SPDIF_CH_STA_RX_E 0x110 +#define TEGRA20_SPDIF_CH_STA_RX_F 0x114 +#define TEGRA20_SPDIF_CH_STA_TX_A 0x140 +#define TEGRA20_SPDIF_CH_STA_TX_B 0x144 +#define TEGRA20_SPDIF_CH_STA_TX_C 0x148 +#define TEGRA20_SPDIF_CH_STA_TX_D 0x14C +#define TEGRA20_SPDIF_CH_STA_TX_E 0x150 +#define TEGRA20_SPDIF_CH_STA_TX_F 0x154 +#define TEGRA20_SPDIF_USR_STA_RX_A 0x180 +#define TEGRA20_SPDIF_USR_DAT_TX_A 0x1C0 + +/* Fields in TEGRA20_SPDIF_CTRL */ + +/* Start capturing from 0=right, 1=left channel */ +#define TEGRA20_SPDIF_CTRL_CAP_LC (1 << 30) + +/* SPDIF receiver(RX) enable */ +#define TEGRA20_SPDIF_CTRL_RX_EN (1 << 29) + +/* SPDIF Transmitter(TX) enable */ +#define TEGRA20_SPDIF_CTRL_TX_EN (1 << 28) + +/* Transmit Channel status */ +#define TEGRA20_SPDIF_CTRL_TC_EN (1 << 27) + +/* Transmit user Data */ +#define TEGRA20_SPDIF_CTRL_TU_EN (1 << 26) + +/* Interrupt on transmit error */ +#define TEGRA20_SPDIF_CTRL_IE_TXE (1 << 25) + +/* Interrupt on receive error */ +#define TEGRA20_SPDIF_CTRL_IE_RXE (1 << 24) + +/* Interrupt on invalid preamble */ +#define TEGRA20_SPDIF_CTRL_IE_P (1 << 23) + +/* Interrupt on "B" preamble */ +#define TEGRA20_SPDIF_CTRL_IE_B (1 << 22) + +/* Interrupt when block of channel status received */ +#define TEGRA20_SPDIF_CTRL_IE_C (1 << 21) + +/* Interrupt when a valid information unit (IU) is received */ +#define TEGRA20_SPDIF_CTRL_IE_U (1 << 20) + +/* Interrupt when RX user FIFO attention level is reached */ +#define TEGRA20_SPDIF_CTRL_QE_RU (1 << 19) + +/* Interrupt when TX user FIFO attention level is reached */ +#define TEGRA20_SPDIF_CTRL_QE_TU (1 << 18) + +/* Interrupt when RX data FIFO attention level is reached */ +#define TEGRA20_SPDIF_CTRL_QE_RX (1 << 17) + +/* Interrupt when TX data FIFO attention level is reached */ +#define TEGRA20_SPDIF_CTRL_QE_TX (1 << 16) + +/* Loopback test mode enable */ +#define TEGRA20_SPDIF_CTRL_LBK_EN (1 << 15) + +/* + * Pack data mode: + * 0 = Single data (16 bit needs to be padded to match the + * interface data bit size). + * 1 = Packeted left/right channel data into a single word. + */ +#define TEGRA20_SPDIF_CTRL_PACK (1 << 14) + +/* + * 00 = 16bit data + * 01 = 20bit data + * 10 = 24bit data + * 11 = raw data + */ +#define TEGRA20_SPDIF_BIT_MODE_16BIT 0 +#define TEGRA20_SPDIF_BIT_MODE_20BIT 1 +#define TEGRA20_SPDIF_BIT_MODE_24BIT 2 +#define TEGRA20_SPDIF_BIT_MODE_RAW 3 + +#define TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT 12 +#define TEGRA20_SPDIF_CTRL_BIT_MODE_MASK (3 << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT (TEGRA20_SPDIF_BIT_MODE_16BIT << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA20_SPDIF_CTRL_BIT_MODE_20BIT (TEGRA20_SPDIF_BIT_MODE_20BIT << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA20_SPDIF_CTRL_BIT_MODE_24BIT (TEGRA20_SPDIF_BIT_MODE_24BIT << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA20_SPDIF_CTRL_BIT_MODE_RAW (TEGRA20_SPDIF_BIT_MODE_RAW << TEGRA20_SPDIF_CTRL_BIT_MODE_SHIFT) + +/* Fields in TEGRA20_SPDIF_STATUS */ + +/* + * Note: IS_P, IS_B, IS_C, and IS_U are sticky bits. Software must + * write a 1 to the corresponding bit location to clear the status. + */ + +/* + * Receiver(RX) shifter is busy receiving data. + * This bit is asserted when the receiver first locked onto the + * preamble of the data stream after RX_EN is asserted. This bit is + * deasserted when either, + * (a) the end of a frame is reached after RX_EN is deeasserted, or + * (b) the SPDIF data stream becomes inactive. + */ +#define TEGRA20_SPDIF_STATUS_RX_BSY (1 << 29) + +/* + * Transmitter(TX) shifter is busy transmitting data. + * This bit is asserted when TX_EN is asserted. + * This bit is deasserted when the end of a frame is reached after + * TX_EN is deasserted. + */ +#define TEGRA20_SPDIF_STATUS_TX_BSY (1 << 28) + +/* + * TX is busy shifting out channel status. + * This bit is asserted when both TX_EN and TC_EN are asserted and + * data from CH_STA_TX_A register is loaded into the internal shifter. + * This bit is deasserted when either, + * (a) the end of a frame is reached after TX_EN is deasserted, or + * (b) CH_STA_TX_F register is loaded into the internal shifter. + */ +#define TEGRA20_SPDIF_STATUS_TC_BSY (1 << 27) + +/* + * TX User data FIFO busy. + * This bit is asserted when TX_EN and TXU_EN are asserted and + * there's data in the TX user FIFO. This bit is deassert when either, + * (a) the end of a frame is reached after TX_EN is deasserted, or + * (b) there's no data left in the TX user FIFO. + */ +#define TEGRA20_SPDIF_STATUS_TU_BSY (1 << 26) + +/* TX FIFO Underrun error status */ +#define TEGRA20_SPDIF_STATUS_TX_ERR (1 << 25) + +/* RX FIFO Overrun error status */ +#define TEGRA20_SPDIF_STATUS_RX_ERR (1 << 24) + +/* Preamble status: 0=Preamble OK, 1=bad/missing preamble */ +#define TEGRA20_SPDIF_STATUS_IS_P (1 << 23) + +/* B-preamble detection status: 0=not detected, 1=B-preamble detected */ +#define TEGRA20_SPDIF_STATUS_IS_B (1 << 22) + +/* + * RX channel block data receive status: + * 0=entire block not recieved yet. + * 1=received entire block of channel status, + */ +#define TEGRA20_SPDIF_STATUS_IS_C (1 << 21) + +/* RX User Data Valid flag: 1=valid IU detected, 0 = no IU detected. */ +#define TEGRA20_SPDIF_STATUS_IS_U (1 << 20) + +/* + * RX User FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA20_SPDIF_STATUS_QS_RU (1 << 19) + +/* + * TX User FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA20_SPDIF_STATUS_QS_TU (1 << 18) + +/* + * RX Data FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA20_SPDIF_STATUS_QS_RX (1 << 17) + +/* + * TX Data FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA20_SPDIF_STATUS_QS_TX (1 << 16) + +/* Fields in TEGRA20_SPDIF_STROBE_CTRL */ + +/* + * Indicates the approximate number of detected SPDIFIN clocks within a + * bi-phase period. + */ +#define TEGRA20_SPDIF_STROBE_CTRL_PERIOD_SHIFT 16 +#define TEGRA20_SPDIF_STROBE_CTRL_PERIOD_MASK (0xff << TEGRA20_SPDIF_STROBE_CTRL_PERIOD_SHIFT) + +/* Data strobe mode: 0=Auto-locked 1=Manual locked */ +#define TEGRA20_SPDIF_STROBE_CTRL_STROBE (1 << 15) + +/* + * Manual data strobe time within the bi-phase clock period (in terms of + * the number of over-sampling clocks). + */ +#define TEGRA20_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT 8 +#define TEGRA20_SPDIF_STROBE_CTRL_DATA_STROBES_MASK (0x1f << TEGRA20_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT) + +/* + * Manual SPDIFIN bi-phase clock period (in terms of the number of + * over-sampling clocks). + */ +#define TEGRA20_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT 0 +#define TEGRA20_SPDIF_STROBE_CTRL_CLOCK_PERIOD_MASK (0x3f << TEGRA20_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT) + +/* Fields in SPDIF_DATA_FIFO_CSR */ + +/* Clear Receiver User FIFO (RX USR.FIFO) */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_CLR (1 << 31) + +#define TEGRA20_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT 0 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS 1 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS 2 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS 3 + +/* RU FIFO attention level */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT 29 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_MASK \ + (0x3 << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU1_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU2_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU3_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU4_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) + +/* Number of RX USR.FIFO levels with valid data. */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT 24 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_MASK (0x1f << TEGRA20_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT) + +/* Clear Transmitter User FIFO (TX USR.FIFO) */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_CLR (1 << 23) + +/* TU FIFO attention level */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT 21 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_MASK \ + (0x3 << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU1_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU2_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU3_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU4_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) + +/* Number of TX USR.FIFO levels that could be filled. */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT 16 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT) + +/* Clear Receiver Data FIFO (RX DATA.FIFO) */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_CLR (1 << 15) + +#define TEGRA20_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT 0 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS 1 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS 2 +#define TEGRA20_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS 3 + +/* RU FIFO attention level */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT 13 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_MASK \ + (0x3 << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU1_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU4_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU8_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU12_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) + +/* Number of RX DATA.FIFO levels with valid data. */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT 8 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_MASK (0x1f << TEGRA20_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT) + +/* Clear Transmitter Data FIFO (TX DATA.FIFO) */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_CLR (1 << 7) + +/* TU FIFO attention level */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT 5 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_MASK \ + (0x3 << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU1_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU4_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU8_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU12_WORD_FULL \ + (TEGRA20_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA20_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) + +/* Number of TX DATA.FIFO levels that could be filled. */ +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT 0 +#define TEGRA20_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT) + +/* Fields in TEGRA20_SPDIF_DATA_OUT */ + +/* + * This register has 5 different formats: + * 16-bit (BIT_MODE=00, PACK=0) + * 20-bit (BIT_MODE=01, PACK=0) + * 24-bit (BIT_MODE=10, PACK=0) + * raw (BIT_MODE=11, PACK=0) + * 16-bit packed (BIT_MODE=00, PACK=1) + */ + +#define TEGRA20_SPDIF_DATA_OUT_DATA_16_SHIFT 0 +#define TEGRA20_SPDIF_DATA_OUT_DATA_16_MASK (0xffff << TEGRA20_SPDIF_DATA_OUT_DATA_16_SHIFT) + +#define TEGRA20_SPDIF_DATA_OUT_DATA_20_SHIFT 0 +#define TEGRA20_SPDIF_DATA_OUT_DATA_20_MASK (0xfffff << TEGRA20_SPDIF_DATA_OUT_DATA_20_SHIFT) + +#define TEGRA20_SPDIF_DATA_OUT_DATA_24_SHIFT 0 +#define TEGRA20_SPDIF_DATA_OUT_DATA_24_MASK (0xffffff << TEGRA20_SPDIF_DATA_OUT_DATA_24_SHIFT) + +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_P (1 << 31) +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_C (1 << 30) +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_U (1 << 29) +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_V (1 << 28) + +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT 8 +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_DATA_MASK (0xfffff << TEGRA20_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT) + +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT 4 +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_AUX_MASK (0xf << TEGRA20_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT) + +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT 0 +#define TEGRA20_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA20_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT) + +#define TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT 16 +#define TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT) + +#define TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT 0 +#define TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA20_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT) + +/* Fields in TEGRA20_SPDIF_DATA_IN */ + +/* + * This register has 5 different formats: + * 16-bit (BIT_MODE=00, PACK=0) + * 20-bit (BIT_MODE=01, PACK=0) + * 24-bit (BIT_MODE=10, PACK=0) + * raw (BIT_MODE=11, PACK=0) + * 16-bit packed (BIT_MODE=00, PACK=1) + * + * Bits 31:24 are common to all modes except 16-bit packed + */ + +#define TEGRA20_SPDIF_DATA_IN_DATA_P (1 << 31) +#define TEGRA20_SPDIF_DATA_IN_DATA_C (1 << 30) +#define TEGRA20_SPDIF_DATA_IN_DATA_U (1 << 29) +#define TEGRA20_SPDIF_DATA_IN_DATA_V (1 << 28) + +#define TEGRA20_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT 24 +#define TEGRA20_SPDIF_DATA_IN_DATA_PREAMBLE_MASK (0xf << TEGRA20_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT) + +#define TEGRA20_SPDIF_DATA_IN_DATA_16_SHIFT 0 +#define TEGRA20_SPDIF_DATA_IN_DATA_16_MASK (0xffff << TEGRA20_SPDIF_DATA_IN_DATA_16_SHIFT) + +#define TEGRA20_SPDIF_DATA_IN_DATA_20_SHIFT 0 +#define TEGRA20_SPDIF_DATA_IN_DATA_20_MASK (0xfffff << TEGRA20_SPDIF_DATA_IN_DATA_20_SHIFT) + +#define TEGRA20_SPDIF_DATA_IN_DATA_24_SHIFT 0 +#define TEGRA20_SPDIF_DATA_IN_DATA_24_MASK (0xffffff << TEGRA20_SPDIF_DATA_IN_DATA_24_SHIFT) + +#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT 8 +#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_DATA_MASK (0xfffff << TEGRA20_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT) + +#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT 4 +#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_AUX_MASK (0xf << TEGRA20_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT) + +#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT 0 +#define TEGRA20_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA20_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT) + +#define TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT 16 +#define TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT) + +#define TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT 0 +#define TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA20_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT) + +/* Fields in TEGRA20_SPDIF_CH_STA_RX_A */ +/* Fields in TEGRA20_SPDIF_CH_STA_RX_B */ +/* Fields in TEGRA20_SPDIF_CH_STA_RX_C */ +/* Fields in TEGRA20_SPDIF_CH_STA_RX_D */ +/* Fields in TEGRA20_SPDIF_CH_STA_RX_E */ +/* Fields in TEGRA20_SPDIF_CH_STA_RX_F */ + +/* + * The 6-word receive channel data page buffer holds a block (192 frames) of + * channel status information. The order of receive is from LSB to MSB + * bit, and from CH_STA_RX_A to CH_STA_RX_F then back to CH_STA_RX_A. + */ + +/* Fields in TEGRA20_SPDIF_CH_STA_TX_A */ +/* Fields in TEGRA20_SPDIF_CH_STA_TX_B */ +/* Fields in TEGRA20_SPDIF_CH_STA_TX_C */ +/* Fields in TEGRA20_SPDIF_CH_STA_TX_D */ +/* Fields in TEGRA20_SPDIF_CH_STA_TX_E */ +/* Fields in TEGRA20_SPDIF_CH_STA_TX_F */ + +/* + * The 6-word transmit channel data page buffer holds a block (192 frames) of + * channel status information. The order of transmission is from LSB to MSB + * bit, and from CH_STA_TX_A to CH_STA_TX_F then back to CH_STA_TX_A. + */ + +/* Fields in TEGRA20_SPDIF_USR_STA_RX_A */ + +/* + * This 4-word deep FIFO receives user FIFO field information. The order of + * receive is from LSB to MSB bit. + */ + +/* Fields in TEGRA20_SPDIF_USR_DAT_TX_A */ + +/* + * This 4-word deep FIFO transmits user FIFO field information. The order of + * transmission is from LSB to MSB bit. + */ + +struct tegra20_spdif { + struct clk *clk_spdif_out; + struct tegra_pcm_dma_params capture_dma_data; + struct tegra_pcm_dma_params playback_dma_data; + struct regmap *regmap; +}; + +#endif diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c new file mode 100644 index 000000000000..bf5610122c76 --- /dev/null +++ b/sound/soc/tegra/tegra30_ahub.c @@ -0,0 +1,632 @@ +/* + * tegra30_ahub.c - Tegra30 AHUB driver + * + * Copyright (c) 2011,2012, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <mach/clk.h> +#include <mach/dma.h> +#include <sound/soc.h> +#include "tegra30_ahub.h" + +#define DRV_NAME "tegra30-ahub" + +static struct tegra30_ahub *ahub; + +static inline void tegra30_apbif_write(u32 reg, u32 val) +{ + regmap_write(ahub->regmap_apbif, reg, val); +} + +static inline u32 tegra30_apbif_read(u32 reg) +{ + u32 val; + regmap_read(ahub->regmap_apbif, reg, &val); + return val; +} + +static inline void tegra30_audio_write(u32 reg, u32 val) +{ + regmap_write(ahub->regmap_ahub, reg, val); +} + +static int tegra30_ahub_runtime_suspend(struct device *dev) +{ + regcache_cache_only(ahub->regmap_apbif, true); + regcache_cache_only(ahub->regmap_ahub, true); + + clk_disable_unprepare(ahub->clk_apbif); + clk_disable_unprepare(ahub->clk_d_audio); + + return 0; +} + +/* + * clk_apbif isn't required for an I2S<->I2S configuration where no PCM data + * is read from or sent to memory. However, that's not something the rest of + * the driver supports right now, so we'll just treat the two clocks as one + * for now. + * + * These functions should not be a plain ref-count. Instead, each active stream + * contributes some requirement to the minimum clock rate, so starting or + * stopping streams should dynamically adjust the clock as required. However, + * this is not yet implemented. + */ +static int tegra30_ahub_runtime_resume(struct device *dev) +{ + int ret; + + ret = clk_prepare_enable(ahub->clk_d_audio); + if (ret) { + dev_err(dev, "clk_enable d_audio failed: %d\n", ret); + return ret; + } + ret = clk_prepare_enable(ahub->clk_apbif); + if (ret) { + dev_err(dev, "clk_enable apbif failed: %d\n", ret); + clk_disable(ahub->clk_d_audio); + return ret; + } + + regcache_cache_only(ahub->regmap_apbif, false); + regcache_cache_only(ahub->regmap_ahub, false); + + return 0; +} + +int tegra30_ahub_allocate_rx_fifo(enum tegra30_ahub_rxcif *rxcif, + unsigned long *fiforeg, + unsigned long *reqsel) +{ + int channel; + u32 reg, val; + + channel = find_first_zero_bit(ahub->rx_usage, + TEGRA30_AHUB_CHANNEL_CTRL_COUNT); + if (channel >= TEGRA30_AHUB_CHANNEL_CTRL_COUNT) + return -EBUSY; + + __set_bit(channel, ahub->rx_usage); + + *rxcif = TEGRA30_AHUB_RXCIF_APBIF_RX0 + channel; + *fiforeg = ahub->apbif_addr + TEGRA30_AHUB_CHANNEL_RXFIFO + + (channel * TEGRA30_AHUB_CHANNEL_RXFIFO_STRIDE); + *reqsel = ahub->dma_sel + channel; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val &= ~(TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK | + TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK); + val |= (7 << TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_SHIFT) | + TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_EN | + TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_16; + tegra30_apbif_write(reg, val); + + reg = TEGRA30_AHUB_CIF_RX_CTRL + + (channel * TEGRA30_AHUB_CIF_RX_CTRL_STRIDE); + val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | + TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | + TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 | + TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_allocate_rx_fifo); + +int tegra30_ahub_enable_rx_fifo(enum tegra30_ahub_rxcif rxcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + int reg, val; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val |= TEGRA30_AHUB_CHANNEL_CTRL_RX_EN; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_enable_rx_fifo); + +int tegra30_ahub_disable_rx_fifo(enum tegra30_ahub_rxcif rxcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + int reg, val; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val &= ~TEGRA30_AHUB_CHANNEL_CTRL_RX_EN; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_disable_rx_fifo); + +int tegra30_ahub_free_rx_fifo(enum tegra30_ahub_rxcif rxcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + + __clear_bit(channel, ahub->rx_usage); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_free_rx_fifo); + +int tegra30_ahub_allocate_tx_fifo(enum tegra30_ahub_txcif *txcif, + unsigned long *fiforeg, + unsigned long *reqsel) +{ + int channel; + u32 reg, val; + + channel = find_first_zero_bit(ahub->tx_usage, + TEGRA30_AHUB_CHANNEL_CTRL_COUNT); + if (channel >= TEGRA30_AHUB_CHANNEL_CTRL_COUNT) + return -EBUSY; + + __set_bit(channel, ahub->tx_usage); + + *txcif = TEGRA30_AHUB_TXCIF_APBIF_TX0 + channel; + *fiforeg = ahub->apbif_addr + TEGRA30_AHUB_CHANNEL_TXFIFO + + (channel * TEGRA30_AHUB_CHANNEL_TXFIFO_STRIDE); + *reqsel = ahub->dma_sel + channel; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val &= ~(TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK | + TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK); + val |= (7 << TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_SHIFT) | + TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_EN | + TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_16; + tegra30_apbif_write(reg, val); + + reg = TEGRA30_AHUB_CIF_TX_CTRL + + (channel * TEGRA30_AHUB_CIF_TX_CTRL_STRIDE); + val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | + TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | + TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 | + TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_allocate_tx_fifo); + +int tegra30_ahub_enable_tx_fifo(enum tegra30_ahub_txcif txcif) +{ + int channel = txcif - TEGRA30_AHUB_TXCIF_APBIF_TX0; + int reg, val; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val |= TEGRA30_AHUB_CHANNEL_CTRL_TX_EN; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_enable_tx_fifo); + +int tegra30_ahub_disable_tx_fifo(enum tegra30_ahub_txcif txcif) +{ + int channel = txcif - TEGRA30_AHUB_TXCIF_APBIF_TX0; + int reg, val; + + reg = TEGRA30_AHUB_CHANNEL_CTRL + + (channel * TEGRA30_AHUB_CHANNEL_CTRL_STRIDE); + val = tegra30_apbif_read(reg); + val &= ~TEGRA30_AHUB_CHANNEL_CTRL_TX_EN; + tegra30_apbif_write(reg, val); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_disable_tx_fifo); + +int tegra30_ahub_free_tx_fifo(enum tegra30_ahub_txcif txcif) +{ + int channel = txcif - TEGRA30_AHUB_TXCIF_APBIF_TX0; + + __clear_bit(channel, ahub->tx_usage); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_free_tx_fifo); + +int tegra30_ahub_set_rx_cif_source(enum tegra30_ahub_rxcif rxcif, + enum tegra30_ahub_txcif txcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + int reg; + + reg = TEGRA30_AHUB_AUDIO_RX + + (channel * TEGRA30_AHUB_AUDIO_RX_STRIDE); + tegra30_audio_write(reg, 1 << txcif); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_set_rx_cif_source); + +int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif) +{ + int channel = rxcif - TEGRA30_AHUB_RXCIF_APBIF_RX0; + int reg; + + reg = TEGRA30_AHUB_AUDIO_RX + + (channel * TEGRA30_AHUB_AUDIO_RX_STRIDE); + tegra30_audio_write(reg, 0); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra30_ahub_unset_rx_cif_source); + +static const char * const configlink_clocks[] __devinitconst = { + "i2s0", + "i2s1", + "i2s2", + "i2s3", + "i2s4", + "dam0", + "dam1", + "dam2", + "spdif_in", +}; + +struct of_dev_auxdata ahub_auxdata[] __devinitdata = { + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080300, "tegra30-i2s.0", NULL), + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080400, "tegra30-i2s.1", NULL), + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080500, "tegra30-i2s.2", NULL), + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080600, "tegra30-i2s.3", NULL), + OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080700, "tegra30-i2s.4", NULL), + {} +}; + +#define LAST_REG(name) \ + (TEGRA30_AHUB_##name + \ + (TEGRA30_AHUB_##name##_STRIDE * TEGRA30_AHUB_##name##_COUNT) - 4) + +#define REG_IN_ARRAY(reg, name) \ + ((reg >= TEGRA30_AHUB_##name) && \ + (reg <= LAST_REG(name) && \ + (!((reg - TEGRA30_AHUB_##name) % TEGRA30_AHUB_##name##_STRIDE)))) + +static bool tegra30_ahub_apbif_wr_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA30_AHUB_CONFIG_LINK_CTRL: + case TEGRA30_AHUB_MISC_CTRL: + case TEGRA30_AHUB_APBDMA_LIVE_STATUS: + case TEGRA30_AHUB_I2S_LIVE_STATUS: + case TEGRA30_AHUB_SPDIF_LIVE_STATUS: + case TEGRA30_AHUB_I2S_INT_MASK: + case TEGRA30_AHUB_DAM_INT_MASK: + case TEGRA30_AHUB_SPDIF_INT_MASK: + case TEGRA30_AHUB_APBIF_INT_MASK: + case TEGRA30_AHUB_I2S_INT_STATUS: + case TEGRA30_AHUB_DAM_INT_STATUS: + case TEGRA30_AHUB_SPDIF_INT_STATUS: + case TEGRA30_AHUB_APBIF_INT_STATUS: + case TEGRA30_AHUB_I2S_INT_SOURCE: + case TEGRA30_AHUB_DAM_INT_SOURCE: + case TEGRA30_AHUB_SPDIF_INT_SOURCE: + case TEGRA30_AHUB_APBIF_INT_SOURCE: + case TEGRA30_AHUB_I2S_INT_SET: + case TEGRA30_AHUB_DAM_INT_SET: + case TEGRA30_AHUB_SPDIF_INT_SET: + case TEGRA30_AHUB_APBIF_INT_SET: + return true; + default: + break; + }; + + if (REG_IN_ARRAY(reg, CHANNEL_CTRL) || + REG_IN_ARRAY(reg, CHANNEL_CLEAR) || + REG_IN_ARRAY(reg, CHANNEL_STATUS) || + REG_IN_ARRAY(reg, CHANNEL_TXFIFO) || + REG_IN_ARRAY(reg, CHANNEL_RXFIFO) || + REG_IN_ARRAY(reg, CIF_TX_CTRL) || + REG_IN_ARRAY(reg, CIF_RX_CTRL) || + REG_IN_ARRAY(reg, DAM_LIVE_STATUS)) + return true; + + return false; +} + +static bool tegra30_ahub_apbif_volatile_reg(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case TEGRA30_AHUB_CONFIG_LINK_CTRL: + case TEGRA30_AHUB_MISC_CTRL: + case TEGRA30_AHUB_APBDMA_LIVE_STATUS: + case TEGRA30_AHUB_I2S_LIVE_STATUS: + case TEGRA30_AHUB_SPDIF_LIVE_STATUS: + case TEGRA30_AHUB_I2S_INT_STATUS: + case TEGRA30_AHUB_DAM_INT_STATUS: + case TEGRA30_AHUB_SPDIF_INT_STATUS: + case TEGRA30_AHUB_APBIF_INT_STATUS: + case TEGRA30_AHUB_I2S_INT_SET: + case TEGRA30_AHUB_DAM_INT_SET: + case TEGRA30_AHUB_SPDIF_INT_SET: + case TEGRA30_AHUB_APBIF_INT_SET: + return true; + default: + break; + }; + + if (REG_IN_ARRAY(reg, CHANNEL_CLEAR) || + REG_IN_ARRAY(reg, CHANNEL_STATUS) || + REG_IN_ARRAY(reg, CHANNEL_TXFIFO) || + REG_IN_ARRAY(reg, CHANNEL_RXFIFO) || + REG_IN_ARRAY(reg, DAM_LIVE_STATUS)) + return true; + + return false; +} + +static bool tegra30_ahub_apbif_precious_reg(struct device *dev, + unsigned int reg) +{ + if (REG_IN_ARRAY(reg, CHANNEL_TXFIFO) || + REG_IN_ARRAY(reg, CHANNEL_RXFIFO)) + return true; + + return false; +} + +static const struct regmap_config tegra30_ahub_apbif_regmap_config = { + .name = "apbif", + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = TEGRA30_AHUB_APBIF_INT_SET, + .writeable_reg = tegra30_ahub_apbif_wr_rd_reg, + .readable_reg = tegra30_ahub_apbif_wr_rd_reg, + .volatile_reg = tegra30_ahub_apbif_volatile_reg, + .precious_reg = tegra30_ahub_apbif_precious_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static bool tegra30_ahub_ahub_wr_rd_reg(struct device *dev, unsigned int reg) +{ + if (REG_IN_ARRAY(reg, AUDIO_RX)) + return true; + + return false; +} + +static const struct regmap_config tegra30_ahub_ahub_regmap_config = { + .name = "ahub", + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = LAST_REG(AUDIO_RX), + .writeable_reg = tegra30_ahub_ahub_wr_rd_reg, + .readable_reg = tegra30_ahub_ahub_wr_rd_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit tegra30_ahub_probe(struct platform_device *pdev) +{ + struct clk *clk; + int i; + struct resource *res0, *res1, *region; + u32 of_dma[2]; + void __iomem *regs_apbif, *regs_ahub; + int ret = 0; + + if (ahub) + return -ENODEV; + + /* + * The AHUB hosts a register bus: the "configlink". For this to + * operate correctly, all devices on this bus must be out of reset. + * Ensure that here. + */ + for (i = 0; i < ARRAY_SIZE(configlink_clocks); i++) { + clk = clk_get_sys(NULL, configlink_clocks[i]); + if (IS_ERR(clk)) { + dev_err(&pdev->dev, "Can't get clock %s\n", + configlink_clocks[i]); + ret = PTR_ERR(clk); + goto err; + } + tegra_periph_reset_deassert(clk); + clk_put(clk); + } + + ahub = devm_kzalloc(&pdev->dev, sizeof(struct tegra30_ahub), + GFP_KERNEL); + if (!ahub) { + dev_err(&pdev->dev, "Can't allocate tegra30_ahub\n"); + ret = -ENOMEM; + goto err; + } + dev_set_drvdata(&pdev->dev, ahub); + + ahub->dev = &pdev->dev; + + ahub->clk_d_audio = clk_get(&pdev->dev, "d_audio"); + if (IS_ERR(ahub->clk_d_audio)) { + dev_err(&pdev->dev, "Can't retrieve ahub d_audio clock\n"); + ret = PTR_ERR(ahub->clk_d_audio); + goto err; + } + + ahub->clk_apbif = clk_get(&pdev->dev, "apbif"); + if (IS_ERR(ahub->clk_apbif)) { + dev_err(&pdev->dev, "Can't retrieve ahub apbif clock\n"); + ret = PTR_ERR(ahub->clk_apbif); + goto err_clk_put_d_audio; + } + + if (of_property_read_u32_array(pdev->dev.of_node, + "nvidia,dma-request-selector", + of_dma, 2) < 0) { + dev_err(&pdev->dev, + "Missing property nvidia,dma-request-selector\n"); + ret = -ENODEV; + goto err_clk_put_d_audio; + } + ahub->dma_sel = of_dma[1]; + + res0 = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res0) { + dev_err(&pdev->dev, "No apbif memory resource\n"); + ret = -ENODEV; + goto err_clk_put_apbif; + } + + region = devm_request_mem_region(&pdev->dev, res0->start, + resource_size(res0), DRV_NAME); + if (!region) { + dev_err(&pdev->dev, "request region apbif failed\n"); + ret = -EBUSY; + goto err_clk_put_apbif; + } + ahub->apbif_addr = res0->start; + + regs_apbif = devm_ioremap(&pdev->dev, res0->start, + resource_size(res0)); + if (!regs_apbif) { + dev_err(&pdev->dev, "ioremap apbif failed\n"); + ret = -ENOMEM; + goto err_clk_put_apbif; + } + + ahub->regmap_apbif = devm_regmap_init_mmio(&pdev->dev, regs_apbif, + &tegra30_ahub_apbif_regmap_config); + if (IS_ERR(ahub->regmap_apbif)) { + dev_err(&pdev->dev, "apbif regmap init failed\n"); + ret = PTR_ERR(ahub->regmap_apbif); + goto err_clk_put_apbif; + } + regcache_cache_only(ahub->regmap_apbif, true); + + res1 = platform_get_resource(pdev, IORESOURCE_MEM, 1); + if (!res1) { + dev_err(&pdev->dev, "No ahub memory resource\n"); + ret = -ENODEV; + goto err_clk_put_apbif; + } + + region = devm_request_mem_region(&pdev->dev, res1->start, + resource_size(res1), DRV_NAME); + if (!region) { + dev_err(&pdev->dev, "request region ahub failed\n"); + ret = -EBUSY; + goto err_clk_put_apbif; + } + + regs_ahub = devm_ioremap(&pdev->dev, res1->start, + resource_size(res1)); + if (!regs_ahub) { + dev_err(&pdev->dev, "ioremap ahub failed\n"); + ret = -ENOMEM; + goto err_clk_put_apbif; + } + + ahub->regmap_ahub = devm_regmap_init_mmio(&pdev->dev, regs_ahub, + &tegra30_ahub_ahub_regmap_config); + if (IS_ERR(ahub->regmap_ahub)) { + dev_err(&pdev->dev, "ahub regmap init failed\n"); + ret = PTR_ERR(ahub->regmap_ahub); + goto err_clk_put_apbif; + } + regcache_cache_only(ahub->regmap_ahub, true); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = tegra30_ahub_runtime_resume(&pdev->dev); + if (ret) + goto err_pm_disable; + } + + of_platform_populate(pdev->dev.of_node, NULL, ahub_auxdata, + &pdev->dev); + + return 0; + +err_pm_disable: + pm_runtime_disable(&pdev->dev); +err_clk_put_apbif: + clk_put(ahub->clk_apbif); +err_clk_put_d_audio: + clk_put(ahub->clk_d_audio); + ahub = 0; +err: + return ret; +} + +static int __devexit tegra30_ahub_remove(struct platform_device *pdev) +{ + if (!ahub) + return -ENODEV; + + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra30_ahub_runtime_suspend(&pdev->dev); + + clk_put(ahub->clk_apbif); + clk_put(ahub->clk_d_audio); + + ahub = 0; + + return 0; +} + +static const struct of_device_id tegra30_ahub_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra30-ahub", }, + {}, +}; + +static const struct dev_pm_ops tegra30_ahub_pm_ops __devinitconst = { + SET_RUNTIME_PM_OPS(tegra30_ahub_runtime_suspend, + tegra30_ahub_runtime_resume, NULL) +}; + +static struct platform_driver tegra30_ahub_driver = { + .probe = tegra30_ahub_probe, + .remove = __devexit_p(tegra30_ahub_remove), + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = tegra30_ahub_of_match, + .pm = &tegra30_ahub_pm_ops, + }, +}; +module_platform_driver(tegra30_ahub_driver); + +MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); +MODULE_DESCRIPTION("Tegra30 AHUB driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra30_ahub_of_match); diff --git a/sound/soc/tegra/tegra30_ahub.h b/sound/soc/tegra/tegra30_ahub.h new file mode 100644 index 000000000000..e690e2eecc92 --- /dev/null +++ b/sound/soc/tegra/tegra30_ahub.h @@ -0,0 +1,483 @@ +/* + * tegra30_ahub.h - Definitions for Tegra30 AHUB driver + * + * Copyright (c) 2011,2012, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#ifndef __TEGRA30_AHUB_H__ +#define __TEGRA30_AHUB_H__ + +/* Fields in *_CIF_RX/TX_CTRL; used by AHUB FIFOs, and all other audio modules */ + +#define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT 28 +#define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US 0xf +#define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK (TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) + +/* Channel count minus 1 */ +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT 24 +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US 7 +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK (TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) + +/* Channel count minus 1 */ +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT 16 +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US 7 +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK (TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) + +#define TEGRA30_AUDIOCIF_BITS_4 0 +#define TEGRA30_AUDIOCIF_BITS_8 1 +#define TEGRA30_AUDIOCIF_BITS_12 2 +#define TEGRA30_AUDIOCIF_BITS_16 3 +#define TEGRA30_AUDIOCIF_BITS_20 4 +#define TEGRA30_AUDIOCIF_BITS_24 5 +#define TEGRA30_AUDIOCIF_BITS_28 6 +#define TEGRA30_AUDIOCIF_BITS_32 7 + +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT 12 +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_MASK (7 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_4 (TEGRA30_AUDIOCIF_BITS_4 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_8 (TEGRA30_AUDIOCIF_BITS_8 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_12 (TEGRA30_AUDIOCIF_BITS_12 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 (TEGRA30_AUDIOCIF_BITS_16 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_20 (TEGRA30_AUDIOCIF_BITS_20 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_24 (TEGRA30_AUDIOCIF_BITS_24 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_28 (TEGRA30_AUDIOCIF_BITS_28 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_32 (TEGRA30_AUDIOCIF_BITS_32 << TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) + +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT 8 +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_MASK (7 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_4 (TEGRA30_AUDIOCIF_BITS_4 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_8 (TEGRA30_AUDIOCIF_BITS_8 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_12 (TEGRA30_AUDIOCIF_BITS_12 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 (TEGRA30_AUDIOCIF_BITS_16 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_20 (TEGRA30_AUDIOCIF_BITS_20 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_24 (TEGRA30_AUDIOCIF_BITS_24 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_28 (TEGRA30_AUDIOCIF_BITS_28 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_32 (TEGRA30_AUDIOCIF_BITS_32 << TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) + +#define TEGRA30_AUDIOCIF_EXPAND_ZERO 0 +#define TEGRA30_AUDIOCIF_EXPAND_ONE 1 +#define TEGRA30_AUDIOCIF_EXPAND_LFSR 2 + +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT 6 +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_MASK (3 << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_ZERO (TEGRA30_AUDIOCIF_EXPAND_ZERO << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_ONE (TEGRA30_AUDIOCIF_EXPAND_ONE << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_EXPAND_LFSR (TEGRA30_AUDIOCIF_EXPAND_LFSR << TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) + +#define TEGRA30_AUDIOCIF_STEREO_CONV_CH0 0 +#define TEGRA30_AUDIOCIF_STEREO_CONV_CH1 1 +#define TEGRA30_AUDIOCIF_STEREO_CONV_AVG 2 + +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT 4 +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_MASK (3 << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_CH0 (TEGRA30_AUDIOCIF_STEREO_CONV_CH0 << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_CH1 (TEGRA30_AUDIOCIF_STEREO_CONV_CH1 << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_AVG (TEGRA30_AUDIOCIF_STEREO_CONV_AVG << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) + +#define TEGRA30_AUDIOCIF_CTRL_REPLICATE 3 + +#define TEGRA30_AUDIOCIF_DIRECTION_TX 0 +#define TEGRA30_AUDIOCIF_DIRECTION_RX 1 + +#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT 2 +#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_MASK (1 << TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX (TEGRA30_AUDIOCIF_DIRECTION_TX << TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX (TEGRA30_AUDIOCIF_DIRECTION_RX << TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) + +#define TEGRA30_AUDIOCIF_TRUNCATE_ROUND 0 +#define TEGRA30_AUDIOCIF_TRUNCATE_CHOP 1 + +#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT 1 +#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_MASK (1 << TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_ROUND (TEGRA30_AUDIOCIF_TRUNCATE_ROUND << TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_TRUNCATE_CHOP (TEGRA30_AUDIOCIF_TRUNCATE_CHOP << TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) + +#define TEGRA30_AUDIOCIF_MONO_CONV_ZERO 0 +#define TEGRA30_AUDIOCIF_MONO_CONV_COPY 1 + +#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT 0 +#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_MASK (1 << TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_ZERO (TEGRA30_AUDIOCIF_MONO_CONV_ZERO << TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT) +#define TEGRA30_AUDIOCIF_CTRL_MONO_CONV_COPY (TEGRA30_AUDIOCIF_MONO_CONV_COPY << TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT) + +/* Registers within TEGRA30_AUDIO_CLUSTER_BASE */ + +/* TEGRA30_AHUB_CHANNEL_CTRL */ + +#define TEGRA30_AHUB_CHANNEL_CTRL 0x0 +#define TEGRA30_AHUB_CHANNEL_CTRL_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_CTRL_COUNT 4 +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_EN (1 << 31) +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_EN (1 << 30) +#define TEGRA30_AHUB_CHANNEL_CTRL_LOOPBACK (1 << 29) + +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_SHIFT 16 +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK_US 0xff +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK (TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_TX_THRESHOLD_SHIFT) + +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_SHIFT 8 +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK_US 0xff +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK (TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_RX_THRESHOLD_SHIFT) + +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_EN (1 << 6) + +#define TEGRA30_PACK_8_4 2 +#define TEGRA30_PACK_16 3 + +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT 4 +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK_US 3 +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK (TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT) +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_8_4 (TEGRA30_PACK_8_4 << TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT) +#define TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_16 (TEGRA30_PACK_16 << TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_SHIFT) + +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_EN (1 << 2) + +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT 0 +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK_US 3 +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK (TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_MASK_US << TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT) +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_8_4 (TEGRA30_PACK_8_4 << TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT) +#define TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_16 (TEGRA30_PACK_16 << TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_SHIFT) + +/* TEGRA30_AHUB_CHANNEL_CLEAR */ + +#define TEGRA30_AHUB_CHANNEL_CLEAR 0x4 +#define TEGRA30_AHUB_CHANNEL_CLEAR_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_CLEAR_COUNT 4 +#define TEGRA30_AHUB_CHANNEL_CLEAR_TX_SOFT_RESET (1 << 31) +#define TEGRA30_AHUB_CHANNEL_CLEAR_RX_SOFT_RESET (1 << 30) + +/* TEGRA30_AHUB_CHANNEL_STATUS */ + +#define TEGRA30_AHUB_CHANNEL_STATUS 0x8 +#define TEGRA30_AHUB_CHANNEL_STATUS_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_STATUS_COUNT 4 +#define TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_SHIFT 24 +#define TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_MASK_US 0xff +#define TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_MASK (TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_MASK_US << TEGRA30_AHUB_CHANNEL_STATUS_TX_FREE_SHIFT) +#define TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_SHIFT 16 +#define TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_MASK_US 0xff +#define TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_MASK (TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_MASK_US << TEGRA30_AHUB_CHANNEL_STATUS_RX_FREE_SHIFT) +#define TEGRA30_AHUB_CHANNEL_STATUS_TX_TRIG (1 << 1) +#define TEGRA30_AHUB_CHANNEL_STATUS_RX_TRIG (1 << 0) + +/* TEGRA30_AHUB_CHANNEL_TXFIFO */ + +#define TEGRA30_AHUB_CHANNEL_TXFIFO 0xc +#define TEGRA30_AHUB_CHANNEL_TXFIFO_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_TXFIFO_COUNT 4 + +/* TEGRA30_AHUB_CHANNEL_RXFIFO */ + +#define TEGRA30_AHUB_CHANNEL_RXFIFO 0x10 +#define TEGRA30_AHUB_CHANNEL_RXFIFO_STRIDE 0x20 +#define TEGRA30_AHUB_CHANNEL_RXFIFO_COUNT 4 + +/* TEGRA30_AHUB_CIF_TX_CTRL */ + +#define TEGRA30_AHUB_CIF_TX_CTRL 0x14 +#define TEGRA30_AHUB_CIF_TX_CTRL_STRIDE 0x20 +#define TEGRA30_AHUB_CIF_TX_CTRL_COUNT 4 +/* Uses field from TEGRA30_AUDIOCIF_CTRL_* */ + +/* TEGRA30_AHUB_CIF_RX_CTRL */ + +#define TEGRA30_AHUB_CIF_RX_CTRL 0x18 +#define TEGRA30_AHUB_CIF_RX_CTRL_STRIDE 0x20 +#define TEGRA30_AHUB_CIF_RX_CTRL_COUNT 4 +/* Uses field from TEGRA30_AUDIOCIF_CTRL_* */ + +/* TEGRA30_AHUB_CONFIG_LINK_CTRL */ + +#define TEGRA30_AHUB_CONFIG_LINK_CTRL 0x80 +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_SHIFT 28 +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_MASK_US 0xf +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_MASK (TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_MASK_US << TEGRA30_AHUB_CONFIG_LINK_CTRL_MASTER_FIFO_FULL_CNT_SHIFT) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_SHIFT 16 +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_MASK_US 0xfff +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_MASK (TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_MASK_US << TEGRA30_AHUB_CONFIG_LINK_CTRL_TIMEOUT_CNT_SHIFT) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_SHIFT 4 +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_MASK_US 0xfff +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_MASK (TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_MASK_US << TEGRA30_AHUB_CONFIG_LINK_CTRL_IDLE_CNT_SHIFT) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_CG_EN (1 << 2) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_CLEAR_TIMEOUT_CNTR (1 << 1) +#define TEGRA30_AHUB_CONFIG_LINK_CTRL_SOFT_RESET (1 << 0) + +/* TEGRA30_AHUB_MISC_CTRL */ + +#define TEGRA30_AHUB_MISC_CTRL 0x84 +#define TEGRA30_AHUB_MISC_CTRL_AUDIO_ACTIVE (1 << 31) +#define TEGRA30_AHUB_MISC_CTRL_AUDIO_CG_EN (1 << 8) +#define TEGRA30_AHUB_MISC_CTRL_AUDIO_OBS_SEL_SHIFT 0 +#define TEGRA30_AHUB_MISC_CTRL_AUDIO_OBS_SEL_MASK (0x1f << TEGRA30_AHUB_MISC_CTRL_AUDIO_OBS_SEL_SHIFT) + +/* TEGRA30_AHUB_APBDMA_LIVE_STATUS */ + +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS 0x88 +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_CIF_FIFO_FULL (1 << 31) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_CIF_FIFO_FULL (1 << 30) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_CIF_FIFO_FULL (1 << 29) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_CIF_FIFO_FULL (1 << 28) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_CIF_FIFO_FULL (1 << 27) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_CIF_FIFO_FULL (1 << 26) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_CIF_FIFO_FULL (1 << 25) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_CIF_FIFO_FULL (1 << 24) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_CIF_FIFO_EMPTY (1 << 23) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_CIF_FIFO_EMPTY (1 << 22) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_CIF_FIFO_EMPTY (1 << 21) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_CIF_FIFO_EMPTY (1 << 20) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_CIF_FIFO_EMPTY (1 << 19) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_CIF_FIFO_EMPTY (1 << 18) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_CIF_FIFO_EMPTY (1 << 17) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_CIF_FIFO_EMPTY (1 << 16) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_DMA_FIFO_FULL (1 << 15) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_DMA_FIFO_FULL (1 << 14) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_DMA_FIFO_FULL (1 << 13) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_DMA_FIFO_FULL (1 << 12) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_DMA_FIFO_FULL (1 << 11) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_DMA_FIFO_FULL (1 << 10) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_DMA_FIFO_FULL (1 << 9) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_DMA_FIFO_FULL (1 << 8) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_RX_DMA_FIFO_EMPTY (1 << 7) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH3_TX_DMA_FIFO_EMPTY (1 << 6) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_RX_DMA_FIFO_EMPTY (1 << 5) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH2_TX_DMA_FIFO_EMPTY (1 << 4) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_RX_DMA_FIFO_EMPTY (1 << 3) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH1_TX_DMA_FIFO_EMPTY (1 << 2) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_RX_DMA_FIFO_EMPTY (1 << 1) +#define TEGRA30_AHUB_APBDMA_LIVE_STATUS_CH0_TX_DMA_FIFO_EMPTY (1 << 0) + +/* TEGRA30_AHUB_I2S_LIVE_STATUS */ + +#define TEGRA30_AHUB_I2S_LIVE_STATUS 0x8c +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_RX_FIFO_FULL (1 << 29) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_TX_FIFO_FULL (1 << 28) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_RX_FIFO_FULL (1 << 27) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_TX_FIFO_FULL (1 << 26) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_RX_FIFO_FULL (1 << 25) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_TX_FIFO_FULL (1 << 24) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_RX_FIFO_FULL (1 << 23) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_TX_FIFO_FULL (1 << 22) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_RX_FIFO_FULL (1 << 21) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_TX_FIFO_FULL (1 << 20) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_RX_FIFO_ENABLED (1 << 19) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_TX_FIFO_ENABLED (1 << 18) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_RX_FIFO_ENABLED (1 << 17) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_TX_FIFO_ENABLED (1 << 16) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_RX_FIFO_ENABLED (1 << 15) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_TX_FIFO_ENABLED (1 << 14) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_RX_FIFO_ENABLED (1 << 13) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_TX_FIFO_ENABLED (1 << 12) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_RX_FIFO_ENABLED (1 << 11) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_TX_FIFO_ENABLED (1 << 10) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_RX_FIFO_EMPTY (1 << 9) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S4_TX_FIFO_EMPTY (1 << 8) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_RX_FIFO_EMPTY (1 << 7) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S3_TX_FIFO_EMPTY (1 << 6) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_RX_FIFO_EMPTY (1 << 5) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S2_TX_FIFO_EMPTY (1 << 4) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_RX_FIFO_EMPTY (1 << 3) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S1_TX_FIFO_EMPTY (1 << 2) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_RX_FIFO_EMPTY (1 << 1) +#define TEGRA30_AHUB_I2S_LIVE_STATUS_I2S0_TX_FIFO_EMPTY (1 << 0) + +/* TEGRA30_AHUB_DAM0_LIVE_STATUS */ + +#define TEGRA30_AHUB_DAM_LIVE_STATUS 0x90 +#define TEGRA30_AHUB_DAM_LIVE_STATUS_STRIDE 0x8 +#define TEGRA30_AHUB_DAM_LIVE_STATUS_COUNT 3 +#define TEGRA30_AHUB_DAM_LIVE_STATUS_TX_ENABLED (1 << 26) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX1_ENABLED (1 << 25) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX0_ENABLED (1 << 24) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_TXFIFO_FULL (1 << 15) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX1FIFO_FULL (1 << 9) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX0FIFO_FULL (1 << 8) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_TXFIFO_EMPTY (1 << 7) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX1FIFO_EMPTY (1 << 1) +#define TEGRA30_AHUB_DAM_LIVE_STATUS_RX0FIFO_EMPTY (1 << 0) + +/* TEGRA30_AHUB_SPDIF_LIVE_STATUS */ + +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS 0xa8 +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_TX_ENABLED (1 << 11) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_RX_ENABLED (1 << 10) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_TX_ENABLED (1 << 9) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_RX_ENABLED (1 << 8) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_TXFIFO_FULL (1 << 7) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_RXFIFO_FULL (1 << 6) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_TXFIFO_FULL (1 << 5) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_RXFIFO_FULL (1 << 4) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_TXFIFO_EMPTY (1 << 3) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_USER_RXFIFO_EMPTY (1 << 2) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_TXFIFO_EMPTY (1 << 1) +#define TEGRA30_AHUB_SPDIF_LIVE_STATUS_DATA_RXFIFO_EMPTY (1 << 0) + +/* TEGRA30_AHUB_I2S_INT_MASK */ + +#define TEGRA30_AHUB_I2S_INT_MASK 0xb0 + +/* TEGRA30_AHUB_DAM_INT_MASK */ + +#define TEGRA30_AHUB_DAM_INT_MASK 0xb4 + +/* TEGRA30_AHUB_SPDIF_INT_MASK */ + +#define TEGRA30_AHUB_SPDIF_INT_MASK 0xbc + +/* TEGRA30_AHUB_APBIF_INT_MASK */ + +#define TEGRA30_AHUB_APBIF_INT_MASK 0xc0 + +/* TEGRA30_AHUB_I2S_INT_STATUS */ + +#define TEGRA30_AHUB_I2S_INT_STATUS 0xc8 + +/* TEGRA30_AHUB_DAM_INT_STATUS */ + +#define TEGRA30_AHUB_DAM_INT_STATUS 0xcc + +/* TEGRA30_AHUB_SPDIF_INT_STATUS */ + +#define TEGRA30_AHUB_SPDIF_INT_STATUS 0xd4 + +/* TEGRA30_AHUB_APBIF_INT_STATUS */ + +#define TEGRA30_AHUB_APBIF_INT_STATUS 0xd8 + +/* TEGRA30_AHUB_I2S_INT_SOURCE */ + +#define TEGRA30_AHUB_I2S_INT_SOURCE 0xe0 + +/* TEGRA30_AHUB_DAM_INT_SOURCE */ + +#define TEGRA30_AHUB_DAM_INT_SOURCE 0xe4 + +/* TEGRA30_AHUB_SPDIF_INT_SOURCE */ + +#define TEGRA30_AHUB_SPDIF_INT_SOURCE 0xec + +/* TEGRA30_AHUB_APBIF_INT_SOURCE */ + +#define TEGRA30_AHUB_APBIF_INT_SOURCE 0xf0 + +/* TEGRA30_AHUB_I2S_INT_SET */ + +#define TEGRA30_AHUB_I2S_INT_SET 0xf8 + +/* TEGRA30_AHUB_DAM_INT_SET */ + +#define TEGRA30_AHUB_DAM_INT_SET 0xfc + +/* TEGRA30_AHUB_SPDIF_INT_SET */ + +#define TEGRA30_AHUB_SPDIF_INT_SET 0x100 + +/* TEGRA30_AHUB_APBIF_INT_SET */ + +#define TEGRA30_AHUB_APBIF_INT_SET 0x104 + +/* Registers within TEGRA30_AHUB_BASE */ + +#define TEGRA30_AHUB_AUDIO_RX 0x0 +#define TEGRA30_AHUB_AUDIO_RX_STRIDE 0x4 +#define TEGRA30_AHUB_AUDIO_RX_COUNT 17 +/* This register repeats once for each entry in enum tegra30_ahub_rxcif */ +/* The fields in this register are 1 bit per entry in tegra30_ahub_txcif */ + +/* + * Terminology: + * AHUB: Audio Hub; a cross-bar switch between the audio devices: DMA FIFOs, + * I2S controllers, SPDIF controllers, and DAMs. + * XBAR: The core cross-bar component of the AHUB. + * CIF: Client Interface; the HW module connecting an audio device to the + * XBAR. + * DAM: Digital Audio Mixer: A HW module that mixes multiple audio streams, + * possibly including sample-rate conversion. + * + * Each TX CIF transmits data into the XBAR. Each RX CIF can receive audio + * transmitted by a particular TX CIF. + * + * This driver is currently very simplistic; many HW features are not + * exposed; DAMs are not supported, only 16-bit stereo audio is supported, + * etc. + */ + +enum tegra30_ahub_txcif { + TEGRA30_AHUB_TXCIF_APBIF_TX0, + TEGRA30_AHUB_TXCIF_APBIF_TX1, + TEGRA30_AHUB_TXCIF_APBIF_TX2, + TEGRA30_AHUB_TXCIF_APBIF_TX3, + TEGRA30_AHUB_TXCIF_I2S0_TX0, + TEGRA30_AHUB_TXCIF_I2S1_TX0, + TEGRA30_AHUB_TXCIF_I2S2_TX0, + TEGRA30_AHUB_TXCIF_I2S3_TX0, + TEGRA30_AHUB_TXCIF_I2S4_TX0, + TEGRA30_AHUB_TXCIF_DAM0_TX0, + TEGRA30_AHUB_TXCIF_DAM1_TX0, + TEGRA30_AHUB_TXCIF_DAM2_TX0, + TEGRA30_AHUB_TXCIF_SPDIF_TX0, + TEGRA30_AHUB_TXCIF_SPDIF_TX1, +}; + +enum tegra30_ahub_rxcif { + TEGRA30_AHUB_RXCIF_APBIF_RX0, + TEGRA30_AHUB_RXCIF_APBIF_RX1, + TEGRA30_AHUB_RXcIF_APBIF_RX2, + TEGRA30_AHUB_RXCIF_APBIF_RX3, + TEGRA30_AHUB_RXCIF_I2S0_RX0, + TEGRA30_AHUB_RXCIF_I2S1_RX0, + TEGRA30_AHUB_RXCIF_I2S2_RX0, + TEGRA30_AHUB_RXCIF_I2S3_RX0, + TEGRA30_AHUB_RXCIF_I2S4_RX0, + TEGRA30_AHUB_RXCIF_DAM0_RX0, + TEGRA30_AHUB_RXCIF_DAM0_RX1, + TEGRA30_AHUB_RXCIF_DAM1_RX0, + TEGRA30_AHUB_RXCIF_DAM2_RX1, + TEGRA30_AHUB_RXCIF_DAM3_RX0, + TEGRA30_AHUB_RXCIF_DAM3_RX1, + TEGRA30_AHUB_RXCIF_SPDIF_RX0, + TEGRA30_AHUB_RXCIF_SPDIF_RX1, +}; + +extern int tegra30_ahub_allocate_rx_fifo(enum tegra30_ahub_rxcif *rxcif, + unsigned long *fiforeg, + unsigned long *reqsel); +extern int tegra30_ahub_enable_rx_fifo(enum tegra30_ahub_rxcif rxcif); +extern int tegra30_ahub_disable_rx_fifo(enum tegra30_ahub_rxcif rxcif); +extern int tegra30_ahub_free_rx_fifo(enum tegra30_ahub_rxcif rxcif); + +extern int tegra30_ahub_allocate_tx_fifo(enum tegra30_ahub_txcif *txcif, + unsigned long *fiforeg, + unsigned long *reqsel); +extern int tegra30_ahub_enable_tx_fifo(enum tegra30_ahub_txcif txcif); +extern int tegra30_ahub_disable_tx_fifo(enum tegra30_ahub_txcif txcif); +extern int tegra30_ahub_free_tx_fifo(enum tegra30_ahub_txcif txcif); + +extern int tegra30_ahub_set_rx_cif_source(enum tegra30_ahub_rxcif rxcif, + enum tegra30_ahub_txcif txcif); +extern int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif); + +struct tegra30_ahub { + struct device *dev; + struct clk *clk_d_audio; + struct clk *clk_apbif; + int dma_sel; + resource_size_t apbif_addr; + struct regmap *regmap_apbif; + struct regmap *regmap_ahub; + DECLARE_BITMAP(rx_usage, TEGRA30_AHUB_CHANNEL_CTRL_COUNT); + DECLARE_BITMAP(tx_usage, TEGRA30_AHUB_CHANNEL_CTRL_COUNT); +}; + +#endif diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c new file mode 100644 index 000000000000..44184228d1f0 --- /dev/null +++ b/sound/soc/tegra/tegra30_i2s.c @@ -0,0 +1,537 @@ +/* + * tegra30_i2s.c - Tegra30 I2S driver + * + * Author: Stephen Warren <swarren@nvidia.com> + * Copyright (c) 2010-2012, NVIDIA CORPORATION. All rights reserved. + * + * Based on code copyright/by: + * + * Copyright (c) 2009-2010, NVIDIA Corporation. + * Scott Peterson <speterson@nvidia.com> + * + * Copyright (C) 2010 Google, Inc. + * Iliyan Malchev <malchev@google.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "tegra30_ahub.h" +#include "tegra30_i2s.h" + +#define DRV_NAME "tegra30-i2s" + +static int tegra30_i2s_runtime_suspend(struct device *dev) +{ + struct tegra30_i2s *i2s = dev_get_drvdata(dev); + + regcache_cache_only(i2s->regmap, true); + + clk_disable_unprepare(i2s->clk_i2s); + + return 0; +} + +static int tegra30_i2s_runtime_resume(struct device *dev) +{ + struct tegra30_i2s *i2s = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(i2s->clk_i2s); + if (ret) { + dev_err(dev, "clk_enable failed: %d\n", ret); + return ret; + } + + regcache_cache_only(i2s->regmap, false); + + return 0; +} + +int tegra30_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + int ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = tegra30_ahub_allocate_tx_fifo(&i2s->playback_fifo_cif, + &i2s->playback_dma_data.addr, + &i2s->playback_dma_data.req_sel); + i2s->playback_dma_data.wrap = 4; + i2s->playback_dma_data.width = 32; + tegra30_ahub_set_rx_cif_source(i2s->playback_i2s_cif, + i2s->playback_fifo_cif); + } else { + ret = tegra30_ahub_allocate_rx_fifo(&i2s->capture_fifo_cif, + &i2s->capture_dma_data.addr, + &i2s->capture_dma_data.req_sel); + i2s->capture_dma_data.wrap = 4; + i2s->capture_dma_data.width = 32; + tegra30_ahub_set_rx_cif_source(i2s->capture_fifo_cif, + i2s->capture_i2s_cif); + } + + return ret; +} + +void tegra30_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + tegra30_ahub_unset_rx_cif_source(i2s->playback_i2s_cif); + tegra30_ahub_free_tx_fifo(i2s->playback_fifo_cif); + } else { + tegra30_ahub_unset_rx_cif_source(i2s->capture_fifo_cif); + tegra30_ahub_free_rx_fifo(i2s->capture_fifo_cif); + } +} + +static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned int mask, val; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + mask = TEGRA30_I2S_CTRL_MASTER_ENABLE; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + val = TEGRA30_I2S_CTRL_MASTER_ENABLE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + mask |= TEGRA30_I2S_CTRL_FRAME_FORMAT_MASK | + TEGRA30_I2S_CTRL_LRCK_MASK; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC; + val |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_DSP_B: + val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC; + val |= TEGRA30_I2S_CTRL_LRCK_R_LOW; + break; + case SND_SOC_DAIFMT_I2S: + val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; + val |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; + val |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_LEFT_J: + val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; + val |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + break; + default: + return -EINVAL; + } + + pm_runtime_get_sync(dai->dev); + regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, mask, val); + pm_runtime_put(dai->dev); + + return 0; +} + +static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct device *dev = dai->dev; + struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned int mask, val, reg; + int ret, sample_size, srate, i2sclock, bitcnt; + + if (params_channels(params) != 2) + return -EINVAL; + + mask = TEGRA30_I2S_CTRL_BIT_SIZE_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + val = TEGRA30_I2S_CTRL_BIT_SIZE_16; + sample_size = 16; + break; + default: + return -EINVAL; + } + + regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, mask, val); + + srate = params_rate(params); + + /* Final "* 2" required by Tegra hardware */ + i2sclock = srate * params_channels(params) * sample_size * 2; + + bitcnt = (i2sclock / (2 * srate)) - 1; + if (bitcnt < 0 || bitcnt > TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US) + return -EINVAL; + + ret = clk_set_rate(i2s->clk_i2s, i2sclock); + if (ret) { + dev_err(dev, "Can't set I2S clock rate: %d\n", ret); + return ret; + } + + val = bitcnt << TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT; + + if (i2sclock % (2 * srate)) + val |= TEGRA30_I2S_TIMING_NON_SYM_ENABLE; + + regmap_write(i2s->regmap, TEGRA30_I2S_TIMING, val); + + val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | + (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | + TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | + TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX; + reg = TEGRA30_I2S_CIF_RX_CTRL; + } else { + val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; + reg = TEGRA30_I2S_CIF_RX_CTRL; + } + + regmap_write(i2s->regmap, reg, val); + + val = (1 << TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT) | + (1 << TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT); + regmap_write(i2s->regmap, TEGRA30_I2S_OFFSET, val); + + return 0; +} + +static void tegra30_i2s_start_playback(struct tegra30_i2s *i2s) +{ + tegra30_ahub_enable_tx_fifo(i2s->playback_fifo_cif); + regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, + TEGRA30_I2S_CTRL_XFER_EN_TX, + TEGRA30_I2S_CTRL_XFER_EN_TX); +} + +static void tegra30_i2s_stop_playback(struct tegra30_i2s *i2s) +{ + tegra30_ahub_disable_tx_fifo(i2s->playback_fifo_cif); + regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, + TEGRA30_I2S_CTRL_XFER_EN_TX, 0); +} + +static void tegra30_i2s_start_capture(struct tegra30_i2s *i2s) +{ + tegra30_ahub_enable_rx_fifo(i2s->capture_fifo_cif); + regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, + TEGRA30_I2S_CTRL_XFER_EN_RX, + TEGRA30_I2S_CTRL_XFER_EN_RX); +} + +static void tegra30_i2s_stop_capture(struct tegra30_i2s *i2s) +{ + tegra30_ahub_disable_rx_fifo(i2s->capture_fifo_cif); + regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, + TEGRA30_I2S_CTRL_XFER_EN_RX, 0); +} + +static int tegra30_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tegra30_i2s_start_playback(i2s); + else + tegra30_i2s_start_capture(i2s); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tegra30_i2s_stop_playback(i2s); + else + tegra30_i2s_stop_capture(i2s); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int tegra30_i2s_probe(struct snd_soc_dai *dai) +{ + struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + dai->capture_dma_data = &i2s->capture_dma_data; + dai->playback_dma_data = &i2s->playback_dma_data; + + return 0; +} + +static struct snd_soc_dai_ops tegra30_i2s_dai_ops = { + .startup = tegra30_i2s_startup, + .shutdown = tegra30_i2s_shutdown, + .set_fmt = tegra30_i2s_set_fmt, + .hw_params = tegra30_i2s_hw_params, + .trigger = tegra30_i2s_trigger, +}; + +static const struct snd_soc_dai_driver tegra30_i2s_dai_template = { + .probe = tegra30_i2s_probe, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &tegra30_i2s_dai_ops, + .symmetric_rates = 1, +}; + +static bool tegra30_i2s_wr_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA30_I2S_CTRL: + case TEGRA30_I2S_TIMING: + case TEGRA30_I2S_OFFSET: + case TEGRA30_I2S_CH_CTRL: + case TEGRA30_I2S_SLOT_CTRL: + case TEGRA30_I2S_CIF_RX_CTRL: + case TEGRA30_I2S_CIF_TX_CTRL: + case TEGRA30_I2S_FLOWCTL: + case TEGRA30_I2S_TX_STEP: + case TEGRA30_I2S_FLOW_STATUS: + case TEGRA30_I2S_FLOW_TOTAL: + case TEGRA30_I2S_FLOW_OVER: + case TEGRA30_I2S_FLOW_UNDER: + case TEGRA30_I2S_LCOEF_1_4_0: + case TEGRA30_I2S_LCOEF_1_4_1: + case TEGRA30_I2S_LCOEF_1_4_2: + case TEGRA30_I2S_LCOEF_1_4_3: + case TEGRA30_I2S_LCOEF_1_4_4: + case TEGRA30_I2S_LCOEF_1_4_5: + case TEGRA30_I2S_LCOEF_2_4_0: + case TEGRA30_I2S_LCOEF_2_4_1: + case TEGRA30_I2S_LCOEF_2_4_2: + return true; + default: + return false; + }; +} + +static bool tegra30_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TEGRA30_I2S_FLOW_STATUS: + case TEGRA30_I2S_FLOW_TOTAL: + case TEGRA30_I2S_FLOW_OVER: + case TEGRA30_I2S_FLOW_UNDER: + return true; + default: + return false; + }; +} + +static const struct regmap_config tegra30_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = TEGRA30_I2S_LCOEF_2_4_2, + .writeable_reg = tegra30_i2s_wr_rd_reg, + .readable_reg = tegra30_i2s_wr_rd_reg, + .volatile_reg = tegra30_i2s_volatile_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static __devinit int tegra30_i2s_platform_probe(struct platform_device *pdev) +{ + struct tegra30_i2s *i2s; + u32 cif_ids[2]; + struct resource *mem, *memregion; + void __iomem *regs; + int ret; + + i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra30_i2s), GFP_KERNEL); + if (!i2s) { + dev_err(&pdev->dev, "Can't allocate tegra30_i2s\n"); + ret = -ENOMEM; + goto err; + } + dev_set_drvdata(&pdev->dev, i2s); + + i2s->dai = tegra30_i2s_dai_template; + i2s->dai.name = dev_name(&pdev->dev); + + ret = of_property_read_u32_array(pdev->dev.of_node, + "nvidia,ahub-cif-ids", cif_ids, + ARRAY_SIZE(cif_ids)); + if (ret < 0) + goto err; + + i2s->playback_i2s_cif = cif_ids[0]; + i2s->capture_i2s_cif = cif_ids[1]; + + i2s->clk_i2s = clk_get(&pdev->dev, NULL); + if (IS_ERR(i2s->clk_i2s)) { + dev_err(&pdev->dev, "Can't retrieve i2s clock\n"); + ret = PTR_ERR(i2s->clk_i2s); + goto err; + } + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "No memory resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + memregion = devm_request_mem_region(&pdev->dev, mem->start, + resource_size(mem), DRV_NAME); + if (!memregion) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + ret = -EBUSY; + goto err_clk_put; + } + + regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); + if (!regs) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err_clk_put; + } + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, regs, + &tegra30_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + ret = PTR_ERR(i2s->regmap); + goto err_clk_put; + } + regcache_cache_only(i2s->regmap, true); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = tegra30_i2s_runtime_resume(&pdev->dev); + if (ret) + goto err_pm_disable; + } + + ret = snd_soc_register_dai(&pdev->dev, &i2s->dai); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + ret = -ENOMEM; + goto err_suspend; + } + + ret = tegra_pcm_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + goto err_unregister_dai; + } + + return 0; + +err_unregister_dai: + snd_soc_unregister_dai(&pdev->dev); +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra30_i2s_runtime_suspend(&pdev->dev); +err_pm_disable: + pm_runtime_disable(&pdev->dev); +err_clk_put: + clk_put(i2s->clk_i2s); +err: + return ret; +} + +static int __devexit tegra30_i2s_platform_remove(struct platform_device *pdev) +{ + struct tegra30_i2s *i2s = dev_get_drvdata(&pdev->dev); + + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + tegra30_i2s_runtime_suspend(&pdev->dev); + + tegra_pcm_platform_unregister(&pdev->dev); + snd_soc_unregister_dai(&pdev->dev); + + clk_put(i2s->clk_i2s); + + return 0; +} + +static const struct of_device_id tegra30_i2s_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra30-i2s", }, + {}, +}; + +static const struct dev_pm_ops tegra30_i2s_pm_ops __devinitconst = { + SET_RUNTIME_PM_OPS(tegra30_i2s_runtime_suspend, + tegra30_i2s_runtime_resume, NULL) +}; + +static struct platform_driver tegra30_i2s_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = tegra30_i2s_of_match, + .pm = &tegra30_i2s_pm_ops, + }, + .probe = tegra30_i2s_platform_probe, + .remove = __devexit_p(tegra30_i2s_platform_remove), +}; +module_platform_driver(tegra30_i2s_driver); + +MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); +MODULE_DESCRIPTION("Tegra30 I2S ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra30_i2s_of_match); diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h new file mode 100644 index 000000000000..34dc47b9581c --- /dev/null +++ b/sound/soc/tegra/tegra30_i2s.h @@ -0,0 +1,241 @@ +/* + * tegra30_i2s.h - Definitions for Tegra30 I2S driver + * + * Copyright (c) 2011,2012, NVIDIA CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#ifndef __TEGRA30_I2S_H__ +#define __TEGRA30_I2S_H__ + +#include "tegra_pcm.h" + +/* Register offsets from TEGRA30_I2S*_BASE */ + +#define TEGRA30_I2S_CTRL 0x0 +#define TEGRA30_I2S_TIMING 0x4 +#define TEGRA30_I2S_OFFSET 0x08 +#define TEGRA30_I2S_CH_CTRL 0x0c +#define TEGRA30_I2S_SLOT_CTRL 0x10 +#define TEGRA30_I2S_CIF_RX_CTRL 0x14 +#define TEGRA30_I2S_CIF_TX_CTRL 0x18 +#define TEGRA30_I2S_FLOWCTL 0x1c +#define TEGRA30_I2S_TX_STEP 0x20 +#define TEGRA30_I2S_FLOW_STATUS 0x24 +#define TEGRA30_I2S_FLOW_TOTAL 0x28 +#define TEGRA30_I2S_FLOW_OVER 0x2c +#define TEGRA30_I2S_FLOW_UNDER 0x30 +#define TEGRA30_I2S_LCOEF_1_4_0 0x34 +#define TEGRA30_I2S_LCOEF_1_4_1 0x38 +#define TEGRA30_I2S_LCOEF_1_4_2 0x3c +#define TEGRA30_I2S_LCOEF_1_4_3 0x40 +#define TEGRA30_I2S_LCOEF_1_4_4 0x44 +#define TEGRA30_I2S_LCOEF_1_4_5 0x48 +#define TEGRA30_I2S_LCOEF_2_4_0 0x4c +#define TEGRA30_I2S_LCOEF_2_4_1 0x50 +#define TEGRA30_I2S_LCOEF_2_4_2 0x54 + +/* Fields in TEGRA30_I2S_CTRL */ + +#define TEGRA30_I2S_CTRL_XFER_EN_TX (1 << 31) +#define TEGRA30_I2S_CTRL_XFER_EN_RX (1 << 30) +#define TEGRA30_I2S_CTRL_CG_EN (1 << 29) +#define TEGRA30_I2S_CTRL_SOFT_RESET (1 << 28) +#define TEGRA30_I2S_CTRL_TX_FLOWCTL_EN (1 << 27) + +#define TEGRA30_I2S_CTRL_OBS_SEL_SHIFT 24 +#define TEGRA30_I2S_CTRL_OBS_SEL_MASK (7 << TEGRA30_I2S_CTRL_OBS_SEL_SHIFT) + +#define TEGRA30_I2S_FRAME_FORMAT_LRCK 0 +#define TEGRA30_I2S_FRAME_FORMAT_FSYNC 1 + +#define TEGRA30_I2S_CTRL_FRAME_FORMAT_SHIFT 12 +#define TEGRA30_I2S_CTRL_FRAME_FORMAT_MASK (7 << TEGRA30_I2S_CTRL_FRAME_FORMAT_SHIFT) +#define TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK (TEGRA30_I2S_FRAME_FORMAT_LRCK << TEGRA30_I2S_CTRL_FRAME_FORMAT_SHIFT) +#define TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC (TEGRA30_I2S_FRAME_FORMAT_FSYNC << TEGRA30_I2S_CTRL_FRAME_FORMAT_SHIFT) + +#define TEGRA30_I2S_CTRL_MASTER_ENABLE (1 << 10) + +#define TEGRA30_I2S_LRCK_LEFT_LOW 0 +#define TEGRA30_I2S_LRCK_RIGHT_LOW 1 + +#define TEGRA30_I2S_CTRL_LRCK_SHIFT 9 +#define TEGRA30_I2S_CTRL_LRCK_MASK (1 << TEGRA30_I2S_CTRL_LRCK_SHIFT) +#define TEGRA30_I2S_CTRL_LRCK_L_LOW (TEGRA30_I2S_LRCK_LEFT_LOW << TEGRA30_I2S_CTRL_LRCK_SHIFT) +#define TEGRA30_I2S_CTRL_LRCK_R_LOW (TEGRA30_I2S_LRCK_RIGHT_LOW << TEGRA30_I2S_CTRL_LRCK_SHIFT) + +#define TEGRA30_I2S_CTRL_LPBK_ENABLE (1 << 8) + +#define TEGRA30_I2S_BIT_CODE_LINEAR 0 +#define TEGRA30_I2S_BIT_CODE_ULAW 1 +#define TEGRA30_I2S_BIT_CODE_ALAW 2 + +#define TEGRA30_I2S_CTRL_BIT_CODE_SHIFT 4 +#define TEGRA30_I2S_CTRL_BIT_CODE_MASK (3 << TEGRA30_I2S_CTRL_BIT_CODE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_CODE_LINEAR (TEGRA30_I2S_BIT_CODE_LINEAR << TEGRA30_I2S_CTRL_BIT_CODE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_CODE_ULAW (TEGRA30_I2S_BIT_CODE_ULAW << TEGRA30_I2S_CTRL_BIT_CODE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_CODE_ALAW (TEGRA30_I2S_BIT_CODE_ALAW << TEGRA30_I2S_CTRL_BIT_CODE_SHIFT) + +#define TEGRA30_I2S_BITS_8 1 +#define TEGRA30_I2S_BITS_12 2 +#define TEGRA30_I2S_BITS_16 3 +#define TEGRA30_I2S_BITS_20 4 +#define TEGRA30_I2S_BITS_24 5 +#define TEGRA30_I2S_BITS_28 6 +#define TEGRA30_I2S_BITS_32 7 + +/* Sample container size; see {RX,TX}_MASK field in CH_CTRL below */ +#define TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT 0 +#define TEGRA30_I2S_CTRL_BIT_SIZE_MASK (7 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_8 (TEGRA30_I2S_BITS_8 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_12 (TEGRA30_I2S_BITS_12 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_16 (TEGRA30_I2S_BITS_16 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_20 (TEGRA30_I2S_BITS_20 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_24 (TEGRA30_I2S_BITS_24 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_28 (TEGRA30_I2S_BITS_28 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA30_I2S_CTRL_BIT_SIZE_32 (TEGRA30_I2S_BITS_32 << TEGRA30_I2S_CTRL_BIT_SIZE_SHIFT) + +/* Fields in TEGRA30_I2S_TIMING */ + +#define TEGRA30_I2S_TIMING_NON_SYM_ENABLE (1 << 12) +#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 +#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff +#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) + +/* Fields in TEGRA30_I2S_OFFSET */ + +#define TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT 16 +#define TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_MASK_US 0x7ff +#define TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_MASK (TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_MASK_US << TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT) +#define TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT 0 +#define TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_MASK_US 0x7ff +#define TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_MASK (TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_MASK_US << TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT) + +/* Fields in TEGRA30_I2S_CH_CTRL */ + +/* (FSYNC width - 1) in bit clocks */ +#define TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_SHIFT 24 +#define TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_MASK_US 0xff +#define TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_MASK (TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_MASK_US << TEGRA30_I2S_CH_CTRL_FSYNC_WIDTH_SHIFT) + +#define TEGRA30_I2S_HIGHZ_NO 0 +#define TEGRA30_I2S_HIGHZ_YES 1 +#define TEGRA30_I2S_HIGHZ_ON_HALF_BIT_CLK 2 + +#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT 12 +#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_MASK (3 << TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT) +#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_NO (TEGRA30_I2S_HIGHZ_NO << TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT) +#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_YES (TEGRA30_I2S_HIGHZ_YES << TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT) +#define TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_ON_HALF_BIT_CLK (TEGRA30_I2S_HIGHZ_ON_HALF_BIT_CLK << TEGRA30_I2S_CH_CTRL_HIGHZ_CTRL_SHIFT) + +#define TEGRA30_I2S_MSB_FIRST 0 +#define TEGRA30_I2S_LSB_FIRST 1 + +#define TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_SHIFT 10 +#define TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_MASK (1 << TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_SHIFT) +#define TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_MSB_FIRST (TEGRA30_I2S_MSB_FIRST << TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_SHIFT) +#define TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_LSB_FIRST (TEGRA30_I2S_LSB_FIRST << TEGRA30_I2S_CH_CTRL_RX_BIT_ORDER_SHIFT) +#define TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_SHIFT 9 +#define TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_MASK (1 << TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_SHIFT) +#define TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_MSB_FIRST (TEGRA30_I2S_MSB_FIRST << TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_SHIFT) +#define TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_LSB_FIRST (TEGRA30_I2S_LSB_FIRST << TEGRA30_I2S_CH_CTRL_TX_BIT_ORDER_SHIFT) + +#define TEGRA30_I2S_POS_EDGE 0 +#define TEGRA30_I2S_NEG_EDGE 1 + +#define TEGRA30_I2S_CH_CTRL_EGDE_CTRL_SHIFT 8 +#define TEGRA30_I2S_CH_CTRL_EGDE_CTRL_MASK (1 << TEGRA30_I2S_CH_CTRL_EGDE_CTRL_SHIFT) +#define TEGRA30_I2S_CH_CTRL_EGDE_CTRL_POS_EDGE (TEGRA30_I2S_POS_EDGE << TEGRA30_I2S_CH_CTRL_EGDE_CTRL_SHIFT) +#define TEGRA30_I2S_CH_CTRL_EGDE_CTRL_NEG_EDGE (TEGRA30_I2S_NEG_EDGE << TEGRA30_I2S_CH_CTRL_EGDE_CTRL_SHIFT) + +/* Sample size is # bits from BIT_SIZE minus this field */ +#define TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_SHIFT 4 +#define TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_MASK_US 7 +#define TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_MASK (TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_MASK_US << TEGRA30_I2S_CH_CTRL_RX_MASK_BITS_SHIFT) + +#define TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_SHIFT 0 +#define TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_MASK_US 7 +#define TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_MASK (TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_MASK_US << TEGRA30_I2S_CH_CTRL_TX_MASK_BITS_SHIFT) + +/* Fields in TEGRA30_I2S_SLOT_CTRL */ + +/* Number of slots in frame, minus 1 */ +#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT 16 +#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US 7 +#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK (TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_SHIFT) + +/* TDM mode slot enable bitmask */ +#define TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_SHIFT 8 +#define TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_MASK (0xff << TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_SHIFT) + +#define TEGRA30_I2S_SLOT_CTRL_TX_SLOT_ENABLES_SHIFT 0 +#define TEGRA30_I2S_SLOT_CTRL_TX_SLOT_ENABLES_MASK (0xff << TEGRA30_I2S_SLOT_CTRL_TX_SLOT_ENABLES_SHIFT) + +/* Fields in TEGRA30_I2S_CIF_RX_CTRL */ +/* Uses field from TEGRA30_AUDIOCIF_CTRL_* in tegra30_ahub.h */ + +/* Fields in TEGRA30_I2S_CIF_TX_CTRL */ +/* Uses field from TEGRA30_AUDIOCIF_CTRL_* in tegra30_ahub.h */ + +/* Fields in TEGRA30_I2S_FLOWCTL */ + +#define TEGRA30_I2S_FILTER_LINEAR 0 +#define TEGRA30_I2S_FILTER_QUAD 1 + +#define TEGRA30_I2S_FLOWCTL_FILTER_SHIFT 31 +#define TEGRA30_I2S_FLOWCTL_FILTER_MASK (1 << TEGRA30_I2S_FLOWCTL_FILTER_SHIFT) +#define TEGRA30_I2S_FLOWCTL_FILTER_LINEAR (TEGRA30_I2S_FILTER_LINEAR << TEGRA30_I2S_FLOWCTL_FILTER_SHIFT) +#define TEGRA30_I2S_FLOWCTL_FILTER_QUAD (TEGRA30_I2S_FILTER_QUAD << TEGRA30_I2S_FLOWCTL_FILTER_SHIFT) + +/* Fields in TEGRA30_I2S_TX_STEP */ + +#define TEGRA30_I2S_TX_STEP_SHIFT 0 +#define TEGRA30_I2S_TX_STEP_MASK_US 0xffff +#define TEGRA30_I2S_TX_STEP_MASK (TEGRA30_I2S_TX_STEP_MASK_US << TEGRA30_I2S_TX_STEP_SHIFT) + +/* Fields in TEGRA30_I2S_FLOW_STATUS */ + +#define TEGRA30_I2S_FLOW_STATUS_UNDERFLOW (1 << 31) +#define TEGRA30_I2S_FLOW_STATUS_OVERFLOW (1 << 30) +#define TEGRA30_I2S_FLOW_STATUS_MONITOR_INT_EN (1 << 4) +#define TEGRA30_I2S_FLOW_STATUS_COUNTER_CLR (1 << 3) +#define TEGRA30_I2S_FLOW_STATUS_MONITOR_CLR (1 << 2) +#define TEGRA30_I2S_FLOW_STATUS_COUNTER_EN (1 << 1) +#define TEGRA30_I2S_FLOW_STATUS_MONITOR_EN (1 << 0) + +/* + * There are no fields in TEGRA30_I2S_FLOW_TOTAL, TEGRA30_I2S_FLOW_OVER, + * TEGRA30_I2S_FLOW_UNDER; they are counters taking the whole register. + */ + +/* Fields in TEGRA30_I2S_LCOEF_* */ + +#define TEGRA30_I2S_LCOEF_COEF_SHIFT 0 +#define TEGRA30_I2S_LCOEF_COEF_MASK_US 0xffff +#define TEGRA30_I2S_LCOEF_COEF_MASK (TEGRA30_I2S_LCOEF_COEF_MASK_US << TEGRA30_I2S_LCOEF_COEF_SHIFT) + +struct tegra30_i2s { + struct snd_soc_dai_driver dai; + int cif_id; + struct clk *clk_i2s; + enum tegra30_ahub_txcif capture_i2s_cif; + enum tegra30_ahub_rxcif capture_fifo_cif; + struct tegra_pcm_dma_params capture_dma_data; + enum tegra30_ahub_rxcif playback_i2s_cif; + enum tegra30_ahub_txcif playback_fifo_cif; + struct tegra_pcm_dma_params playback_dma_data; + struct regmap *regmap; +}; + +#endif diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index e45ccd851f6a..d684df294c0c 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -1,16 +1,17 @@ /* * tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver -* -* Copyright (C) 2011 The AC100 Kernel Team <ac100@lists.lauchpad.net> -* -* Authors: Leon Romanovsky <leon@leon.nu> -* Andrey Danin <danindrey@mail.ru> -* Marc Dietrich <marvin24@gmx.de> -* -* This program is free software; you can redistribute it and/or modify -* it under the terms of the GNU General Public License version 2 as -* published by the Free Software Foundation. -*/ + * + * Copyright (C) 2011 The AC100 Kernel Team <ac100@lists.lauchpad.net> + * Copyright (C) 2012 - NVIDIA, Inc. + * + * Authors: Leon Romanovsky <leon@leon.nu> + * Andrey Danin <danindrey@mail.ru> + * Marc Dietrich <marvin24@gmx.de> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ #include <asm/mach-types.h> @@ -28,19 +29,12 @@ #include "../codecs/alc5632.h" -#include "tegra_das.h" -#include "tegra_i2s.h" -#include "tegra_pcm.h" #include "tegra_asoc_utils.h" #define DRV_NAME "tegra-alc5632" -#define GPIO_HP_DET BIT(0) - struct tegra_alc5632 { struct tegra_asoc_utils_data util_data; - struct platform_device *pcm_dev; - int gpio_requested; int gpio_hp_det; }; @@ -49,7 +43,7 @@ static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_card *card = codec->card; struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); int srate, mclk; @@ -111,9 +105,9 @@ static const struct snd_kcontrol_new tegra_alc5632_controls[] = { static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - struct device_node *np = codec->card->dev->of_node; struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(codec->card); snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, @@ -122,14 +116,11 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(tegra_alc5632_hs_jack_pins), tegra_alc5632_hs_jack_pins); - machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); - if (gpio_is_valid(machine->gpio_hp_det)) { tegra_alc5632_hp_jack_gpio.gpio = machine->gpio_hp_det; snd_soc_jack_add_gpios(&tegra_alc5632_hs_jack, 1, &tegra_alc5632_hp_jack_gpio); - machine->gpio_requested |= GPIO_HP_DET; } snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); @@ -140,7 +131,6 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link tegra_alc5632_dai = { .name = "ALC5632", .stream_name = "ALC5632 PCM", - .platform_name = "tegra-pcm-audio", .codec_dai_name = "alc5632-hifi", .init = tegra_alc5632_asoc_init, .ops = &tegra_alc5632_asoc_ops, @@ -163,6 +153,7 @@ static struct snd_soc_card snd_soc_tegra_alc5632 = { static __devinit int tegra_alc5632_probe(struct platform_device *pdev) { + struct device_node *np = pdev->dev.of_node; struct snd_soc_card *card = &snd_soc_tegra_alc5632; struct tegra_alc5632 *alc5632; int ret; @@ -179,14 +170,16 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, alc5632); - alc5632->pcm_dev = ERR_PTR(-EINVAL); - if (!(pdev->dev.of_node)) { dev_err(&pdev->dev, "Must be instantiated using device tree\n"); ret = -EINVAL; goto err; } + alc5632->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + if (alc5632->gpio_hp_det == -ENODEV) + return -EPROBE_DEFER; + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); if (ret) goto err; @@ -205,27 +198,20 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) goto err; } - tegra_alc5632_dai.cpu_dai_of_node = of_parse_phandle( + tegra_alc5632_dai.cpu_of_node = of_parse_phandle( pdev->dev.of_node, "nvidia,i2s-controller", 0); - if (!tegra_alc5632_dai.cpu_dai_of_node) { + if (!tegra_alc5632_dai.cpu_of_node) { dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; goto err; } - alc5632->pcm_dev = platform_device_register_simple( - "tegra-pcm-audio", -1, NULL, 0); - if (IS_ERR(alc5632->pcm_dev)) { - dev_err(&pdev->dev, - "Can't instantiate tegra-pcm-audio\n"); - ret = PTR_ERR(alc5632->pcm_dev); - goto err; - } + tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_of_node; ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); if (ret) - goto err_unregister; + goto err; ret = snd_soc_register_card(card); if (ret) { @@ -238,9 +224,6 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&alc5632->util_data); -err_unregister: - if (!IS_ERR(alc5632->pcm_dev)) - platform_device_unregister(alc5632->pcm_dev); err: return ret; } @@ -250,17 +233,12 @@ static int __devexit tegra_alc5632_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(card); - if (machine->gpio_requested & GPIO_HP_DET) - snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack, - 1, - &tegra_alc5632_hp_jack_gpio); - machine->gpio_requested = 0; + snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack, 1, + &tegra_alc5632_hp_jack_gpio); snd_soc_unregister_card(card); tegra_asoc_utils_fini(&machine->util_data); - if (!IS_ERR(machine->pcm_dev)) - platform_device_unregister(machine->pcm_dev); return 0; } diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index f8428e410e05..6872c77a1196 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -2,7 +2,7 @@ * tegra_asoc_utils.c - Harmony machine ASoC driver * * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010,2012 - NVIDIA, Inc. * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -25,6 +25,7 @@ #include <linux/err.h> #include <linux/kernel.h> #include <linux/module.h> +#include <linux/of.h> #include "tegra_asoc_utils.h" @@ -40,7 +41,10 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, case 22050: case 44100: case 88200: - new_baseclock = 56448000; + if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) + new_baseclock = 56448000; + else + new_baseclock = 564480000; break; case 8000: case 16000: @@ -48,7 +52,10 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, case 48000: case 64000: case 96000: - new_baseclock = 73728000; + if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) + new_baseclock = 73728000; + else + new_baseclock = 552960000; break; default: return -EINVAL; @@ -62,9 +69,9 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, data->set_baseclock = 0; data->set_mclk = 0; - clk_disable(data->clk_cdev1); - clk_disable(data->clk_pll_a_out0); - clk_disable(data->clk_pll_a); + clk_disable_unprepare(data->clk_cdev1); + clk_disable_unprepare(data->clk_pll_a_out0); + clk_disable_unprepare(data->clk_pll_a); err = clk_set_rate(data->clk_pll_a, new_baseclock); if (err) { @@ -78,21 +85,21 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, return err; } - /* Don't set cdev1 rate; its locked to pll_a_out0 */ + /* Don't set cdev1/extern1 rate; it's locked to pll_a_out0 */ - err = clk_enable(data->clk_pll_a); + err = clk_prepare_enable(data->clk_pll_a); if (err) { dev_err(data->dev, "Can't enable pll_a: %d\n", err); return err; } - err = clk_enable(data->clk_pll_a_out0); + err = clk_prepare_enable(data->clk_pll_a_out0); if (err) { dev_err(data->dev, "Can't enable pll_a_out0: %d\n", err); return err; } - err = clk_enable(data->clk_cdev1); + err = clk_prepare_enable(data->clk_cdev1); if (err) { dev_err(data->dev, "Can't enable cdev1: %d\n", err); return err; @@ -112,6 +119,17 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, data->dev = dev; + if (of_machine_is_compatible("nvidia,tegra20")) + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; + else if (of_machine_is_compatible("nvidia,tegra30")) + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30; + else if (!dev->of_node) + /* non-DT is always Tegra20 */ + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; + else + /* DT boot, but unknown SoC */ + return -EINVAL; + data->clk_pll_a = clk_get_sys(NULL, "pll_a"); if (IS_ERR(data->clk_pll_a)) { dev_err(data->dev, "Can't retrieve clk pll_a\n"); @@ -126,15 +144,24 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, goto err_put_pll_a; } - data->clk_cdev1 = clk_get_sys(NULL, "cdev1"); + if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) + data->clk_cdev1 = clk_get_sys(NULL, "cdev1"); + else + data->clk_cdev1 = clk_get_sys("extern1", NULL); if (IS_ERR(data->clk_cdev1)) { dev_err(data->dev, "Can't retrieve clk cdev1\n"); ret = PTR_ERR(data->clk_cdev1); goto err_put_pll_a_out0; } + ret = tegra_asoc_utils_set_rate(data, 44100, 256 * 44100); + if (ret) + goto err_put_cdev1; + return 0; +err_put_cdev1: + clk_put(data->clk_cdev1); err_put_pll_a_out0: clk_put(data->clk_pll_a_out0); err_put_pll_a: diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h index 4818195da25c..44db1dbb8f21 100644 --- a/sound/soc/tegra/tegra_asoc_utils.h +++ b/sound/soc/tegra/tegra_asoc_utils.h @@ -2,7 +2,7 @@ * tegra_asoc_utils.h - Definitions for Tegra DAS driver * * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010,2012 - NVIDIA, Inc. * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -26,8 +26,14 @@ struct clk; struct device; +enum tegra_asoc_utils_soc { + TEGRA_ASOC_UTILS_SOC_TEGRA20, + TEGRA_ASOC_UTILS_SOC_TEGRA30, +}; + struct tegra_asoc_utils_data { struct device *dev; + enum tegra_asoc_utils_soc soc; struct clk *clk_pll_a; struct clk *clk_pll_a_out0; struct clk *clk_cdev1; @@ -42,4 +48,3 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data); #endif - diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c deleted file mode 100644 index 3b3c1ba4d235..000000000000 --- a/sound/soc/tegra/tegra_das.c +++ /dev/null @@ -1,261 +0,0 @@ -/* - * tegra_das.c - Tegra DAS driver - * - * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include <linux/module.h> -#include <linux/debugfs.h> -#include <linux/device.h> -#include <linux/platform_device.h> -#include <linux/seq_file.h> -#include <linux/slab.h> -#include <linux/io.h> -#include <mach/iomap.h> -#include <sound/soc.h> -#include "tegra_das.h" - -#define DRV_NAME "tegra-das" - -static struct tegra_das *das; - -static inline void tegra_das_write(u32 reg, u32 val) -{ - __raw_writel(val, das->regs + reg); -} - -static inline u32 tegra_das_read(u32 reg) -{ - return __raw_readl(das->regs + reg); -} - -int tegra_das_connect_dap_to_dac(int dap, int dac) -{ - u32 addr; - u32 reg; - - if (!das) - return -ENODEV; - - addr = TEGRA_DAS_DAP_CTRL_SEL + - (dap * TEGRA_DAS_DAP_CTRL_SEL_STRIDE); - reg = dac << TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P; - - tegra_das_write(addr, reg); - - return 0; -} -EXPORT_SYMBOL_GPL(tegra_das_connect_dap_to_dac); - -int tegra_das_connect_dap_to_dap(int dap, int otherdap, int master, - int sdata1rx, int sdata2rx) -{ - u32 addr; - u32 reg; - - if (!das) - return -ENODEV; - - addr = TEGRA_DAS_DAP_CTRL_SEL + - (dap * TEGRA_DAS_DAP_CTRL_SEL_STRIDE); - reg = otherdap << TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P | - !!sdata2rx << TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P | - !!sdata1rx << TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P | - !!master << TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P; - - tegra_das_write(addr, reg); - - return 0; -} -EXPORT_SYMBOL_GPL(tegra_das_connect_dap_to_dap); - -int tegra_das_connect_dac_to_dap(int dac, int dap) -{ - u32 addr; - u32 reg; - - if (!das) - return -ENODEV; - - addr = TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL + - (dac * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE); - reg = dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P | - dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P | - dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P; - - tegra_das_write(addr, reg); - - return 0; -} -EXPORT_SYMBOL_GPL(tegra_das_connect_dac_to_dap); - -#ifdef CONFIG_DEBUG_FS -static int tegra_das_show(struct seq_file *s, void *unused) -{ - int i; - u32 addr; - u32 reg; - - for (i = 0; i < TEGRA_DAS_DAP_CTRL_SEL_COUNT; i++) { - addr = TEGRA_DAS_DAP_CTRL_SEL + - (i * TEGRA_DAS_DAP_CTRL_SEL_STRIDE); - reg = tegra_das_read(addr); - seq_printf(s, "TEGRA_DAS_DAP_CTRL_SEL[%d] = %08x\n", i, reg); - } - - for (i = 0; i < TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT; i++) { - addr = TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL + - (i * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE); - reg = tegra_das_read(addr); - seq_printf(s, "TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL[%d] = %08x\n", - i, reg); - } - - return 0; -} - -static int tegra_das_debug_open(struct inode *inode, struct file *file) -{ - return single_open(file, tegra_das_show, inode->i_private); -} - -static const struct file_operations tegra_das_debug_fops = { - .open = tegra_das_debug_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; - -static void tegra_das_debug_add(struct tegra_das *das) -{ - das->debug = debugfs_create_file(DRV_NAME, S_IRUGO, - snd_soc_debugfs_root, das, - &tegra_das_debug_fops); -} - -static void tegra_das_debug_remove(struct tegra_das *das) -{ - if (das->debug) - debugfs_remove(das->debug); -} -#else -static inline void tegra_das_debug_add(struct tegra_das *das) -{ -} - -static inline void tegra_das_debug_remove(struct tegra_das *das) -{ -} -#endif - -static int __devinit tegra_das_probe(struct platform_device *pdev) -{ - struct resource *res, *region; - int ret = 0; - - if (das) - return -ENODEV; - - das = devm_kzalloc(&pdev->dev, sizeof(struct tegra_das), GFP_KERNEL); - if (!das) { - dev_err(&pdev->dev, "Can't allocate tegra_das\n"); - ret = -ENOMEM; - goto err; - } - das->dev = &pdev->dev; - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err; - } - - region = devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), pdev->name); - if (!region) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err; - } - - das->regs = devm_ioremap(&pdev->dev, res->start, resource_size(res)); - if (!das->regs) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; - goto err; - } - - ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1, - TEGRA_DAS_DAP_SEL_DAC1); - if (ret) { - dev_err(&pdev->dev, "Can't set up DAS DAP connection\n"); - goto err; - } - ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1, - TEGRA_DAS_DAC_SEL_DAP1); - if (ret) { - dev_err(&pdev->dev, "Can't set up DAS DAC connection\n"); - goto err; - } - - tegra_das_debug_add(das); - - platform_set_drvdata(pdev, das); - - return 0; - -err: - das = NULL; - return ret; -} - -static int __devexit tegra_das_remove(struct platform_device *pdev) -{ - if (!das) - return -ENODEV; - - tegra_das_debug_remove(das); - - das = NULL; - - return 0; -} - -static const struct of_device_id tegra_das_of_match[] __devinitconst = { - { .compatible = "nvidia,tegra20-das", }, - {}, -}; - -static struct platform_driver tegra_das_driver = { - .probe = tegra_das_probe, - .remove = __devexit_p(tegra_das_remove), - .driver = { - .name = DRV_NAME, - .owner = THIS_MODULE, - .of_match_table = tegra_das_of_match, - }, -}; -module_platform_driver(tegra_das_driver); - -MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); -MODULE_DESCRIPTION("Tegra DAS driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:" DRV_NAME); -MODULE_DEVICE_TABLE(of, tegra_das_of_match); diff --git a/sound/soc/tegra/tegra_das.h b/sound/soc/tegra/tegra_das.h deleted file mode 100644 index 2c96c7b3c459..000000000000 --- a/sound/soc/tegra/tegra_das.h +++ /dev/null @@ -1,135 +0,0 @@ -/* - * tegra_das.h - Definitions for Tegra DAS driver - * - * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#ifndef __TEGRA_DAS_H__ -#define __TEGRA_DAS_H__ - -/* Register TEGRA_DAS_DAP_CTRL_SEL */ -#define TEGRA_DAS_DAP_CTRL_SEL 0x00 -#define TEGRA_DAS_DAP_CTRL_SEL_COUNT 5 -#define TEGRA_DAS_DAP_CTRL_SEL_STRIDE 4 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P 31 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_S 1 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P 30 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_S 1 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P 29 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_S 1 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P 0 -#define TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_S 5 - -/* Values for field TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL */ -#define TEGRA_DAS_DAP_SEL_DAC1 0 -#define TEGRA_DAS_DAP_SEL_DAC2 1 -#define TEGRA_DAS_DAP_SEL_DAC3 2 -#define TEGRA_DAS_DAP_SEL_DAP1 16 -#define TEGRA_DAS_DAP_SEL_DAP2 17 -#define TEGRA_DAS_DAP_SEL_DAP3 18 -#define TEGRA_DAS_DAP_SEL_DAP4 19 -#define TEGRA_DAS_DAP_SEL_DAP5 20 - -/* Register TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL */ -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL 0x40 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT 3 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE 4 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P 28 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_S 4 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P 24 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_S 4 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P 0 -#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_S 4 - -/* - * Values for: - * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL - * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL - * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL - */ -#define TEGRA_DAS_DAC_SEL_DAP1 0 -#define TEGRA_DAS_DAC_SEL_DAP2 1 -#define TEGRA_DAS_DAC_SEL_DAP3 2 -#define TEGRA_DAS_DAC_SEL_DAP4 3 -#define TEGRA_DAS_DAC_SEL_DAP5 4 - -/* - * Names/IDs of the DACs/DAPs. - */ - -#define TEGRA_DAS_DAP_ID_1 0 -#define TEGRA_DAS_DAP_ID_2 1 -#define TEGRA_DAS_DAP_ID_3 2 -#define TEGRA_DAS_DAP_ID_4 3 -#define TEGRA_DAS_DAP_ID_5 4 - -#define TEGRA_DAS_DAC_ID_1 0 -#define TEGRA_DAS_DAC_ID_2 1 -#define TEGRA_DAS_DAC_ID_3 2 - -struct tegra_das { - struct device *dev; - void __iomem *regs; - struct dentry *debug; -}; - -/* - * Terminology: - * DAS: Digital audio switch (HW module controlled by this driver) - * DAP: Digital audio port (port/pins on Tegra device) - * DAC: Digital audio controller (e.g. I2S or AC97 controller elsewhere) - * - * The Tegra DAS is a mux/cross-bar which can connect each DAP to a specific - * DAC, or another DAP. When DAPs are connected, one must be the master and - * one the slave. Each DAC allows selection of a specific DAP for input, to - * cater for the case where N DAPs are connected to 1 DAC for broadcast - * output. - * - * This driver is dumb; no attempt is made to ensure that a valid routing - * configuration is programmed. - */ - -/* - * Connect a DAP to to a DAC - * dap_id: DAP to connect: TEGRA_DAS_DAP_ID_* - * dac_sel: DAC to connect to: TEGRA_DAS_DAP_SEL_DAC* - */ -extern int tegra_das_connect_dap_to_dac(int dap_id, int dac_sel); - -/* - * Connect a DAP to to another DAP - * dap_id: DAP to connect: TEGRA_DAS_DAP_ID_* - * other_dap_sel: DAP to connect to: TEGRA_DAS_DAP_SEL_DAP* - * master: Is this DAP the master (1) or slave (0) - * sdata1rx: Is this DAP's SDATA1 pin RX (1) or TX (0) - * sdata2rx: Is this DAP's SDATA2 pin RX (1) or TX (0) - */ -extern int tegra_das_connect_dap_to_dap(int dap_id, int other_dap_sel, - int master, int sdata1rx, - int sdata2rx); - -/* - * Connect a DAC's input to a DAP - * (DAC outputs are selected by the DAP) - * dac_id: DAC ID to connect: TEGRA_DAS_DAC_ID_* - * dap_sel: DAP to receive input from: TEGRA_DAS_DAC_SEL_DAP* - */ -extern int tegra_das_connect_dac_to_dap(int dac_id, int dap_sel); - -#endif diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c deleted file mode 100644 index e53349912b2e..000000000000 --- a/sound/soc/tegra/tegra_i2s.c +++ /dev/null @@ -1,459 +0,0 @@ -/* - * tegra_i2s.c - Tegra I2S driver - * - * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. - * - * Based on code copyright/by: - * - * Copyright (c) 2009-2010, NVIDIA Corporation. - * Scott Peterson <speterson@nvidia.com> - * - * Copyright (C) 2010 Google, Inc. - * Iliyan Malchev <malchev@google.com> - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include <linux/clk.h> -#include <linux/module.h> -#include <linux/debugfs.h> -#include <linux/device.h> -#include <linux/platform_device.h> -#include <linux/seq_file.h> -#include <linux/slab.h> -#include <linux/io.h> -#include <linux/of.h> -#include <mach/iomap.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include "tegra_i2s.h" - -#define DRV_NAME "tegra-i2s" - -static inline void tegra_i2s_write(struct tegra_i2s *i2s, u32 reg, u32 val) -{ - __raw_writel(val, i2s->regs + reg); -} - -static inline u32 tegra_i2s_read(struct tegra_i2s *i2s, u32 reg) -{ - return __raw_readl(i2s->regs + reg); -} - -#ifdef CONFIG_DEBUG_FS -static int tegra_i2s_show(struct seq_file *s, void *unused) -{ -#define REG(r) { r, #r } - static const struct { - int offset; - const char *name; - } regs[] = { - REG(TEGRA_I2S_CTRL), - REG(TEGRA_I2S_STATUS), - REG(TEGRA_I2S_TIMING), - REG(TEGRA_I2S_FIFO_SCR), - REG(TEGRA_I2S_PCM_CTRL), - REG(TEGRA_I2S_NW_CTRL), - REG(TEGRA_I2S_TDM_CTRL), - REG(TEGRA_I2S_TDM_TX_RX_CTRL), - }; -#undef REG - - struct tegra_i2s *i2s = s->private; - int i; - - clk_enable(i2s->clk_i2s); - - for (i = 0; i < ARRAY_SIZE(regs); i++) { - u32 val = tegra_i2s_read(i2s, regs[i].offset); - seq_printf(s, "%s = %08x\n", regs[i].name, val); - } - - clk_disable(i2s->clk_i2s); - - return 0; -} - -static int tegra_i2s_debug_open(struct inode *inode, struct file *file) -{ - return single_open(file, tegra_i2s_show, inode->i_private); -} - -static const struct file_operations tegra_i2s_debug_fops = { - .open = tegra_i2s_debug_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; - -static void tegra_i2s_debug_add(struct tegra_i2s *i2s) -{ - i2s->debug = debugfs_create_file(i2s->dai.name, S_IRUGO, - snd_soc_debugfs_root, i2s, - &tegra_i2s_debug_fops); -} - -static void tegra_i2s_debug_remove(struct tegra_i2s *i2s) -{ - if (i2s->debug) - debugfs_remove(i2s->debug); -} -#else -static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s) -{ -} - -static inline void tegra_i2s_debug_remove(struct tegra_i2s *i2s) -{ -} -#endif - -static int tegra_i2s_set_fmt(struct snd_soc_dai *dai, - unsigned int fmt) -{ - struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - break; - default: - return -EINVAL; - } - - i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_MASTER_ENABLE; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_MASTER_ENABLE; - break; - case SND_SOC_DAIFMT_CBM_CFM: - break; - default: - return -EINVAL; - } - - i2s->reg_ctrl &= ~(TEGRA_I2S_CTRL_BIT_FORMAT_MASK | - TEGRA_I2S_CTRL_LRCK_MASK); - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_DSP_A: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_DSP; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; - break; - case SND_SOC_DAIFMT_DSP_B: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_DSP; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_R_LOW; - break; - case SND_SOC_DAIFMT_I2S: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_I2S; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; - break; - case SND_SOC_DAIFMT_RIGHT_J: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_RJM; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; - break; - case SND_SOC_DAIFMT_LEFT_J: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_LJM; - i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; - break; - default: - return -EINVAL; - } - - return 0; -} - -static int tegra_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct device *dev = substream->pcm->card->dev; - struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); - u32 reg; - int ret, sample_size, srate, i2sclock, bitcnt; - - i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_BIT_SIZE_MASK; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_16; - sample_size = 16; - break; - case SNDRV_PCM_FORMAT_S24_LE: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_24; - sample_size = 24; - break; - case SNDRV_PCM_FORMAT_S32_LE: - i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_32; - sample_size = 32; - break; - default: - return -EINVAL; - } - - srate = params_rate(params); - - /* Final "* 2" required by Tegra hardware */ - i2sclock = srate * params_channels(params) * sample_size * 2; - - ret = clk_set_rate(i2s->clk_i2s, i2sclock); - if (ret) { - dev_err(dev, "Can't set I2S clock rate: %d\n", ret); - return ret; - } - - bitcnt = (i2sclock / (2 * srate)) - 1; - if (bitcnt < 0 || bitcnt > TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US) - return -EINVAL; - reg = bitcnt << TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT; - - if (i2sclock % (2 * srate)) - reg |= TEGRA_I2S_TIMING_NON_SYM_ENABLE; - - if (!i2s->clk_refs) - clk_enable(i2s->clk_i2s); - - tegra_i2s_write(i2s, TEGRA_I2S_TIMING, reg); - - tegra_i2s_write(i2s, TEGRA_I2S_FIFO_SCR, - TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS | - TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS); - - if (!i2s->clk_refs) - clk_disable(i2s->clk_i2s); - - return 0; -} - -static void tegra_i2s_start_playback(struct tegra_i2s *i2s) -{ - i2s->reg_ctrl |= TEGRA_I2S_CTRL_FIFO1_ENABLE; - tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); -} - -static void tegra_i2s_stop_playback(struct tegra_i2s *i2s) -{ - i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_FIFO1_ENABLE; - tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); -} - -static void tegra_i2s_start_capture(struct tegra_i2s *i2s) -{ - i2s->reg_ctrl |= TEGRA_I2S_CTRL_FIFO2_ENABLE; - tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); -} - -static void tegra_i2s_stop_capture(struct tegra_i2s *i2s) -{ - i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_FIFO2_ENABLE; - tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); -} - -static int tegra_i2s_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - case SNDRV_PCM_TRIGGER_RESUME: - if (!i2s->clk_refs) - clk_enable(i2s->clk_i2s); - i2s->clk_refs++; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - tegra_i2s_start_playback(i2s); - else - tegra_i2s_start_capture(i2s); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - case SNDRV_PCM_TRIGGER_SUSPEND: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - tegra_i2s_stop_playback(i2s); - else - tegra_i2s_stop_capture(i2s); - i2s->clk_refs--; - if (!i2s->clk_refs) - clk_disable(i2s->clk_i2s); - break; - default: - return -EINVAL; - } - - return 0; -} - -static int tegra_i2s_probe(struct snd_soc_dai *dai) -{ - struct tegra_i2s * i2s = snd_soc_dai_get_drvdata(dai); - - dai->capture_dma_data = &i2s->capture_dma_data; - dai->playback_dma_data = &i2s->playback_dma_data; - - return 0; -} - -static const struct snd_soc_dai_ops tegra_i2s_dai_ops = { - .set_fmt = tegra_i2s_set_fmt, - .hw_params = tegra_i2s_hw_params, - .trigger = tegra_i2s_trigger, -}; - -static const struct snd_soc_dai_driver tegra_i2s_dai_template = { - .probe = tegra_i2s_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = &tegra_i2s_dai_ops, - .symmetric_rates = 1, -}; - -static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) -{ - struct tegra_i2s * i2s; - struct resource *mem, *memregion, *dmareq; - u32 of_dma[2]; - u32 dma_ch; - int ret; - - i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra_i2s), GFP_KERNEL); - if (!i2s) { - dev_err(&pdev->dev, "Can't allocate tegra_i2s\n"); - ret = -ENOMEM; - goto err; - } - dev_set_drvdata(&pdev->dev, i2s); - - i2s->dai = tegra_i2s_dai_template; - i2s->dai.name = dev_name(&pdev->dev); - - i2s->clk_i2s = clk_get(&pdev->dev, NULL); - if (IS_ERR(i2s->clk_i2s)) { - dev_err(&pdev->dev, "Can't retrieve i2s clock\n"); - ret = PTR_ERR(i2s->clk_i2s); - goto err; - } - - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem) { - dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err_clk_put; - } - - dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmareq) { - if (of_property_read_u32_array(pdev->dev.of_node, - "nvidia,dma-request-selector", - of_dma, 2) < 0) { - dev_err(&pdev->dev, "No DMA resource\n"); - ret = -ENODEV; - goto err_clk_put; - } - dma_ch = of_dma[1]; - } else { - dma_ch = dmareq->start; - } - - memregion = devm_request_mem_region(&pdev->dev, mem->start, - resource_size(mem), DRV_NAME); - if (!memregion) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err_clk_put; - } - - i2s->regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!i2s->regs) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; - goto err_clk_put; - } - - i2s->capture_dma_data.addr = mem->start + TEGRA_I2S_FIFO2; - i2s->capture_dma_data.wrap = 4; - i2s->capture_dma_data.width = 32; - i2s->capture_dma_data.req_sel = dma_ch; - - i2s->playback_dma_data.addr = mem->start + TEGRA_I2S_FIFO1; - i2s->playback_dma_data.wrap = 4; - i2s->playback_dma_data.width = 32; - i2s->playback_dma_data.req_sel = dma_ch; - - i2s->reg_ctrl = TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED; - - ret = snd_soc_register_dai(&pdev->dev, &i2s->dai); - if (ret) { - dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); - ret = -ENOMEM; - goto err_clk_put; - } - - tegra_i2s_debug_add(i2s); - - return 0; - -err_clk_put: - clk_put(i2s->clk_i2s); -err: - return ret; -} - -static int __devexit tegra_i2s_platform_remove(struct platform_device *pdev) -{ - struct tegra_i2s *i2s = dev_get_drvdata(&pdev->dev); - - snd_soc_unregister_dai(&pdev->dev); - - tegra_i2s_debug_remove(i2s); - - clk_put(i2s->clk_i2s); - - return 0; -} - -static const struct of_device_id tegra_i2s_of_match[] __devinitconst = { - { .compatible = "nvidia,tegra20-i2s", }, - {}, -}; - -static struct platform_driver tegra_i2s_driver = { - .driver = { - .name = DRV_NAME, - .owner = THIS_MODULE, - .of_match_table = tegra_i2s_of_match, - }, - .probe = tegra_i2s_platform_probe, - .remove = __devexit_p(tegra_i2s_platform_remove), -}; -module_platform_driver(tegra_i2s_driver); - -MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); -MODULE_DESCRIPTION("Tegra I2S ASoC driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:" DRV_NAME); -MODULE_DEVICE_TABLE(of, tegra_i2s_of_match); diff --git a/sound/soc/tegra/tegra_i2s.h b/sound/soc/tegra/tegra_i2s.h deleted file mode 100644 index 15ce1e2e8bde..000000000000 --- a/sound/soc/tegra/tegra_i2s.h +++ /dev/null @@ -1,166 +0,0 @@ -/* - * tegra_i2s.h - Definitions for Tegra I2S driver - * - * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. - * - * Based on code copyright/by: - * - * Copyright (c) 2009-2010, NVIDIA Corporation. - * Scott Peterson <speterson@nvidia.com> - * - * Copyright (C) 2010 Google, Inc. - * Iliyan Malchev <malchev@google.com> - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#ifndef __TEGRA_I2S_H__ -#define __TEGRA_I2S_H__ - -#include "tegra_pcm.h" - -/* Register offsets from TEGRA_I2S1_BASE and TEGRA_I2S2_BASE */ - -#define TEGRA_I2S_CTRL 0x00 -#define TEGRA_I2S_STATUS 0x04 -#define TEGRA_I2S_TIMING 0x08 -#define TEGRA_I2S_FIFO_SCR 0x0c -#define TEGRA_I2S_PCM_CTRL 0x10 -#define TEGRA_I2S_NW_CTRL 0x14 -#define TEGRA_I2S_TDM_CTRL 0x20 -#define TEGRA_I2S_TDM_TX_RX_CTRL 0x24 -#define TEGRA_I2S_FIFO1 0x40 -#define TEGRA_I2S_FIFO2 0x80 - -/* Fields in TEGRA_I2S_CTRL */ - -#define TEGRA_I2S_CTRL_FIFO2_TX_ENABLE (1 << 30) -#define TEGRA_I2S_CTRL_FIFO1_ENABLE (1 << 29) -#define TEGRA_I2S_CTRL_FIFO2_ENABLE (1 << 28) -#define TEGRA_I2S_CTRL_FIFO1_RX_ENABLE (1 << 27) -#define TEGRA_I2S_CTRL_FIFO_LPBK_ENABLE (1 << 26) -#define TEGRA_I2S_CTRL_MASTER_ENABLE (1 << 25) - -#define TEGRA_I2S_LRCK_LEFT_LOW 0 -#define TEGRA_I2S_LRCK_RIGHT_LOW 1 - -#define TEGRA_I2S_CTRL_LRCK_SHIFT 24 -#define TEGRA_I2S_CTRL_LRCK_MASK (1 << TEGRA_I2S_CTRL_LRCK_SHIFT) -#define TEGRA_I2S_CTRL_LRCK_L_LOW (TEGRA_I2S_LRCK_LEFT_LOW << TEGRA_I2S_CTRL_LRCK_SHIFT) -#define TEGRA_I2S_CTRL_LRCK_R_LOW (TEGRA_I2S_LRCK_RIGHT_LOW << TEGRA_I2S_CTRL_LRCK_SHIFT) - -#define TEGRA_I2S_BIT_FORMAT_I2S 0 -#define TEGRA_I2S_BIT_FORMAT_RJM 1 -#define TEGRA_I2S_BIT_FORMAT_LJM 2 -#define TEGRA_I2S_BIT_FORMAT_DSP 3 - -#define TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT 10 -#define TEGRA_I2S_CTRL_BIT_FORMAT_MASK (3 << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_BIT_FORMAT_I2S (TEGRA_I2S_BIT_FORMAT_I2S << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_BIT_FORMAT_RJM (TEGRA_I2S_BIT_FORMAT_RJM << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_BIT_FORMAT_LJM (TEGRA_I2S_BIT_FORMAT_LJM << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_BIT_FORMAT_DSP (TEGRA_I2S_BIT_FORMAT_DSP << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) - -#define TEGRA_I2S_BIT_SIZE_16 0 -#define TEGRA_I2S_BIT_SIZE_20 1 -#define TEGRA_I2S_BIT_SIZE_24 2 -#define TEGRA_I2S_BIT_SIZE_32 3 - -#define TEGRA_I2S_CTRL_BIT_SIZE_SHIFT 8 -#define TEGRA_I2S_CTRL_BIT_SIZE_MASK (3 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) -#define TEGRA_I2S_CTRL_BIT_SIZE_16 (TEGRA_I2S_BIT_SIZE_16 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) -#define TEGRA_I2S_CTRL_BIT_SIZE_20 (TEGRA_I2S_BIT_SIZE_20 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) -#define TEGRA_I2S_CTRL_BIT_SIZE_24 (TEGRA_I2S_BIT_SIZE_24 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) -#define TEGRA_I2S_CTRL_BIT_SIZE_32 (TEGRA_I2S_BIT_SIZE_32 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) - -#define TEGRA_I2S_FIFO_16_LSB 0 -#define TEGRA_I2S_FIFO_20_LSB 1 -#define TEGRA_I2S_FIFO_24_LSB 2 -#define TEGRA_I2S_FIFO_32 3 -#define TEGRA_I2S_FIFO_PACKED 7 - -#define TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT 4 -#define TEGRA_I2S_CTRL_FIFO_FORMAT_MASK (7 << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_16_LSB (TEGRA_I2S_FIFO_16_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_20_LSB (TEGRA_I2S_FIFO_20_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_24_LSB (TEGRA_I2S_FIFO_24_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_32 (TEGRA_I2S_FIFO_32 << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) -#define TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED (TEGRA_I2S_FIFO_PACKED << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) - -#define TEGRA_I2S_CTRL_IE_FIFO1_ERR (1 << 3) -#define TEGRA_I2S_CTRL_IE_FIFO2_ERR (1 << 2) -#define TEGRA_I2S_CTRL_QE_FIFO1 (1 << 1) -#define TEGRA_I2S_CTRL_QE_FIFO2 (1 << 0) - -/* Fields in TEGRA_I2S_STATUS */ - -#define TEGRA_I2S_STATUS_FIFO1_RDY (1 << 31) -#define TEGRA_I2S_STATUS_FIFO2_RDY (1 << 30) -#define TEGRA_I2S_STATUS_FIFO1_BSY (1 << 29) -#define TEGRA_I2S_STATUS_FIFO2_BSY (1 << 28) -#define TEGRA_I2S_STATUS_FIFO1_ERR (1 << 3) -#define TEGRA_I2S_STATUS_FIFO2_ERR (1 << 2) -#define TEGRA_I2S_STATUS_QS_FIFO1 (1 << 1) -#define TEGRA_I2S_STATUS_QS_FIFO2 (1 << 0) - -/* Fields in TEGRA_I2S_TIMING */ - -#define TEGRA_I2S_TIMING_NON_SYM_ENABLE (1 << 12) -#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 -#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff -#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) - -/* Fields in TEGRA_I2S_FIFO_SCR */ - -#define TEGRA_I2S_FIFO_SCR_FIFO2_FULL_EMPTY_COUNT_SHIFT 24 -#define TEGRA_I2S_FIFO_SCR_FIFO1_FULL_EMPTY_COUNT_SHIFT 16 -#define TEGRA_I2S_FIFO_SCR_FIFO_FULL_EMPTY_COUNT_MASK 0x3f - -#define TEGRA_I2S_FIFO_SCR_FIFO2_CLR (1 << 12) -#define TEGRA_I2S_FIFO_SCR_FIFO1_CLR (1 << 8) - -#define TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT 0 -#define TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS 1 -#define TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS 2 -#define TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS 3 - -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT 4 -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_MASK (3 << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_ONE_SLOT (TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_EIGHT_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_TWELVE_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) - -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT 0 -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_MASK (3 << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_ONE_SLOT (TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_EIGHT_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) -#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_TWELVE_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) - -struct tegra_i2s { - struct snd_soc_dai_driver dai; - struct clk *clk_i2s; - int clk_refs; - struct tegra_pcm_dma_params capture_dma_data; - struct tegra_pcm_dma_params playback_dma_data; - void __iomem *regs; - struct dentry *debug; - u32 reg_ctrl; -}; - -#endif diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 8b4457137c7c..5658bcec1931 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -2,7 +2,7 @@ * tegra_pcm.c - Tegra PCM driver * * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010,2012 - NVIDIA, Inc. * * Based on code copyright/by: * @@ -29,18 +29,17 @@ * */ -#include <linux/module.h> #include <linux/dma-mapping.h> +#include <linux/module.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> #include "tegra_pcm.h" -#define DRV_NAME "tegra-pcm-audio" - static const struct snd_pcm_hardware tegra_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -58,6 +57,7 @@ static const struct snd_pcm_hardware tegra_pcm_hardware = { .fifo_size = 4, }; +#if defined(CONFIG_TEGRA_SYSTEM_DMA) static void tegra_pcm_queue_dma(struct tegra_runtime_data *prtd) { struct snd_pcm_substream *substream = prtd->substream; @@ -287,6 +287,119 @@ static struct snd_pcm_ops tegra_pcm_ops = { .pointer = tegra_pcm_pointer, .mmap = tegra_pcm_mmap, }; +#else +static int tegra_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->platform->dev; + int ret; + + /* Set HW params now that initialization is complete */ + snd_soc_set_runtime_hwparams(substream, &tegra_pcm_hardware); + + ret = snd_dmaengine_pcm_open(substream, NULL, NULL); + if (ret) { + dev_err(dev, "dmaengine pcm open failed with err %d\n", ret); + return ret; + } + + return 0; +} + +static int tegra_pcm_close(struct snd_pcm_substream *substream) +{ + snd_dmaengine_pcm_close(substream); + return 0; +} + +static int tegra_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->platform->dev; + struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); + struct tegra_pcm_dma_params *dmap; + struct dma_slave_config slave_config; + int ret; + + dmap = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + ret = snd_hwparams_to_dma_slave_config(substream, params, + &slave_config); + if (ret) { + dev_err(dev, "hw params config failed with err %d\n", ret); + return ret; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + slave_config.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + slave_config.dst_addr = dmap->addr; + slave_config.src_maxburst = 0; + } else { + slave_config.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + slave_config.src_addr = dmap->addr; + slave_config.dst_maxburst = 0; + } + slave_config.slave_id = dmap->req_sel; + + ret = dmaengine_slave_config(chan, &slave_config); + if (ret < 0) { + dev_err(dev, "dma slave config failed with err %d\n", ret); + return ret; + } + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + return 0; +} + +static int tegra_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + return 0; +} + +static int tegra_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + return snd_dmaengine_pcm_trigger(substream, + SNDRV_PCM_TRIGGER_START); + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + return snd_dmaengine_pcm_trigger(substream, + SNDRV_PCM_TRIGGER_STOP); + default: + return -EINVAL; + } + return 0; +} + +static int tegra_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops tegra_pcm_ops = { + .open = tegra_pcm_open, + .close = tegra_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = tegra_pcm_hw_params, + .hw_free = tegra_pcm_hw_free, + .trigger = tegra_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, + .mmap = tegra_pcm_mmap, +}; +#endif static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { @@ -372,28 +485,18 @@ static struct snd_soc_platform_driver tegra_pcm_platform = { .pcm_free = tegra_pcm_free, }; -static int __devinit tegra_pcm_platform_probe(struct platform_device *pdev) +int __devinit tegra_pcm_platform_register(struct device *dev) { - return snd_soc_register_platform(&pdev->dev, &tegra_pcm_platform); + return snd_soc_register_platform(dev, &tegra_pcm_platform); } +EXPORT_SYMBOL_GPL(tegra_pcm_platform_register); -static int __devexit tegra_pcm_platform_remove(struct platform_device *pdev) +void __devexit tegra_pcm_platform_unregister(struct device *dev) { - snd_soc_unregister_platform(&pdev->dev); - return 0; + snd_soc_unregister_platform(dev); } - -static struct platform_driver tegra_pcm_driver = { - .driver = { - .name = DRV_NAME, - .owner = THIS_MODULE, - }, - .probe = tegra_pcm_platform_probe, - .remove = __devexit_p(tegra_pcm_platform_remove), -}; -module_platform_driver(tegra_pcm_driver); +EXPORT_SYMBOL_GPL(tegra_pcm_platform_unregister); MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); MODULE_DESCRIPTION("Tegra PCM ASoC driver"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra_pcm.h b/sound/soc/tegra/tegra_pcm.h index dbb90339fe0d..a3a450352dcf 100644 --- a/sound/soc/tegra/tegra_pcm.h +++ b/sound/soc/tegra/tegra_pcm.h @@ -2,7 +2,7 @@ * tegra_pcm.h - Definitions for Tegra PCM driver * * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010,2012 - NVIDIA, Inc. * * Based on code copyright/by: * @@ -40,6 +40,7 @@ struct tegra_pcm_dma_params { unsigned long req_sel; }; +#if defined(CONFIG_TEGRA_SYSTEM_DMA) struct tegra_runtime_data { struct snd_pcm_substream *substream; spinlock_t lock; @@ -51,5 +52,9 @@ struct tegra_runtime_data { struct tegra_dma_req dma_req[2]; struct tegra_dma_channel *dma_chan; }; +#endif + +int tegra_pcm_platform_register(struct device *dev); +void tegra_pcm_platform_unregister(struct device *dev); #endif diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c deleted file mode 100644 index 9ff2c601445f..000000000000 --- a/sound/soc/tegra/tegra_spdif.c +++ /dev/null @@ -1,364 +0,0 @@ -/* - * tegra_spdif.c - Tegra SPDIF driver - * - * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2011 - NVIDIA, Inc. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include <linux/clk.h> -#include <linux/module.h> -#include <linux/debugfs.h> -#include <linux/device.h> -#include <linux/platform_device.h> -#include <linux/seq_file.h> -#include <linux/slab.h> -#include <linux/io.h> -#include <mach/iomap.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include "tegra_spdif.h" - -#define DRV_NAME "tegra-spdif" - -static inline void tegra_spdif_write(struct tegra_spdif *spdif, u32 reg, - u32 val) -{ - __raw_writel(val, spdif->regs + reg); -} - -static inline u32 tegra_spdif_read(struct tegra_spdif *spdif, u32 reg) -{ - return __raw_readl(spdif->regs + reg); -} - -#ifdef CONFIG_DEBUG_FS -static int tegra_spdif_show(struct seq_file *s, void *unused) -{ -#define REG(r) { r, #r } - static const struct { - int offset; - const char *name; - } regs[] = { - REG(TEGRA_SPDIF_CTRL), - REG(TEGRA_SPDIF_STATUS), - REG(TEGRA_SPDIF_STROBE_CTRL), - REG(TEGRA_SPDIF_DATA_FIFO_CSR), - REG(TEGRA_SPDIF_CH_STA_RX_A), - REG(TEGRA_SPDIF_CH_STA_RX_B), - REG(TEGRA_SPDIF_CH_STA_RX_C), - REG(TEGRA_SPDIF_CH_STA_RX_D), - REG(TEGRA_SPDIF_CH_STA_RX_E), - REG(TEGRA_SPDIF_CH_STA_RX_F), - REG(TEGRA_SPDIF_CH_STA_TX_A), - REG(TEGRA_SPDIF_CH_STA_TX_B), - REG(TEGRA_SPDIF_CH_STA_TX_C), - REG(TEGRA_SPDIF_CH_STA_TX_D), - REG(TEGRA_SPDIF_CH_STA_TX_E), - REG(TEGRA_SPDIF_CH_STA_TX_F), - }; -#undef REG - - struct tegra_spdif *spdif = s->private; - int i; - - clk_enable(spdif->clk_spdif_out); - - for (i = 0; i < ARRAY_SIZE(regs); i++) { - u32 val = tegra_spdif_read(spdif, regs[i].offset); - seq_printf(s, "%s = %08x\n", regs[i].name, val); - } - - clk_disable(spdif->clk_spdif_out); - - return 0; -} - -static int tegra_spdif_debug_open(struct inode *inode, struct file *file) -{ - return single_open(file, tegra_spdif_show, inode->i_private); -} - -static const struct file_operations tegra_spdif_debug_fops = { - .open = tegra_spdif_debug_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; - -static void tegra_spdif_debug_add(struct tegra_spdif *spdif) -{ - spdif->debug = debugfs_create_file(DRV_NAME, S_IRUGO, - snd_soc_debugfs_root, spdif, - &tegra_spdif_debug_fops); -} - -static void tegra_spdif_debug_remove(struct tegra_spdif *spdif) -{ - if (spdif->debug) - debugfs_remove(spdif->debug); -} -#else -static inline void tegra_spdif_debug_add(struct tegra_spdif *spdif) -{ -} - -static inline void tegra_spdif_debug_remove(struct tegra_spdif *spdif) -{ -} -#endif - -static int tegra_spdif_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct device *dev = substream->pcm->card->dev; - struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); - int ret, spdifclock; - - spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_PACK; - spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_BIT_MODE_MASK; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_PACK; - spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_BIT_MODE_16BIT; - break; - default: - return -EINVAL; - } - - switch (params_rate(params)) { - case 32000: - spdifclock = 4096000; - break; - case 44100: - spdifclock = 5644800; - break; - case 48000: - spdifclock = 6144000; - break; - case 88200: - spdifclock = 11289600; - break; - case 96000: - spdifclock = 12288000; - break; - case 176400: - spdifclock = 22579200; - break; - case 192000: - spdifclock = 24576000; - break; - default: - return -EINVAL; - } - - ret = clk_set_rate(spdif->clk_spdif_out, spdifclock); - if (ret) { - dev_err(dev, "Can't set SPDIF clock rate: %d\n", ret); - return ret; - } - - return 0; -} - -static void tegra_spdif_start_playback(struct tegra_spdif *spdif) -{ - spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_TX_EN; - tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl); -} - -static void tegra_spdif_stop_playback(struct tegra_spdif *spdif) -{ - spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_TX_EN; - tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl); -} - -static int tegra_spdif_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - case SNDRV_PCM_TRIGGER_RESUME: - if (!spdif->clk_refs) - clk_enable(spdif->clk_spdif_out); - spdif->clk_refs++; - tegra_spdif_start_playback(spdif); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - case SNDRV_PCM_TRIGGER_SUSPEND: - tegra_spdif_stop_playback(spdif); - spdif->clk_refs--; - if (!spdif->clk_refs) - clk_disable(spdif->clk_spdif_out); - break; - default: - return -EINVAL; - } - - return 0; -} - -static int tegra_spdif_probe(struct snd_soc_dai *dai) -{ - struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); - - dai->capture_dma_data = NULL; - dai->playback_dma_data = &spdif->playback_dma_data; - - return 0; -} - -static const struct snd_soc_dai_ops tegra_spdif_dai_ops = { - .hw_params = tegra_spdif_hw_params, - .trigger = tegra_spdif_trigger, -}; - -static struct snd_soc_dai_driver tegra_spdif_dai = { - .name = DRV_NAME, - .probe = tegra_spdif_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = &tegra_spdif_dai_ops, -}; - -static __devinit int tegra_spdif_platform_probe(struct platform_device *pdev) -{ - struct tegra_spdif *spdif; - struct resource *mem, *memregion, *dmareq; - int ret; - - spdif = kzalloc(sizeof(struct tegra_spdif), GFP_KERNEL); - if (!spdif) { - dev_err(&pdev->dev, "Can't allocate tegra_spdif\n"); - ret = -ENOMEM; - goto exit; - } - dev_set_drvdata(&pdev->dev, spdif); - - spdif->clk_spdif_out = clk_get(&pdev->dev, "spdif_out"); - if (IS_ERR(spdif->clk_spdif_out)) { - pr_err("Can't retrieve spdif clock\n"); - ret = PTR_ERR(spdif->clk_spdif_out); - goto err_free; - } - - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem) { - dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err_clk_put; - } - - dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmareq) { - dev_err(&pdev->dev, "No DMA resource\n"); - ret = -ENODEV; - goto err_clk_put; - } - - memregion = request_mem_region(mem->start, resource_size(mem), - DRV_NAME); - if (!memregion) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err_clk_put; - } - - spdif->regs = ioremap(mem->start, resource_size(mem)); - if (!spdif->regs) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; - goto err_release; - } - - spdif->playback_dma_data.addr = mem->start + TEGRA_SPDIF_DATA_OUT; - spdif->playback_dma_data.wrap = 4; - spdif->playback_dma_data.width = 32; - spdif->playback_dma_data.req_sel = dmareq->start; - - ret = snd_soc_register_dai(&pdev->dev, &tegra_spdif_dai); - if (ret) { - dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); - ret = -ENOMEM; - goto err_unmap; - } - - tegra_spdif_debug_add(spdif); - - return 0; - -err_unmap: - iounmap(spdif->regs); -err_release: - release_mem_region(mem->start, resource_size(mem)); -err_clk_put: - clk_put(spdif->clk_spdif_out); -err_free: - kfree(spdif); -exit: - return ret; -} - -static int __devexit tegra_spdif_platform_remove(struct platform_device *pdev) -{ - struct tegra_spdif *spdif = dev_get_drvdata(&pdev->dev); - struct resource *res; - - snd_soc_unregister_dai(&pdev->dev); - - tegra_spdif_debug_remove(spdif); - - iounmap(spdif->regs); - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(res->start, resource_size(res)); - - clk_put(spdif->clk_spdif_out); - - kfree(spdif); - - return 0; -} - -static struct platform_driver tegra_spdif_driver = { - .driver = { - .name = DRV_NAME, - .owner = THIS_MODULE, - }, - .probe = tegra_spdif_platform_probe, - .remove = __devexit_p(tegra_spdif_platform_remove), -}; - -module_platform_driver(tegra_spdif_driver); - -MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); -MODULE_DESCRIPTION("Tegra SPDIF ASoC driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra_spdif.h b/sound/soc/tegra/tegra_spdif.h deleted file mode 100644 index 2e03db430279..000000000000 --- a/sound/soc/tegra/tegra_spdif.h +++ /dev/null @@ -1,473 +0,0 @@ -/* - * tegra_spdif.h - Definitions for Tegra SPDIF driver - * - * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2011 - NVIDIA, Inc. - * - * Based on code copyright/by: - * Copyright (c) 2008-2009, NVIDIA Corporation - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#ifndef __TEGRA_SPDIF_H__ -#define __TEGRA_SPDIF_H__ - -#include "tegra_pcm.h" - -/* Offsets from TEGRA_SPDIF_BASE */ - -#define TEGRA_SPDIF_CTRL 0x0 -#define TEGRA_SPDIF_STATUS 0x4 -#define TEGRA_SPDIF_STROBE_CTRL 0x8 -#define TEGRA_SPDIF_DATA_FIFO_CSR 0x0C -#define TEGRA_SPDIF_DATA_OUT 0x40 -#define TEGRA_SPDIF_DATA_IN 0x80 -#define TEGRA_SPDIF_CH_STA_RX_A 0x100 -#define TEGRA_SPDIF_CH_STA_RX_B 0x104 -#define TEGRA_SPDIF_CH_STA_RX_C 0x108 -#define TEGRA_SPDIF_CH_STA_RX_D 0x10C -#define TEGRA_SPDIF_CH_STA_RX_E 0x110 -#define TEGRA_SPDIF_CH_STA_RX_F 0x114 -#define TEGRA_SPDIF_CH_STA_TX_A 0x140 -#define TEGRA_SPDIF_CH_STA_TX_B 0x144 -#define TEGRA_SPDIF_CH_STA_TX_C 0x148 -#define TEGRA_SPDIF_CH_STA_TX_D 0x14C -#define TEGRA_SPDIF_CH_STA_TX_E 0x150 -#define TEGRA_SPDIF_CH_STA_TX_F 0x154 -#define TEGRA_SPDIF_USR_STA_RX_A 0x180 -#define TEGRA_SPDIF_USR_DAT_TX_A 0x1C0 - -/* Fields in TEGRA_SPDIF_CTRL */ - -/* Start capturing from 0=right, 1=left channel */ -#define TEGRA_SPDIF_CTRL_CAP_LC (1 << 30) - -/* SPDIF receiver(RX) enable */ -#define TEGRA_SPDIF_CTRL_RX_EN (1 << 29) - -/* SPDIF Transmitter(TX) enable */ -#define TEGRA_SPDIF_CTRL_TX_EN (1 << 28) - -/* Transmit Channel status */ -#define TEGRA_SPDIF_CTRL_TC_EN (1 << 27) - -/* Transmit user Data */ -#define TEGRA_SPDIF_CTRL_TU_EN (1 << 26) - -/* Interrupt on transmit error */ -#define TEGRA_SPDIF_CTRL_IE_TXE (1 << 25) - -/* Interrupt on receive error */ -#define TEGRA_SPDIF_CTRL_IE_RXE (1 << 24) - -/* Interrupt on invalid preamble */ -#define TEGRA_SPDIF_CTRL_IE_P (1 << 23) - -/* Interrupt on "B" preamble */ -#define TEGRA_SPDIF_CTRL_IE_B (1 << 22) - -/* Interrupt when block of channel status received */ -#define TEGRA_SPDIF_CTRL_IE_C (1 << 21) - -/* Interrupt when a valid information unit (IU) is received */ -#define TEGRA_SPDIF_CTRL_IE_U (1 << 20) - -/* Interrupt when RX user FIFO attention level is reached */ -#define TEGRA_SPDIF_CTRL_QE_RU (1 << 19) - -/* Interrupt when TX user FIFO attention level is reached */ -#define TEGRA_SPDIF_CTRL_QE_TU (1 << 18) - -/* Interrupt when RX data FIFO attention level is reached */ -#define TEGRA_SPDIF_CTRL_QE_RX (1 << 17) - -/* Interrupt when TX data FIFO attention level is reached */ -#define TEGRA_SPDIF_CTRL_QE_TX (1 << 16) - -/* Loopback test mode enable */ -#define TEGRA_SPDIF_CTRL_LBK_EN (1 << 15) - -/* - * Pack data mode: - * 0 = Single data (16 bit needs to be padded to match the - * interface data bit size). - * 1 = Packeted left/right channel data into a single word. - */ -#define TEGRA_SPDIF_CTRL_PACK (1 << 14) - -/* - * 00 = 16bit data - * 01 = 20bit data - * 10 = 24bit data - * 11 = raw data - */ -#define TEGRA_SPDIF_BIT_MODE_16BIT 0 -#define TEGRA_SPDIF_BIT_MODE_20BIT 1 -#define TEGRA_SPDIF_BIT_MODE_24BIT 2 -#define TEGRA_SPDIF_BIT_MODE_RAW 3 - -#define TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT 12 -#define TEGRA_SPDIF_CTRL_BIT_MODE_MASK (3 << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) -#define TEGRA_SPDIF_CTRL_BIT_MODE_16BIT (TEGRA_SPDIF_BIT_MODE_16BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) -#define TEGRA_SPDIF_CTRL_BIT_MODE_20BIT (TEGRA_SPDIF_BIT_MODE_20BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) -#define TEGRA_SPDIF_CTRL_BIT_MODE_24BIT (TEGRA_SPDIF_BIT_MODE_24BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) -#define TEGRA_SPDIF_CTRL_BIT_MODE_RAW (TEGRA_SPDIF_BIT_MODE_RAW << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) - -/* Fields in TEGRA_SPDIF_STATUS */ - -/* - * Note: IS_P, IS_B, IS_C, and IS_U are sticky bits. Software must - * write a 1 to the corresponding bit location to clear the status. - */ - -/* - * Receiver(RX) shifter is busy receiving data. - * This bit is asserted when the receiver first locked onto the - * preamble of the data stream after RX_EN is asserted. This bit is - * deasserted when either, - * (a) the end of a frame is reached after RX_EN is deeasserted, or - * (b) the SPDIF data stream becomes inactive. - */ -#define TEGRA_SPDIF_STATUS_RX_BSY (1 << 29) - -/* - * Transmitter(TX) shifter is busy transmitting data. - * This bit is asserted when TX_EN is asserted. - * This bit is deasserted when the end of a frame is reached after - * TX_EN is deasserted. - */ -#define TEGRA_SPDIF_STATUS_TX_BSY (1 << 28) - -/* - * TX is busy shifting out channel status. - * This bit is asserted when both TX_EN and TC_EN are asserted and - * data from CH_STA_TX_A register is loaded into the internal shifter. - * This bit is deasserted when either, - * (a) the end of a frame is reached after TX_EN is deasserted, or - * (b) CH_STA_TX_F register is loaded into the internal shifter. - */ -#define TEGRA_SPDIF_STATUS_TC_BSY (1 << 27) - -/* - * TX User data FIFO busy. - * This bit is asserted when TX_EN and TXU_EN are asserted and - * there's data in the TX user FIFO. This bit is deassert when either, - * (a) the end of a frame is reached after TX_EN is deasserted, or - * (b) there's no data left in the TX user FIFO. - */ -#define TEGRA_SPDIF_STATUS_TU_BSY (1 << 26) - -/* TX FIFO Underrun error status */ -#define TEGRA_SPDIF_STATUS_TX_ERR (1 << 25) - -/* RX FIFO Overrun error status */ -#define TEGRA_SPDIF_STATUS_RX_ERR (1 << 24) - -/* Preamble status: 0=Preamble OK, 1=bad/missing preamble */ -#define TEGRA_SPDIF_STATUS_IS_P (1 << 23) - -/* B-preamble detection status: 0=not detected, 1=B-preamble detected */ -#define TEGRA_SPDIF_STATUS_IS_B (1 << 22) - -/* - * RX channel block data receive status: - * 0=entire block not recieved yet. - * 1=received entire block of channel status, - */ -#define TEGRA_SPDIF_STATUS_IS_C (1 << 21) - -/* RX User Data Valid flag: 1=valid IU detected, 0 = no IU detected. */ -#define TEGRA_SPDIF_STATUS_IS_U (1 << 20) - -/* - * RX User FIFO Status: - * 1=attention level reached, 0=attention level not reached. - */ -#define TEGRA_SPDIF_STATUS_QS_RU (1 << 19) - -/* - * TX User FIFO Status: - * 1=attention level reached, 0=attention level not reached. - */ -#define TEGRA_SPDIF_STATUS_QS_TU (1 << 18) - -/* - * RX Data FIFO Status: - * 1=attention level reached, 0=attention level not reached. - */ -#define TEGRA_SPDIF_STATUS_QS_RX (1 << 17) - -/* - * TX Data FIFO Status: - * 1=attention level reached, 0=attention level not reached. - */ -#define TEGRA_SPDIF_STATUS_QS_TX (1 << 16) - -/* Fields in TEGRA_SPDIF_STROBE_CTRL */ - -/* - * Indicates the approximate number of detected SPDIFIN clocks within a - * bi-phase period. - */ -#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT 16 -#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_MASK (0xff << TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT) - -/* Data strobe mode: 0=Auto-locked 1=Manual locked */ -#define TEGRA_SPDIF_STROBE_CTRL_STROBE (1 << 15) - -/* - * Manual data strobe time within the bi-phase clock period (in terms of - * the number of over-sampling clocks). - */ -#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT 8 -#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_MASK (0x1f << TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT) - -/* - * Manual SPDIFIN bi-phase clock period (in terms of the number of - * over-sampling clocks). - */ -#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT 0 -#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_MASK (0x3f << TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT) - -/* Fields in SPDIF_DATA_FIFO_CSR */ - -/* Clear Receiver User FIFO (RX USR.FIFO) */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_CLR (1 << 31) - -#define TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT 0 -#define TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS 1 -#define TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS 2 -#define TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS 3 - -/* RU FIFO attention level */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT 29 -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_MASK \ - (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU1_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU2_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU3_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU4_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) - -/* Number of RX USR.FIFO levels with valid data. */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT 24 -#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_MASK (0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT) - -/* Clear Transmitter User FIFO (TX USR.FIFO) */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_CLR (1 << 23) - -/* TU FIFO attention level */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT 21 -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_MASK \ - (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU1_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU2_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU3_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU4_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) - -/* Number of TX USR.FIFO levels that could be filled. */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT 16 -#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT) - -/* Clear Receiver Data FIFO (RX DATA.FIFO) */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_CLR (1 << 15) - -#define TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT 0 -#define TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS 1 -#define TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS 2 -#define TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS 3 - -/* RU FIFO attention level */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT 13 -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_MASK \ - (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU1_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU4_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU8_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU12_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) - -/* Number of RX DATA.FIFO levels with valid data. */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT 8 -#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_MASK (0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT) - -/* Clear Transmitter Data FIFO (TX DATA.FIFO) */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_CLR (1 << 7) - -/* TU FIFO attention level */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT 5 -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_MASK \ - (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU1_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU4_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU8_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU12_WORD_FULL \ - (TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) - -/* Number of TX DATA.FIFO levels that could be filled. */ -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT 0 -#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT) - -/* Fields in TEGRA_SPDIF_DATA_OUT */ - -/* - * This register has 5 different formats: - * 16-bit (BIT_MODE=00, PACK=0) - * 20-bit (BIT_MODE=01, PACK=0) - * 24-bit (BIT_MODE=10, PACK=0) - * raw (BIT_MODE=11, PACK=0) - * 16-bit packed (BIT_MODE=00, PACK=1) - */ - -#define TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_16_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_20_MASK (0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_24_MASK (0xffffff << TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_P (1 << 31) -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_C (1 << 30) -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_U (1 << 29) -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_V (1 << 28) - -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT 8 -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_MASK (0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT 4 -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_MASK (0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT 16 -#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT) - -#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT 0 -#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT) - -/* Fields in TEGRA_SPDIF_DATA_IN */ - -/* - * This register has 5 different formats: - * 16-bit (BIT_MODE=00, PACK=0) - * 20-bit (BIT_MODE=01, PACK=0) - * 24-bit (BIT_MODE=10, PACK=0) - * raw (BIT_MODE=11, PACK=0) - * 16-bit packed (BIT_MODE=00, PACK=1) - * - * Bits 31:24 are common to all modes except 16-bit packed - */ - -#define TEGRA_SPDIF_DATA_IN_DATA_P (1 << 31) -#define TEGRA_SPDIF_DATA_IN_DATA_C (1 << 30) -#define TEGRA_SPDIF_DATA_IN_DATA_U (1 << 29) -#define TEGRA_SPDIF_DATA_IN_DATA_V (1 << 28) - -#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT 24 -#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_16_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_20_MASK (0xfffff << TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_24_MASK (0xffffff << TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT 8 -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_MASK (0xfffff << TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT 4 -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT 16 -#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT) - -#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT 0 -#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT) - -/* Fields in TEGRA_SPDIF_CH_STA_RX_A */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_B */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_C */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_D */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_E */ -/* Fields in TEGRA_SPDIF_CH_STA_RX_F */ - -/* - * The 6-word receive channel data page buffer holds a block (192 frames) of - * channel status information. The order of receive is from LSB to MSB - * bit, and from CH_STA_RX_A to CH_STA_RX_F then back to CH_STA_RX_A. - */ - -/* Fields in TEGRA_SPDIF_CH_STA_TX_A */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_B */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_C */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_D */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_E */ -/* Fields in TEGRA_SPDIF_CH_STA_TX_F */ - -/* - * The 6-word transmit channel data page buffer holds a block (192 frames) of - * channel status information. The order of transmission is from LSB to MSB - * bit, and from CH_STA_TX_A to CH_STA_TX_F then back to CH_STA_TX_A. - */ - -/* Fields in TEGRA_SPDIF_USR_STA_RX_A */ - -/* - * This 4-word deep FIFO receives user FIFO field information. The order of - * receive is from LSB to MSB bit. - */ - -/* Fields in TEGRA_SPDIF_USR_DAT_TX_A */ - -/* - * This 4-word deep FIFO transmits user FIFO field information. The order of - * transmission is from LSB to MSB bit. - */ - -struct tegra_spdif { - struct clk *clk_spdif_out; - int clk_refs; - struct tegra_pcm_dma_params capture_dma_data; - struct tegra_pcm_dma_params playback_dma_data; - void __iomem *regs; - struct dentry *debug; - u32 reg_ctrl; -}; - -#endif diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c new file mode 100644 index 000000000000..ea9166d5c4eb --- /dev/null +++ b/sound/soc/tegra/tegra_wm8753.c @@ -0,0 +1,224 @@ +/* + * tegra_wm8753.c - Tegra machine ASoC driver for boards using WM8753 codec. + * + * Author: Stephen Warren <swarren@nvidia.com> + * Copyright (C) 2010-2012 - NVIDIA, Inc. + * + * Based on code copyright/by: + * + * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd. + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <asm/mach-types.h> + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../codecs/wm8753.h" + +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-snd-wm8753" + +struct tegra_wm8753 { + struct tegra_asoc_utils_data util_data; +}; + +static int tegra_wm8753_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_card *card = codec->card; + struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card); + int srate, mclk; + int err; + + srate = params_rate(params); + switch (srate) { + case 11025: + case 22050: + case 44100: + case 88200: + mclk = 11289600; + break; + default: + mclk = 12288000; + break; + } + + err = tegra_asoc_utils_set_rate(&machine->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static struct snd_soc_ops tegra_wm8753_ops = { + .hw_params = tegra_wm8753_hw_params, +}; + +static const struct snd_soc_dapm_widget tegra_wm8753_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static struct snd_soc_dai_link tegra_wm8753_dai = { + .name = "WM8753", + .stream_name = "WM8753 PCM", + .codec_dai_name = "wm8753-hifi", + .ops = &tegra_wm8753_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_tegra_wm8753 = { + .name = "tegra-wm8753", + .owner = THIS_MODULE, + .dai_link = &tegra_wm8753_dai, + .num_links = 1, + + .dapm_widgets = tegra_wm8753_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_wm8753_dapm_widgets), + .fully_routed = true, +}; + +static __devinit int tegra_wm8753_driver_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_tegra_wm8753; + struct tegra_wm8753 *machine; + int ret; + + machine = devm_kzalloc(&pdev->dev, sizeof(struct tegra_wm8753), + GFP_KERNEL); + if (!machine) { + dev_err(&pdev->dev, "Can't allocate tegra_wm8753 struct\n"); + ret = -ENOMEM; + goto err; + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing"); + if (ret) + goto err; + + tegra_wm8753_dai.codec_of_node = of_parse_phandle( + pdev->dev.of_node, "nvidia,audio-codec", 0); + if (!tegra_wm8753_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_wm8753_dai.cpu_of_node = of_parse_phandle( + pdev->dev.of_node, "nvidia,i2s-controller", 0); + if (!tegra_wm8753_dai.cpu_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_wm8753_dai.platform_of_node = + tegra_wm8753_dai.cpu_of_node; + + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); + if (ret) + goto err; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto err_fini_utils; + } + + return 0; + +err_fini_utils: + tegra_asoc_utils_fini(&machine->util_data); +err: + return ret; +} + +static int __devexit tegra_wm8753_driver_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&machine->util_data); + + return 0; +} + +static const struct of_device_id tegra_wm8753_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra-audio-wm8753", }, + {}, +}; + +static struct platform_driver tegra_wm8753_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = tegra_wm8753_of_match, + }, + .probe = tegra_wm8753_driver_probe, + .remove = __devexit_p(tegra_wm8753_driver_remove), +}; +module_platform_driver(tegra_wm8753_driver); + +MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); +MODULE_DESCRIPTION("Tegra+WM8753 machine ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_wm8753_of_match); diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 566655e23b7d..0c5bb33d258e 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -2,7 +2,7 @@ * tegra_wm8903.c - Tegra machine ASoC driver for boards using WM8903 codec. * * Author: Stephen Warren <swarren@nvidia.com> - * Copyright (C) 2010-2011 - NVIDIA, Inc. + * Copyright (C) 2010-2012 - NVIDIA, Inc. * * Based on code copyright/by: * @@ -28,8 +28,6 @@ * */ -#include <asm/mach-types.h> - #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> @@ -46,24 +44,13 @@ #include "../codecs/wm8903.h" -#include "tegra_das.h" -#include "tegra_i2s.h" -#include "tegra_pcm.h" #include "tegra_asoc_utils.h" #define DRV_NAME "tegra-snd-wm8903" -#define GPIO_SPKR_EN BIT(0) -#define GPIO_HP_MUTE BIT(1) -#define GPIO_INT_MIC_EN BIT(2) -#define GPIO_EXT_MIC_EN BIT(3) -#define GPIO_HP_DET BIT(4) - struct tegra_wm8903 { struct tegra_wm8903_platform_data pdata; - struct platform_device *pcm_dev; struct tegra_asoc_utils_data util_data; - int gpio_requested; }; static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, @@ -71,8 +58,7 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_card *card = codec->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; @@ -99,24 +85,6 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, return err; } - err = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (err < 0) { - dev_err(card->dev, "codec_dai fmt not set\n"); - return err; - } - - err = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (err < 0) { - dev_err(card->dev, "cpu_dai fmt not set\n"); - return err; - } - err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, SND_SOC_CLOCK_IN); if (err < 0) { @@ -164,7 +132,7 @@ static int tegra_wm8903_event_int_spk(struct snd_soc_dapm_widget *w, struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); struct tegra_wm8903_platform_data *pdata = &machine->pdata; - if (!(machine->gpio_requested & GPIO_SPKR_EN)) + if (!gpio_is_valid(pdata->gpio_spkr_en)) return 0; gpio_set_value_cansleep(pdata->gpio_spkr_en, @@ -181,7 +149,7 @@ static int tegra_wm8903_event_hp(struct snd_soc_dapm_widget *w, struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); struct tegra_wm8903_platform_data *pdata = &machine->pdata; - if (!(machine->gpio_requested & GPIO_HP_MUTE)) + if (!gpio_is_valid(pdata->gpio_hp_mute)) return 0; gpio_set_value_cansleep(pdata->gpio_hp_mute, @@ -207,122 +175,18 @@ static const struct snd_soc_dapm_route harmony_audio_map[] = { {"IN1L", NULL, "Mic Jack"}, }; -static const struct snd_soc_dapm_route seaboard_audio_map[] = { - {"Headphone Jack", NULL, "HPOUTR"}, - {"Headphone Jack", NULL, "HPOUTL"}, - {"Int Spk", NULL, "ROP"}, - {"Int Spk", NULL, "RON"}, - {"Int Spk", NULL, "LOP"}, - {"Int Spk", NULL, "LON"}, - {"Mic Jack", NULL, "MICBIAS"}, - {"IN1R", NULL, "Mic Jack"}, -}; - -static const struct snd_soc_dapm_route kaen_audio_map[] = { - {"Headphone Jack", NULL, "HPOUTR"}, - {"Headphone Jack", NULL, "HPOUTL"}, - {"Int Spk", NULL, "ROP"}, - {"Int Spk", NULL, "RON"}, - {"Int Spk", NULL, "LOP"}, - {"Int Spk", NULL, "LON"}, - {"Mic Jack", NULL, "MICBIAS"}, - {"IN2R", NULL, "Mic Jack"}, -}; - -static const struct snd_soc_dapm_route aebl_audio_map[] = { - {"Headphone Jack", NULL, "HPOUTR"}, - {"Headphone Jack", NULL, "HPOUTL"}, - {"Int Spk", NULL, "LINEOUTR"}, - {"Int Spk", NULL, "LINEOUTL"}, - {"Mic Jack", NULL, "MICBIAS"}, - {"IN1R", NULL, "Mic Jack"}, -}; - static const struct snd_kcontrol_new tegra_wm8903_controls[] = { SOC_DAPM_PIN_SWITCH("Int Spk"), }; static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_card *card = codec->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); struct tegra_wm8903_platform_data *pdata = &machine->pdata; - struct device_node *np = card->dev->of_node; - int ret; - - if (card->dev->platform_data) { - memcpy(pdata, card->dev->platform_data, sizeof(*pdata)); - } else if (np) { - /* - * This part must be in init() rather than probe() in order to - * guarantee that the WM8903 has been probed, and hence its - * GPIO controller registered, which is a pre-condition for - * of_get_named_gpio() to be able to map the phandles in the - * properties to the controller node. Given this, all - * pdata handling is in init() for consistency. - */ - pdata->gpio_spkr_en = of_get_named_gpio(np, - "nvidia,spkr-en-gpios", 0); - pdata->gpio_hp_mute = of_get_named_gpio(np, - "nvidia,hp-mute-gpios", 0); - pdata->gpio_hp_det = of_get_named_gpio(np, - "nvidia,hp-det-gpios", 0); - pdata->gpio_int_mic_en = of_get_named_gpio(np, - "nvidia,int-mic-en-gpios", 0); - pdata->gpio_ext_mic_en = of_get_named_gpio(np, - "nvidia,ext-mic-en-gpios", 0); - } else { - dev_err(card->dev, "No platform data supplied\n"); - return -EINVAL; - } - - if (gpio_is_valid(pdata->gpio_spkr_en)) { - ret = gpio_request(pdata->gpio_spkr_en, "spkr_en"); - if (ret) { - dev_err(card->dev, "cannot get spkr_en gpio\n"); - return ret; - } - machine->gpio_requested |= GPIO_SPKR_EN; - - gpio_direction_output(pdata->gpio_spkr_en, 0); - } - - if (gpio_is_valid(pdata->gpio_hp_mute)) { - ret = gpio_request(pdata->gpio_hp_mute, "hp_mute"); - if (ret) { - dev_err(card->dev, "cannot get hp_mute gpio\n"); - return ret; - } - machine->gpio_requested |= GPIO_HP_MUTE; - - gpio_direction_output(pdata->gpio_hp_mute, 1); - } - - if (gpio_is_valid(pdata->gpio_int_mic_en)) { - ret = gpio_request(pdata->gpio_int_mic_en, "int_mic_en"); - if (ret) { - dev_err(card->dev, "cannot get int_mic_en gpio\n"); - return ret; - } - machine->gpio_requested |= GPIO_INT_MIC_EN; - - /* Disable int mic; enable signal is active-high */ - gpio_direction_output(pdata->gpio_int_mic_en, 0); - } - - if (gpio_is_valid(pdata->gpio_ext_mic_en)) { - ret = gpio_request(pdata->gpio_ext_mic_en, "ext_mic_en"); - if (ret) { - dev_err(card->dev, "cannot get ext_mic_en gpio\n"); - return ret; - } - machine->gpio_requested |= GPIO_EXT_MIC_EN; - - /* Enable ext mic; enable signal is active-low */ - gpio_direction_output(pdata->gpio_ext_mic_en, 0); - } if (gpio_is_valid(pdata->gpio_hp_det)) { tegra_wm8903_hp_jack_gpio.gpio = pdata->gpio_hp_det; @@ -334,7 +198,6 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack, 1, &tegra_wm8903_hp_jack_gpio); - machine->gpio_requested |= GPIO_HP_DET; } snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, @@ -350,15 +213,29 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int tegra_wm8903_remove(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd = &(card->rtd[0]); + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + + wm8903_mic_detect(codec, NULL, 0, 0); + + return 0; +} + static struct snd_soc_dai_link tegra_wm8903_dai = { .name = "WM8903", .stream_name = "WM8903 PCM", .codec_name = "wm8903.0-001a", - .platform_name = "tegra-pcm-audio", - .cpu_dai_name = "tegra-i2s.0", + .platform_name = "tegra20-i2s.0", + .cpu_dai_name = "tegra20-i2s.0", .codec_dai_name = "wm8903-hifi", .init = tegra_wm8903_init, .ops = &tegra_wm8903_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, }; static struct snd_soc_card snd_soc_tegra_wm8903 = { @@ -367,6 +244,8 @@ static struct snd_soc_card snd_soc_tegra_wm8903 = { .dai_link = &tegra_wm8903_dai, .num_links = 1, + .remove = tegra_wm8903_remove, + .controls = tegra_wm8903_controls, .num_controls = ARRAY_SIZE(tegra_wm8903_controls), .dapm_widgets = tegra_wm8903_dapm_widgets, @@ -376,8 +255,10 @@ static struct snd_soc_card snd_soc_tegra_wm8903 = { static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) { + struct device_node *np = pdev->dev.of_node; struct snd_soc_card *card = &snd_soc_tegra_wm8903; struct tegra_wm8903 *machine; + struct tegra_wm8903_platform_data *pdata; int ret; if (!pdev->dev.platform_data && !pdev->dev.of_node) { @@ -392,13 +273,42 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) ret = -ENOMEM; goto err; } - machine->pcm_dev = ERR_PTR(-EINVAL); + pdata = &machine->pdata; card->dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); - if (pdev->dev.of_node) { + if (pdev->dev.platform_data) { + memcpy(pdata, card->dev->platform_data, sizeof(*pdata)); + } else if (np) { + pdata->gpio_spkr_en = of_get_named_gpio(np, + "nvidia,spkr-en-gpios", 0); + if (pdata->gpio_spkr_en == -ENODEV) + return -EPROBE_DEFER; + + pdata->gpio_hp_mute = of_get_named_gpio(np, + "nvidia,hp-mute-gpios", 0); + if (pdata->gpio_hp_mute == -ENODEV) + return -EPROBE_DEFER; + + pdata->gpio_hp_det = of_get_named_gpio(np, + "nvidia,hp-det-gpios", 0); + if (pdata->gpio_hp_det == -ENODEV) + return -EPROBE_DEFER; + + pdata->gpio_int_mic_en = of_get_named_gpio(np, + "nvidia,int-mic-en-gpios", 0); + if (pdata->gpio_int_mic_en == -ENODEV) + return -EPROBE_DEFER; + + pdata->gpio_ext_mic_en = of_get_named_gpio(np, + "nvidia,ext-mic-en-gpios", 0); + if (pdata->gpio_ext_mic_en == -ENODEV) + return -EPROBE_DEFER; + } + + if (np) { ret = snd_soc_of_parse_card_name(card, "nvidia,model"); if (ret) goto err; @@ -409,8 +319,8 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) goto err; tegra_wm8903_dai.codec_name = NULL; - tegra_wm8903_dai.codec_of_node = of_parse_phandle( - pdev->dev.of_node, "nvidia,audio-codec", 0); + tegra_wm8903_dai.codec_of_node = of_parse_phandle(np, + "nvidia,audio-codec", 0); if (!tegra_wm8903_dai.codec_of_node) { dev_err(&pdev->dev, "Property 'nvidia,audio-codec' missing or invalid\n"); @@ -419,42 +329,64 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) } tegra_wm8903_dai.cpu_dai_name = NULL; - tegra_wm8903_dai.cpu_dai_of_node = of_parse_phandle( - pdev->dev.of_node, "nvidia,i2s-controller", 0); - if (!tegra_wm8903_dai.cpu_dai_of_node) { + tegra_wm8903_dai.cpu_of_node = of_parse_phandle(np, + "nvidia,i2s-controller", 0); + if (!tegra_wm8903_dai.cpu_of_node) { dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; goto err; } - machine->pcm_dev = platform_device_register_simple( - "tegra-pcm-audio", -1, NULL, 0); - if (IS_ERR(machine->pcm_dev)) { - dev_err(&pdev->dev, - "Can't instantiate tegra-pcm-audio\n"); - ret = PTR_ERR(machine->pcm_dev); - goto err; - } + tegra_wm8903_dai.platform_name = NULL; + tegra_wm8903_dai.platform_of_node = + tegra_wm8903_dai.cpu_of_node; } else { - if (machine_is_harmony()) { - card->dapm_routes = harmony_audio_map; - card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map); - } else if (machine_is_seaboard()) { - card->dapm_routes = seaboard_audio_map; - card->num_dapm_routes = ARRAY_SIZE(seaboard_audio_map); - } else if (machine_is_kaen()) { - card->dapm_routes = kaen_audio_map; - card->num_dapm_routes = ARRAY_SIZE(kaen_audio_map); - } else { - card->dapm_routes = aebl_audio_map; - card->num_dapm_routes = ARRAY_SIZE(aebl_audio_map); + card->dapm_routes = harmony_audio_map; + card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map); + } + + if (gpio_is_valid(pdata->gpio_spkr_en)) { + ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_spkr_en, + GPIOF_OUT_INIT_LOW, "spkr_en"); + if (ret) { + dev_err(card->dev, "cannot get spkr_en gpio\n"); + return ret; + } + } + + if (gpio_is_valid(pdata->gpio_hp_mute)) { + ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_hp_mute, + GPIOF_OUT_INIT_HIGH, "hp_mute"); + if (ret) { + dev_err(card->dev, "cannot get hp_mute gpio\n"); + return ret; + } + } + + if (gpio_is_valid(pdata->gpio_int_mic_en)) { + /* Disable int mic; enable signal is active-high */ + ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_int_mic_en, + GPIOF_OUT_INIT_LOW, "int_mic_en"); + if (ret) { + dev_err(card->dev, "cannot get int_mic_en gpio\n"); + return ret; + } + } + + if (gpio_is_valid(pdata->gpio_ext_mic_en)) { + /* Enable ext mic; enable signal is active-low */ + ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_ext_mic_en, + GPIOF_OUT_INIT_LOW, "ext_mic_en"); + if (ret) { + dev_err(card->dev, "cannot get ext_mic_en gpio\n"); + return ret; } } ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) - goto err_unregister; + goto err; ret = snd_soc_register_card(card); if (ret) { @@ -467,9 +399,6 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&machine->util_data); -err_unregister: - if (!IS_ERR(machine->pcm_dev)) - platform_device_unregister(machine->pcm_dev); err: return ret; } @@ -478,27 +407,13 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); - struct tegra_wm8903_platform_data *pdata = &machine->pdata; - if (machine->gpio_requested & GPIO_HP_DET) - snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack, - 1, - &tegra_wm8903_hp_jack_gpio); - if (machine->gpio_requested & GPIO_EXT_MIC_EN) - gpio_free(pdata->gpio_ext_mic_en); - if (machine->gpio_requested & GPIO_INT_MIC_EN) - gpio_free(pdata->gpio_int_mic_en); - if (machine->gpio_requested & GPIO_HP_MUTE) - gpio_free(pdata->gpio_hp_mute); - if (machine->gpio_requested & GPIO_SPKR_EN) - gpio_free(pdata->gpio_spkr_en); - machine->gpio_requested = 0; + snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack, 1, + &tegra_wm8903_hp_jack_gpio); snd_soc_unregister_card(card); tegra_asoc_utils_fini(&machine->util_data); - if (!IS_ERR(machine->pcm_dev)) - platform_device_unregister(machine->pcm_dev); return 0; } diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 2bdfc550cff8..e69a4f7000d6 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -27,6 +27,7 @@ #include <asm/mach-types.h> #include <linux/module.h> +#include <linux/of.h> #include <linux/platform_device.h> #include <linux/slab.h> @@ -38,9 +39,6 @@ #include "../codecs/tlv320aic23.h" -#include "tegra_das.h" -#include "tegra_i2s.h" -#include "tegra_pcm.h" #include "tegra_asoc_utils.h" #define DRV_NAME "tegra-snd-trimslice" @@ -54,8 +52,7 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_card *card = codec->card; struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card); int srate, mclk; @@ -70,24 +67,6 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream, return err; } - err = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (err < 0) { - dev_err(card->dev, "codec_dai fmt not set\n"); - return err; - } - - err = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (err < 0) { - dev_err(card->dev, "cpu_dai fmt not set\n"); - return err; - } - err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, SND_SOC_CLOCK_IN); if (err < 0) { @@ -119,10 +98,13 @@ static struct snd_soc_dai_link trimslice_tlv320aic23_dai = { .name = "TLV320AIC23", .stream_name = "AIC23", .codec_name = "tlv320aic23-codec.2-001a", - .platform_name = "tegra-pcm-audio", - .cpu_dai_name = "tegra-i2s.0", + .platform_name = "tegra20-i2s.0", + .cpu_dai_name = "tegra20-i2s.0", .codec_dai_name = "tlv320aic23-hifi", .ops = &trimslice_asoc_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, }; static struct snd_soc_card snd_soc_trimslice = { @@ -152,6 +134,32 @@ static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev) goto err; } + if (pdev->dev.of_node) { + trimslice_tlv320aic23_dai.codec_name = NULL; + trimslice_tlv320aic23_dai.codec_of_node = of_parse_phandle( + pdev->dev.of_node, "nvidia,audio-codec", 0); + if (!trimslice_tlv320aic23_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + trimslice_tlv320aic23_dai.cpu_dai_name = NULL; + trimslice_tlv320aic23_dai.cpu_of_node = of_parse_phandle( + pdev->dev.of_node, "nvidia,i2s-controller", 0); + if (!trimslice_tlv320aic23_dai.cpu_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + trimslice_tlv320aic23_dai.platform_name = NULL; + trimslice_tlv320aic23_dai.platform_of_node = + trimslice_tlv320aic23_dai.cpu_of_node; + } + ret = tegra_asoc_utils_init(&trimslice->util_data, &pdev->dev); if (ret) goto err; @@ -187,10 +195,17 @@ static int __devexit tegra_snd_trimslice_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id trimslice_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra-audio-trimslice", }, + {}, +}; +MODULE_DEVICE_TABLE(of, trimslice_of_match); + static struct platform_driver tegra_snd_trimslice_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = trimslice_of_match, }, .probe = tegra_snd_trimslice_probe, .remove = __devexit_p(tegra_snd_trimslice_remove), diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig new file mode 100644 index 000000000000..069330d82be5 --- /dev/null +++ b/sound/soc/ux500/Kconfig @@ -0,0 +1,32 @@ +# +# Ux500 SoC audio configuration +# +menuconfig SND_SOC_UX500 + tristate "SoC Audio support for Ux500 platform" + depends on SND_SOC + depends on MFD_DB8500_PRCMU + help + Say Y if you want to enable ASoC-support for + any of the Ux500 platforms (e.g. U8500). + +config SND_SOC_UX500_PLAT_MSP_I2S + tristate + depends on SND_SOC_UX500 + +config SND_SOC_UX500_PLAT_DMA + tristate "Platform - DB8500 (DMA)" + depends on SND_SOC_UX500 + select SND_SOC_DMAENGINE_PCM + help + Say Y if you want to enable the Ux500 platform-driver. + ++config SND_SOC_UX500_MACH_MOP500 ++ tristate "Machine - MOP500 (Ux500 + AB8500)" + depends on AB8500_CORE && AB8500_GPADC && SND_SOC_UX500 + select SND_SOC_AB8500_CODEC + select SND_SOC_UX500_PLAT_MSP_I2S + select SND_SOC_UX500_PLAT_DMA + help + Select this to enable the MOP500 machine-driver. + This will enable platform-drivers for: Ux500 + This will enable codec-drivers for: AB8500 diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile new file mode 100644 index 000000000000..cce0c11a4d86 --- /dev/null +++ b/sound/soc/ux500/Makefile @@ -0,0 +1,10 @@ +# Ux500 Platform Support + +snd-soc-ux500-plat-msp-i2s-objs := ux500_msp_dai.o ux500_msp_i2s.o +obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o + +snd-soc-ux500-plat-dma-objs := ux500_pcm.o +obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o + +snd-soc-ux500-mach-mop500-objs := mop500.o mop500_ab8500.o +obj-$(CONFIG_SND_SOC_UX500_MACH_MOP500) += snd-soc-ux500-mach-mop500.o diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c new file mode 100644 index 000000000000..31c4d26d0359 --- /dev/null +++ b/sound/soc/ux500/mop500.c @@ -0,0 +1,113 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja (ola.o.lilja@stericsson.com) + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <asm/mach-types.h> + +#include <linux/module.h> +#include <linux/io.h> +#include <linux/spi/spi.h> + +#include <sound/soc.h> +#include <sound/initval.h> + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" + +#include <mop500_ab8500.h> + +/* Define the whole MOP500 soundcard, linking platform to the codec-drivers */ +struct snd_soc_dai_link mop500_dai_links[] = { + { + .name = "ab8500_0", + .stream_name = "ab8500_0", + .cpu_dai_name = "ux500-msp-i2s.1", + .codec_dai_name = "ab8500-codec-dai.0", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = mop500_ab8500_machine_init, + .ops = mop500_ab8500_ops, + }, + { + .name = "ab8500_1", + .stream_name = "ab8500_1", + .cpu_dai_name = "ux500-msp-i2s.3", + .codec_dai_name = "ab8500-codec-dai.1", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = NULL, + .ops = mop500_ab8500_ops, + }, +}; + +static struct snd_soc_card mop500_card = { + .name = "MOP500-card", + .probe = NULL, + .dai_link = mop500_dai_links, + .num_links = ARRAY_SIZE(mop500_dai_links), +}; + +static int __devinit mop500_probe(struct platform_device *pdev) +{ + int ret; + + pr_debug("%s: Enter.\n", __func__); + + dev_dbg(&pdev->dev, "%s: Enter.\n", __func__); + + mop500_card.dev = &pdev->dev; + + dev_dbg(&pdev->dev, "%s: Card %s: Set platform drvdata.\n", + __func__, mop500_card.name); + platform_set_drvdata(pdev, &mop500_card); + + snd_soc_card_set_drvdata(&mop500_card, NULL); + + dev_dbg(&pdev->dev, "%s: Card %s: num_links = %d\n", + __func__, mop500_card.name, mop500_card.num_links); + dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: name = %s\n", + __func__, mop500_card.name, mop500_card.dai_link[0].name); + dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: stream_name = %s\n", + __func__, mop500_card.name, + mop500_card.dai_link[0].stream_name); + + ret = snd_soc_register_card(&mop500_card); + if (ret) + dev_err(&pdev->dev, + "Error: snd_soc_register_card failed (%d)!\n", + ret); + + return ret; +} + +static int __devexit mop500_remove(struct platform_device *pdev) +{ + struct snd_soc_card *mop500_card = platform_get_drvdata(pdev); + + pr_debug("%s: Enter.\n", __func__); + + snd_soc_unregister_card(mop500_card); + mop500_ab8500_remove(mop500_card); + + return 0; +} + +static struct platform_driver snd_soc_mop500_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "snd-soc-mop500", + }, + .probe = mop500_probe, + .remove = __devexit_p(mop500_remove), +}; + +module_platform_driver(snd_soc_mop500_driver); diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c new file mode 100644 index 000000000000..78cce236693e --- /dev/null +++ b/sound/soc/ux500/mop500_ab8500.c @@ -0,0 +1,431 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com>, + * Kristoffer Karlsson <kristoffer.karlsson@stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/device.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <mach/hardware.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" +#include "../codecs/ab8500-codec.h" + +#define TX_SLOT_MONO 0x0008 +#define TX_SLOT_STEREO 0x000a +#define RX_SLOT_MONO 0x0001 +#define RX_SLOT_STEREO 0x0003 +#define TX_SLOT_8CH 0x00FF +#define RX_SLOT_8CH 0x00FF + +#define DEF_TX_SLOTS TX_SLOT_STEREO +#define DEF_RX_SLOTS RX_SLOT_MONO + +#define DRIVERMODE_NORMAL 0 +#define DRIVERMODE_CODEC_ONLY 1 + +/* Slot configuration */ +static unsigned int tx_slots = DEF_TX_SLOTS; +static unsigned int rx_slots = DEF_RX_SLOTS; + +/* Clocks */ +static const char * const enum_mclk[] = { + "SYSCLK", + "ULPCLK" +}; +enum mclk { + MCLK_SYSCLK, + MCLK_ULPCLK, +}; + +static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_mclk, enum_mclk); + +/* Private data for machine-part MOP500<->AB8500 */ +struct mop500_ab8500_drvdata { + /* Clocks */ + enum mclk mclk_sel; + struct clk *clk_ptr_intclk; + struct clk *clk_ptr_sysclk; + struct clk *clk_ptr_ulpclk; +}; + +static inline const char *get_mclk_str(enum mclk mclk_sel) +{ + switch (mclk_sel) { + case MCLK_SYSCLK: + return "SYSCLK"; + case MCLK_ULPCLK: + return "ULPCLK"; + default: + return "Unknown"; + } +} + +static int mop500_ab8500_set_mclk(struct device *dev, + struct mop500_ab8500_drvdata *drvdata) +{ + int status; + struct clk *clk_ptr; + + if (IS_ERR(drvdata->clk_ptr_intclk)) { + dev_err(dev, + "%s: ERROR: intclk not initialized!\n", __func__); + return -EIO; + } + + switch (drvdata->mclk_sel) { + case MCLK_SYSCLK: + clk_ptr = drvdata->clk_ptr_sysclk; + break; + case MCLK_ULPCLK: + clk_ptr = drvdata->clk_ptr_ulpclk; + break; + default: + return -EINVAL; + } + + if (IS_ERR(clk_ptr)) { + dev_err(dev, "%s: ERROR: %s not initialized!\n", __func__, + get_mclk_str(drvdata->mclk_sel)); + return -EIO; + } + + status = clk_set_parent(drvdata->clk_ptr_intclk, clk_ptr); + if (status) + dev_err(dev, + "%s: ERROR: Setting intclk parent to %s failed (ret = %d)!", + __func__, get_mclk_str(drvdata->mclk_sel), status); + else + dev_dbg(dev, + "%s: intclk parent changed to %s.\n", + __func__, get_mclk_str(drvdata->mclk_sel)); + + return status; +} + +/* + * Control-events + */ + +static int mclk_input_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mop500_ab8500_drvdata *drvdata = + snd_soc_card_get_drvdata(codec->card); + + ucontrol->value.enumerated.item[0] = drvdata->mclk_sel; + + return 0; +} + +static int mclk_input_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mop500_ab8500_drvdata *drvdata = + snd_soc_card_get_drvdata(codec->card); + unsigned int val = ucontrol->value.enumerated.item[0]; + + if (val > (unsigned int)MCLK_ULPCLK) + return -EINVAL; + if (drvdata->mclk_sel == val) + return 0; + + drvdata->mclk_sel = val; + + return 1; +} + +/* + * Controls + */ + +static struct snd_kcontrol_new mop500_ab8500_ctrls[] = { + SOC_ENUM_EXT("Master Clock Select", + soc_enum_mclk, + mclk_input_control_get, mclk_input_control_put), + /* Digital interface - Clocks */ + SOC_SINGLE("Digital Interface Master Generator Switch", + AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENMASTGEN, + 1, 0), + SOC_SINGLE("Digital Interface 0 Bit-clock Switch", + AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK0, + 1, 0), + SOC_SINGLE("Digital Interface 1 Bit-clock Switch", + AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK1, + 1, 0), + SOC_DAPM_PIN_SWITCH("Headset Left"), + SOC_DAPM_PIN_SWITCH("Headset Right"), + SOC_DAPM_PIN_SWITCH("Earpiece"), + SOC_DAPM_PIN_SWITCH("Speaker Left"), + SOC_DAPM_PIN_SWITCH("Speaker Right"), + SOC_DAPM_PIN_SWITCH("LineOut Left"), + SOC_DAPM_PIN_SWITCH("LineOut Right"), + SOC_DAPM_PIN_SWITCH("Vibra 1"), + SOC_DAPM_PIN_SWITCH("Vibra 2"), + SOC_DAPM_PIN_SWITCH("Mic 1"), + SOC_DAPM_PIN_SWITCH("Mic 2"), + SOC_DAPM_PIN_SWITCH("LineIn Left"), + SOC_DAPM_PIN_SWITCH("LineIn Right"), + SOC_DAPM_PIN_SWITCH("DMic 1"), + SOC_DAPM_PIN_SWITCH("DMic 2"), + SOC_DAPM_PIN_SWITCH("DMic 3"), + SOC_DAPM_PIN_SWITCH("DMic 4"), + SOC_DAPM_PIN_SWITCH("DMic 5"), + SOC_DAPM_PIN_SWITCH("DMic 6"), +}; + +/* ASoC */ + +int mop500_ab8500_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* Set audio-clock source */ + return mop500_ab8500_set_mclk(rtd->card->dev, + snd_soc_card_get_drvdata(rtd->card)); +} + +void mop500_ab8500_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->card->dev; + + dev_dbg(dev, "%s: Enter\n", __func__); + + /* Reset slots configuration to default(s) */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tx_slots = DEF_TX_SLOTS; + else + rx_slots = DEF_RX_SLOTS; +} + +int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct device *dev = rtd->card->dev; + unsigned int fmt; + int channels, ret = 0, driver_mode, slots; + unsigned int sw_codec, sw_cpu; + bool is_playback; + + dev_dbg(dev, "%s: Enter\n", __func__); + + dev_dbg(dev, "%s: substream->pcm->name = %s\n" + "substream->pcm->id = %s.\n" + "substream->name = %s.\n" + "substream->number = %d.\n", + __func__, + substream->pcm->name, + substream->pcm->id, + substream->name, + substream->number); + + channels = params_channels(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S32_LE: + sw_cpu = 32; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + sw_cpu = 16; + break; + + default: + return -EINVAL; + } + + /* Setup codec depending on driver-mode */ + if (channels == 8) + driver_mode = DRIVERMODE_CODEC_ONLY; + else + driver_mode = DRIVERMODE_NORMAL; + dev_dbg(dev, "%s: Driver-mode: %s.\n", __func__, + (driver_mode == DRIVERMODE_NORMAL) ? "NORMAL" : "CODEC_ONLY"); + + /* Setup format */ + + if (driver_mode == DRIVERMODE_NORMAL) { + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CONT; + } else { + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_GATED; + } + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + dev_err(dev, + "%s: ERROR: snd_soc_dai_set_fmt failed for codec_dai (ret = %d)!\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + dev_err(dev, + "%s: ERROR: snd_soc_dai_set_fmt failed for cpu_dai (ret = %d)!\n", + __func__, ret); + return ret; + } + + /* Setup TDM-slots */ + + is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + switch (channels) { + case 1: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_MONO : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_MONO; + break; + case 2: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_STEREO : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_STEREO; + break; + case 8: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_8CH : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_8CH; + break; + default: + return -EINVAL; + } + + if (driver_mode == DRIVERMODE_NORMAL) + sw_codec = sw_cpu; + else + sw_codec = 20; + + dev_dbg(dev, "%s: CPU-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__, + tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, tx_slots, rx_slots, slots, + sw_cpu); + if (ret) + return ret; + + dev_dbg(dev, "%s: CODEC-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__, + tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(codec_dai, tx_slots, rx_slots, slots, + sw_codec); + if (ret) + return ret; + + return 0; +} + +struct snd_soc_ops mop500_ab8500_ops[] = { + { + .hw_params = mop500_ab8500_hw_params, + .startup = mop500_ab8500_startup, + .shutdown = mop500_ab8500_shutdown, + } +}; + +int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct device *dev = rtd->card->dev; + struct mop500_ab8500_drvdata *drvdata; + int ret; + + dev_dbg(dev, "%s Enter.\n", __func__); + + /* Create driver private-data struct */ + drvdata = devm_kzalloc(dev, sizeof(struct mop500_ab8500_drvdata), + GFP_KERNEL); + snd_soc_card_set_drvdata(rtd->card, drvdata); + + /* Setup clocks */ + + drvdata->clk_ptr_sysclk = clk_get(dev, "sysclk"); + if (IS_ERR(drvdata->clk_ptr_sysclk)) + dev_warn(dev, "%s: WARNING: clk_get failed for 'sysclk'!\n", + __func__); + drvdata->clk_ptr_ulpclk = clk_get(dev, "ulpclk"); + if (IS_ERR(drvdata->clk_ptr_ulpclk)) + dev_warn(dev, "%s: WARNING: clk_get failed for 'ulpclk'!\n", + __func__); + drvdata->clk_ptr_intclk = clk_get(dev, "intclk"); + if (IS_ERR(drvdata->clk_ptr_intclk)) + dev_warn(dev, "%s: WARNING: clk_get failed for 'intclk'!\n", + __func__); + + /* Set intclk default parent to ulpclk */ + drvdata->mclk_sel = MCLK_ULPCLK; + ret = mop500_ab8500_set_mclk(dev, drvdata); + if (ret < 0) + dev_warn(dev, "%s: WARNING: mop500_ab8500_set_mclk!\n", + __func__); + + drvdata->mclk_sel = MCLK_ULPCLK; + + /* Add controls */ + ret = snd_soc_add_codec_controls(codec, mop500_ab8500_ctrls, + ARRAY_SIZE(mop500_ab8500_ctrls)); + if (ret < 0) { + pr_err("%s: Failed to add machine-controls (%d)!\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dapm_disable_pin(&codec->dapm, "Earpiece"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 3"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 4"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 5"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 6"); + + return ret; +} + +void mop500_ab8500_remove(struct snd_soc_card *card) +{ + struct mop500_ab8500_drvdata *drvdata = snd_soc_card_get_drvdata(card); + + if (drvdata->clk_ptr_sysclk != NULL) + clk_put(drvdata->clk_ptr_sysclk); + if (drvdata->clk_ptr_ulpclk != NULL) + clk_put(drvdata->clk_ptr_ulpclk); + if (drvdata->clk_ptr_intclk != NULL) + clk_put(drvdata->clk_ptr_intclk); + + snd_soc_card_set_drvdata(card, drvdata); +} diff --git a/sound/soc/ux500/mop500_ab8500.h b/sound/soc/ux500/mop500_ab8500.h new file mode 100644 index 000000000000..cca5b33964b6 --- /dev/null +++ b/sound/soc/ux500/mop500_ab8500.h @@ -0,0 +1,22 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#ifndef MOP500_AB8500_H +#define MOP500_AB8500_H + +extern struct snd_soc_ops mop500_ab8500_ops[]; + +int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *runtime); +void mop500_ab8500_remove(struct snd_soc_card *card); + +#endif diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c new file mode 100644 index 000000000000..62ac0285bfaf --- /dev/null +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -0,0 +1,843 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com>, + * Roger Nilsson <roger.xr.nilsson@stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/bitops.h> +#include <linux/platform_device.h> +#include <linux/clk.h> +#include <linux/regulator/consumer.h> +#include <linux/mfd/dbx500-prcmu.h> + +#include <mach/hardware.h> +#include <mach/board-mop500-msp.h> + +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "ux500_msp_i2s.h" +#include "ux500_msp_dai.h" + +static int setup_pcm_multichan(struct snd_soc_dai *dai, + struct ux500_msp_config *msp_config) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + struct msp_multichannel_config *multi = + &msp_config->multichannel_config; + + if (drvdata->slots > 1) { + msp_config->multichannel_configured = 1; + + multi->tx_multichannel_enable = true; + multi->rx_multichannel_enable = true; + multi->rx_comparison_enable_mode = MSP_COMPARISON_DISABLED; + + multi->tx_channel_0_enable = drvdata->tx_mask; + multi->tx_channel_1_enable = 0; + multi->tx_channel_2_enable = 0; + multi->tx_channel_3_enable = 0; + + multi->rx_channel_0_enable = drvdata->rx_mask; + multi->rx_channel_1_enable = 0; + multi->rx_channel_2_enable = 0; + multi->rx_channel_3_enable = 0; + + dev_dbg(dai->dev, + "%s: Multichannel enabled. Slots: %d, TX: %u, RX: %u\n", + __func__, drvdata->slots, multi->tx_channel_0_enable, + multi->rx_channel_0_enable); + } + + return 0; +} + +static int setup_frameper(struct snd_soc_dai *dai, unsigned int rate, + struct msp_protdesc *prot_desc) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + switch (drvdata->slots) { + case 1: + switch (rate) { + case 8000: + prot_desc->frame_period = + FRAME_PER_SINGLE_SLOT_8_KHZ; + break; + + case 16000: + prot_desc->frame_period = + FRAME_PER_SINGLE_SLOT_16_KHZ; + break; + + case 44100: + prot_desc->frame_period = + FRAME_PER_SINGLE_SLOT_44_1_KHZ; + break; + + case 48000: + prot_desc->frame_period = + FRAME_PER_SINGLE_SLOT_48_KHZ; + break; + + default: + dev_err(dai->dev, + "%s: Error: Unsupported sample-rate (freq = %d)!\n", + __func__, rate); + return -EINVAL; + } + break; + + case 2: + prot_desc->frame_period = FRAME_PER_2_SLOTS; + break; + + case 8: + prot_desc->frame_period = FRAME_PER_8_SLOTS; + break; + + case 16: + prot_desc->frame_period = FRAME_PER_16_SLOTS; + break; + default: + dev_err(dai->dev, + "%s: Error: Unsupported slot-count (slots = %d)!\n", + __func__, drvdata->slots); + return -EINVAL; + } + + prot_desc->clocks_per_frame = + prot_desc->frame_period+1; + + dev_dbg(dai->dev, "%s: Clocks per frame: %u\n", + __func__, + prot_desc->clocks_per_frame); + + return 0; +} + +static int setup_pcm_framing(struct snd_soc_dai *dai, unsigned int rate, + struct msp_protdesc *prot_desc) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + u32 frame_length = MSP_FRAME_LEN_1; + prot_desc->frame_width = 0; + + switch (drvdata->slots) { + case 1: + frame_length = MSP_FRAME_LEN_1; + break; + + case 2: + frame_length = MSP_FRAME_LEN_2; + break; + + case 8: + frame_length = MSP_FRAME_LEN_8; + break; + + case 16: + frame_length = MSP_FRAME_LEN_16; + break; + default: + dev_err(dai->dev, + "%s: Error: Unsupported slot-count (slots = %d)!\n", + __func__, drvdata->slots); + return -EINVAL; + } + + prot_desc->tx_frame_len_1 = frame_length; + prot_desc->rx_frame_len_1 = frame_length; + prot_desc->tx_frame_len_2 = frame_length; + prot_desc->rx_frame_len_2 = frame_length; + + prot_desc->tx_elem_len_1 = MSP_ELEM_LEN_16; + prot_desc->rx_elem_len_1 = MSP_ELEM_LEN_16; + prot_desc->tx_elem_len_2 = MSP_ELEM_LEN_16; + prot_desc->rx_elem_len_2 = MSP_ELEM_LEN_16; + + return setup_frameper(dai, rate, prot_desc); +} + +static int setup_clocking(struct snd_soc_dai *dai, + unsigned int fmt, + struct ux500_msp_config *msp_config) +{ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + + case SND_SOC_DAIFMT_NB_IF: + msp_config->tx_fsync_pol ^= 1 << TFSPOL_SHIFT; + msp_config->rx_fsync_pol ^= 1 << RFSPOL_SHIFT; + + break; + + default: + dev_err(dai->dev, + "%s: Error: Unsopported inversion (fmt = 0x%x)!\n", + __func__, fmt); + + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + dev_dbg(dai->dev, "%s: Codec is master.\n", __func__); + + msp_config->iodelay = 0x20; + msp_config->rx_fsync_sel = 0; + msp_config->tx_fsync_sel = 1 << TFSSEL_SHIFT; + msp_config->tx_clk_sel = 0; + msp_config->rx_clk_sel = 0; + msp_config->srg_clk_sel = 0x2 << SCKSEL_SHIFT; + + break; + + case SND_SOC_DAIFMT_CBS_CFS: + dev_dbg(dai->dev, "%s: Codec is slave.\n", __func__); + + msp_config->tx_clk_sel = TX_CLK_SEL_SRG; + msp_config->tx_fsync_sel = TX_SYNC_SRG_PROG; + msp_config->rx_clk_sel = RX_CLK_SEL_SRG; + msp_config->rx_fsync_sel = RX_SYNC_SRG; + msp_config->srg_clk_sel = 1 << SCKSEL_SHIFT; + + break; + + default: + dev_err(dai->dev, "%s: Error: Unsopported master (fmt = 0x%x)!\n", + __func__, fmt); + + return -EINVAL; + } + + return 0; +} + +static int setup_pcm_protdesc(struct snd_soc_dai *dai, + unsigned int fmt, + struct msp_protdesc *prot_desc) +{ + prot_desc->rx_phase_mode = MSP_SINGLE_PHASE; + prot_desc->tx_phase_mode = MSP_SINGLE_PHASE; + prot_desc->rx_phase2_start_mode = MSP_PHASE2_START_MODE_IMEDIATE; + prot_desc->tx_phase2_start_mode = MSP_PHASE2_START_MODE_IMEDIATE; + prot_desc->rx_byte_order = MSP_BTF_MS_BIT_FIRST; + prot_desc->tx_byte_order = MSP_BTF_MS_BIT_FIRST; + prot_desc->tx_fsync_pol = MSP_FSYNC_POL(MSP_FSYNC_POL_ACT_HI); + prot_desc->rx_fsync_pol = MSP_FSYNC_POL_ACT_HI << RFSPOL_SHIFT; + + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_DSP_A) { + dev_dbg(dai->dev, "%s: DSP_A.\n", __func__); + prot_desc->rx_clk_pol = MSP_RISING_EDGE; + prot_desc->tx_clk_pol = MSP_FALLING_EDGE; + + prot_desc->rx_data_delay = MSP_DELAY_1; + prot_desc->tx_data_delay = MSP_DELAY_1; + } else { + dev_dbg(dai->dev, "%s: DSP_B.\n", __func__); + prot_desc->rx_clk_pol = MSP_FALLING_EDGE; + prot_desc->tx_clk_pol = MSP_RISING_EDGE; + + prot_desc->rx_data_delay = MSP_DELAY_0; + prot_desc->tx_data_delay = MSP_DELAY_0; + } + + prot_desc->rx_half_word_swap = MSP_SWAP_NONE; + prot_desc->tx_half_word_swap = MSP_SWAP_NONE; + prot_desc->compression_mode = MSP_COMPRESS_MODE_LINEAR; + prot_desc->expansion_mode = MSP_EXPAND_MODE_LINEAR; + prot_desc->frame_sync_ignore = MSP_FSYNC_IGNORE; + + return 0; +} + +static int setup_i2s_protdesc(struct msp_protdesc *prot_desc) +{ + prot_desc->rx_phase_mode = MSP_DUAL_PHASE; + prot_desc->tx_phase_mode = MSP_DUAL_PHASE; + prot_desc->rx_phase2_start_mode = MSP_PHASE2_START_MODE_FSYNC; + prot_desc->tx_phase2_start_mode = MSP_PHASE2_START_MODE_FSYNC; + prot_desc->rx_byte_order = MSP_BTF_MS_BIT_FIRST; + prot_desc->tx_byte_order = MSP_BTF_MS_BIT_FIRST; + prot_desc->tx_fsync_pol = MSP_FSYNC_POL(MSP_FSYNC_POL_ACT_LO); + prot_desc->rx_fsync_pol = MSP_FSYNC_POL_ACT_LO << RFSPOL_SHIFT; + + prot_desc->rx_frame_len_1 = MSP_FRAME_LEN_1; + prot_desc->rx_frame_len_2 = MSP_FRAME_LEN_1; + prot_desc->tx_frame_len_1 = MSP_FRAME_LEN_1; + prot_desc->tx_frame_len_2 = MSP_FRAME_LEN_1; + prot_desc->rx_elem_len_1 = MSP_ELEM_LEN_16; + prot_desc->rx_elem_len_2 = MSP_ELEM_LEN_16; + prot_desc->tx_elem_len_1 = MSP_ELEM_LEN_16; + prot_desc->tx_elem_len_2 = MSP_ELEM_LEN_16; + + prot_desc->rx_clk_pol = MSP_RISING_EDGE; + prot_desc->tx_clk_pol = MSP_FALLING_EDGE; + + prot_desc->rx_data_delay = MSP_DELAY_0; + prot_desc->tx_data_delay = MSP_DELAY_0; + + prot_desc->tx_half_word_swap = MSP_SWAP_NONE; + prot_desc->rx_half_word_swap = MSP_SWAP_NONE; + prot_desc->compression_mode = MSP_COMPRESS_MODE_LINEAR; + prot_desc->expansion_mode = MSP_EXPAND_MODE_LINEAR; + prot_desc->frame_sync_ignore = MSP_FSYNC_IGNORE; + + return 0; +} + +static int setup_msp_config(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai, + struct ux500_msp_config *msp_config) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + struct msp_protdesc *prot_desc = &msp_config->protdesc; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int fmt = drvdata->fmt; + int ret; + + memset(msp_config, 0, sizeof(*msp_config)); + + msp_config->f_inputclk = drvdata->master_clk; + + msp_config->tx_fifo_config = TX_FIFO_ENABLE; + msp_config->rx_fifo_config = RX_FIFO_ENABLE; + msp_config->def_elem_len = 1; + msp_config->direction = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + MSP_DIR_TX : MSP_DIR_RX; + msp_config->data_size = MSP_DATA_BITS_32; + msp_config->frame_freq = runtime->rate; + + dev_dbg(dai->dev, "%s: f_inputclk = %u, frame_freq = %u.\n", + __func__, msp_config->f_inputclk, msp_config->frame_freq); + /* To avoid division by zero */ + prot_desc->clocks_per_frame = 1; + + dev_dbg(dai->dev, "%s: rate: %u, channels: %d.\n", __func__, + runtime->rate, runtime->channels); + switch (fmt & + (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + dev_dbg(dai->dev, "%s: SND_SOC_DAIFMT_I2S.\n", __func__); + + msp_config->default_protdesc = 1; + msp_config->protocol = MSP_I2S_PROTOCOL; + break; + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + dev_dbg(dai->dev, "%s: SND_SOC_DAIFMT_I2S.\n", __func__); + + msp_config->data_size = MSP_DATA_BITS_16; + msp_config->protocol = MSP_I2S_PROTOCOL; + + ret = setup_i2s_protdesc(prot_desc); + if (ret < 0) + return ret; + + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM: + dev_dbg(dai->dev, "%s: PCM format.\n", __func__); + + msp_config->data_size = MSP_DATA_BITS_16; + msp_config->protocol = MSP_PCM_PROTOCOL; + + ret = setup_pcm_protdesc(dai, fmt, prot_desc); + if (ret < 0) + return ret; + + ret = setup_pcm_multichan(dai, msp_config); + if (ret < 0) + return ret; + + ret = setup_pcm_framing(dai, runtime->rate, prot_desc); + if (ret < 0) + return ret; + + break; + + default: + dev_err(dai->dev, "%s: Error: Unsopported format (%d)!\n", + __func__, fmt); + return -EINVAL; + } + + return setup_clocking(dai, fmt, msp_config); +} + +static int ux500_msp_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + dev_dbg(dai->dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id, + snd_pcm_stream_str(substream)); + + /* Enable regulator */ + ret = regulator_enable(drvdata->reg_vape); + if (ret != 0) { + dev_err(drvdata->msp->dev, + "%s: Failed to enable regulator!\n", __func__); + return ret; + } + + /* Enable clock */ + dev_dbg(dai->dev, "%s: Enabling MSP-clock.\n", __func__); + clk_enable(drvdata->clk); + + return 0; +} + +static void ux500_msp_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret; + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + + dev_dbg(dai->dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id, + snd_pcm_stream_str(substream)); + + if (drvdata->vape_opp_constraint == 1) { + prcmu_qos_update_requirement(PRCMU_QOS_APE_OPP, + "ux500_msp_i2s", 50); + drvdata->vape_opp_constraint = 0; + } + + if (ux500_msp_i2s_close(drvdata->msp, + is_playback ? MSP_DIR_TX : MSP_DIR_RX)) { + dev_err(dai->dev, + "%s: Error: MSP %d (%s): Unable to close i2s.\n", + __func__, dai->id, snd_pcm_stream_str(substream)); + } + + /* Disable clock */ + clk_disable(drvdata->clk); + + /* Disable regulator */ + ret = regulator_disable(drvdata->reg_vape); + if (ret < 0) + dev_err(dai->dev, + "%s: ERROR: Failed to disable regulator (%d)!\n", + __func__, ret); +} + +static int ux500_msp_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + struct snd_pcm_runtime *runtime = substream->runtime; + struct ux500_msp_config msp_config; + + dev_dbg(dai->dev, "%s: MSP %d (%s): Enter (rate = %d).\n", __func__, + dai->id, snd_pcm_stream_str(substream), runtime->rate); + + setup_msp_config(substream, dai, &msp_config); + + ret = ux500_msp_i2s_open(drvdata->msp, &msp_config); + if (ret < 0) { + dev_err(dai->dev, "%s: Error: msp_setup failed (ret = %d)!\n", + __func__, ret); + return ret; + } + + /* Set OPP-level */ + if ((drvdata->fmt & SND_SOC_DAIFMT_MASTER_MASK) && + (drvdata->msp->f_bitclk > 19200000)) { + /* If the bit-clock is higher than 19.2MHz, Vape should be + * run in 100% OPP. Only when bit-clock is used (MSP master) */ + prcmu_qos_update_requirement(PRCMU_QOS_APE_OPP, + "ux500-msp-i2s", 100); + drvdata->vape_opp_constraint = 1; + } else { + prcmu_qos_update_requirement(PRCMU_QOS_APE_OPP, + "ux500-msp-i2s", 50); + drvdata->vape_opp_constraint = 0; + } + + return ret; +} + +static int ux500_msp_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + unsigned int mask, slots_active; + struct snd_pcm_runtime *runtime = substream->runtime; + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + dev_dbg(dai->dev, "%s: MSP %d (%s): Enter.\n", + __func__, dai->id, snd_pcm_stream_str(substream)); + + switch (drvdata->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + 1, 2); + break; + + case SND_SOC_DAIFMT_DSP_B: + case SND_SOC_DAIFMT_DSP_A: + mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + drvdata->tx_mask : + drvdata->rx_mask; + + slots_active = hweight32(mask); + dev_dbg(dai->dev, "TDM-slots active: %d", slots_active); + + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + slots_active, slots_active); + break; + + default: + dev_err(dai->dev, + "%s: Error: Unsupported protocol (fmt = 0x%x)!\n", + __func__, drvdata->fmt); + return -EINVAL; + } + + return 0; +} + +static int ux500_msp_dai_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + dev_dbg(dai->dev, "%s: MSP %d: Enter.\n", __func__, dai->id); + + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_MASTER_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + break; + + default: + dev_err(dai->dev, + "%s: Error: Unsupported protocol/master (fmt = 0x%x)!\n", + __func__, drvdata->fmt); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + case SND_SOC_DAIFMT_NB_IF: + case SND_SOC_DAIFMT_IB_IF: + break; + + default: + dev_err(dai->dev, + "%s: Error: Unsupported inversion (fmt = 0x%x)!\n", + __func__, drvdata->fmt); + return -EINVAL; + } + + drvdata->fmt = fmt; + return 0; +} + +static int ux500_msp_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, + unsigned int rx_mask, + int slots, int slot_width) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + unsigned int cap; + + switch (slots) { + case 1: + cap = 0x01; + break; + case 2: + cap = 0x03; + break; + case 8: + cap = 0xFF; + break; + case 16: + cap = 0xFFFF; + break; + default: + dev_err(dai->dev, "%s: Error: Unsupported slot-count (%d)!\n", + __func__, slots); + return -EINVAL; + } + drvdata->slots = slots; + + if (!(slot_width == 16)) { + dev_err(dai->dev, "%s: Error: Unsupported slot-width (%d)!\n", + __func__, slot_width); + return -EINVAL; + } + drvdata->slot_width = slot_width; + + drvdata->tx_mask = tx_mask & cap; + drvdata->rx_mask = rx_mask & cap; + + return 0; +} + +static int ux500_msp_dai_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + dev_dbg(dai->dev, "%s: MSP %d: Enter. clk-id: %d, freq: %u.\n", + __func__, dai->id, clk_id, freq); + + switch (clk_id) { + case UX500_MSP_MASTER_CLOCK: + drvdata->master_clk = freq; + break; + + default: + dev_err(dai->dev, "%s: MSP %d: Invalid clk-id (%d)!\n", + __func__, dai->id, clk_id); + return -EINVAL; + } + + return 0; +} + +static int ux500_msp_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + int ret = 0; + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + dev_dbg(dai->dev, "%s: MSP %d (%s): Enter (msp->id = %d, cmd = %d).\n", + __func__, dai->id, snd_pcm_stream_str(substream), + (int)drvdata->msp->id, cmd); + + ret = ux500_msp_i2s_trigger(drvdata->msp, cmd, substream->stream); + + return ret; +} + +static int ux500_msp_dai_probe(struct snd_soc_dai *dai) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + + drvdata->playback_dma_data.dma_cfg = drvdata->msp->dma_cfg_tx; + drvdata->capture_dma_data.dma_cfg = drvdata->msp->dma_cfg_rx; + + dai->playback_dma_data = &drvdata->playback_dma_data; + dai->capture_dma_data = &drvdata->capture_dma_data; + + drvdata->playback_dma_data.data_size = drvdata->slot_width; + drvdata->capture_dma_data.data_size = drvdata->slot_width; + + return 0; +} + +static struct snd_soc_dai_ops ux500_msp_dai_ops[] = { + { + .set_sysclk = ux500_msp_dai_set_dai_sysclk, + .set_fmt = ux500_msp_dai_set_dai_fmt, + .set_tdm_slot = ux500_msp_dai_set_tdm_slot, + .startup = ux500_msp_dai_startup, + .shutdown = ux500_msp_dai_shutdown, + .prepare = ux500_msp_dai_prepare, + .trigger = ux500_msp_dai_trigger, + .hw_params = ux500_msp_dai_hw_params, + } +}; + +static struct snd_soc_dai_driver ux500_msp_dai_drv[UX500_NBR_OF_DAI] = { + { + .name = "ux500-msp-i2s.0", + .probe = ux500_msp_dai_probe, + .id = 0, + .suspend = NULL, + .resume = NULL, + .playback = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .capture = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .ops = ux500_msp_dai_ops, + }, + { + .name = "ux500-msp-i2s.1", + .probe = ux500_msp_dai_probe, + .id = 1, + .suspend = NULL, + .resume = NULL, + .playback = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .capture = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .ops = ux500_msp_dai_ops, + }, + { + .name = "ux500-msp-i2s.2", + .id = 2, + .probe = ux500_msp_dai_probe, + .suspend = NULL, + .resume = NULL, + .playback = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .capture = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .ops = ux500_msp_dai_ops, + }, + { + .name = "ux500-msp-i2s.3", + .probe = ux500_msp_dai_probe, + .id = 3, + .suspend = NULL, + .resume = NULL, + .playback = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .capture = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .ops = ux500_msp_dai_ops, + }, +}; + +static int __devinit ux500_msp_drv_probe(struct platform_device *pdev) +{ + struct ux500_msp_i2s_drvdata *drvdata; + int ret = 0; + + dev_dbg(&pdev->dev, "%s: Enter (pdev->name = %s).\n", __func__, + pdev->name); + + drvdata = devm_kzalloc(&pdev->dev, + sizeof(struct ux500_msp_i2s_drvdata), + GFP_KERNEL); + drvdata->fmt = 0; + drvdata->slots = 1; + drvdata->tx_mask = 0x01; + drvdata->rx_mask = 0x01; + drvdata->slot_width = 16; + drvdata->master_clk = MSP_INPUT_FREQ_APB; + + drvdata->reg_vape = devm_regulator_get(&pdev->dev, "v-ape"); + if (IS_ERR(drvdata->reg_vape)) { + ret = (int)PTR_ERR(drvdata->reg_vape); + dev_err(&pdev->dev, + "%s: ERROR: Failed to get Vape supply (%d)!\n", + __func__, ret); + return ret; + } + prcmu_qos_add_requirement(PRCMU_QOS_APE_OPP, (char *)pdev->name, 50); + + drvdata->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(drvdata->clk)) { + ret = (int)PTR_ERR(drvdata->clk); + dev_err(&pdev->dev, "%s: ERROR: clk_get failed (%d)!\n", + __func__, ret); + goto err_clk; + } + + ret = ux500_msp_i2s_init_msp(pdev, &drvdata->msp, + pdev->dev.platform_data); + if (!drvdata->msp) { + dev_err(&pdev->dev, + "%s: ERROR: Failed to init MSP-struct (%d)!", + __func__, ret); + goto err_init_msp; + } + dev_set_drvdata(&pdev->dev, drvdata); + + ret = snd_soc_register_dai(&pdev->dev, + &ux500_msp_dai_drv[drvdata->msp->id]); + if (ret < 0) { + dev_err(&pdev->dev, "Error: %s: Failed to register MSP%d!\n", + __func__, drvdata->msp->id); + goto err_init_msp; + } + + return 0; + +err_init_msp: + clk_put(drvdata->clk); + +err_clk: + devm_regulator_put(drvdata->reg_vape); + + return ret; +} + +static int __devexit ux500_msp_drv_remove(struct platform_device *pdev) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(ux500_msp_dai_drv)); + + devm_regulator_put(drvdata->reg_vape); + prcmu_qos_remove_requirement(PRCMU_QOS_APE_OPP, "ux500_msp_i2s"); + + clk_put(drvdata->clk); + + ux500_msp_i2s_cleanup_msp(pdev, drvdata->msp); + + return 0; +} + +static struct platform_driver msp_i2s_driver = { + .driver = { + .name = "ux500-msp-i2s", + .owner = THIS_MODULE, + }, + .probe = ux500_msp_drv_probe, + .remove = ux500_msp_drv_remove, +}; +module_platform_driver(msp_i2s_driver); + +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ux500/ux500_msp_dai.h b/sound/soc/ux500/ux500_msp_dai.h new file mode 100644 index 000000000000..98202a34a5dd --- /dev/null +++ b/sound/soc/ux500/ux500_msp_dai.h @@ -0,0 +1,79 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com>, + * Roger Nilsson <roger.xr.nilsson@stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#ifndef UX500_msp_dai_H +#define UX500_msp_dai_H + +#include <linux/types.h> +#include <linux/spinlock.h> + +#include "ux500_msp_i2s.h" + +#define UX500_NBR_OF_DAI 4 + +#define UX500_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +#define UX500_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) + +#define FRAME_PER_SINGLE_SLOT_8_KHZ 31 +#define FRAME_PER_SINGLE_SLOT_16_KHZ 124 +#define FRAME_PER_SINGLE_SLOT_44_1_KHZ 63 +#define FRAME_PER_SINGLE_SLOT_48_KHZ 49 +#define FRAME_PER_2_SLOTS 31 +#define FRAME_PER_8_SLOTS 138 +#define FRAME_PER_16_SLOTS 277 + +#ifndef CONFIG_SND_SOC_UX500_AB5500 +#define UX500_MSP_INTERNAL_CLOCK_FREQ 40000000 +#define UX500_MSP1_INTERNAL_CLOCK_FREQ UX500_MSP_INTERNAL_CLOCK_FREQ +#else +#define UX500_MSP_INTERNAL_CLOCK_FREQ 13000000 +#define UX500_MSP1_INTERNAL_CLOCK_FREQ (UX500_MSP_INTERNAL_CLOCK_FREQ * 2) +#endif + +#define UX500_MSP_MIN_CHANNELS 1 +#define UX500_MSP_MAX_CHANNELS 8 + +#define PLAYBACK_CONFIGURED 1 +#define CAPTURE_CONFIGURED 2 + +enum ux500_msp_clock_id { + UX500_MSP_MASTER_CLOCK, +}; + +struct ux500_msp_i2s_drvdata { + struct ux500_msp *msp; + struct regulator *reg_vape; + struct ux500_msp_dma_params playback_dma_data; + struct ux500_msp_dma_params capture_dma_data; + unsigned int fmt; + unsigned int tx_mask; + unsigned int rx_mask; + int slots; + int slot_width; + u8 configured; + int data_delay; + + /* Clocks */ + unsigned int master_clk; + struct clk *clk; + + /* Regulators */ + int vape_opp_constraint; +}; + +int ux500_msp_dai_set_data_delay(struct snd_soc_dai *dai, int delay); + +#endif diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c new file mode 100644 index 000000000000..ee14d2dac2f5 --- /dev/null +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -0,0 +1,742 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com>, + * Roger Nilsson <roger.xr.nilsson@stericsson.com>, + * Sandeep Kaushik <sandeep.kaushik@st.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/slab.h> + +#include <mach/hardware.h> +#include <mach/board-mop500-msp.h> + +#include <sound/soc.h> + +#include "ux500_msp_i2s.h" + + /* Protocol desciptors */ +static const struct msp_protdesc prot_descs[] = { + { /* I2S */ + MSP_SINGLE_PHASE, + MSP_SINGLE_PHASE, + MSP_PHASE2_START_MODE_IMEDIATE, + MSP_PHASE2_START_MODE_IMEDIATE, + MSP_BTF_MS_BIT_FIRST, + MSP_BTF_MS_BIT_FIRST, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_ELEM_LEN_32, + MSP_ELEM_LEN_32, + MSP_ELEM_LEN_32, + MSP_ELEM_LEN_32, + MSP_DELAY_1, + MSP_DELAY_1, + MSP_RISING_EDGE, + MSP_FALLING_EDGE, + MSP_FSYNC_POL_ACT_LO, + MSP_FSYNC_POL_ACT_LO, + MSP_SWAP_NONE, + MSP_SWAP_NONE, + MSP_COMPRESS_MODE_LINEAR, + MSP_EXPAND_MODE_LINEAR, + MSP_FSYNC_IGNORE, + 31, + 15, + 32, + }, { /* PCM */ + MSP_DUAL_PHASE, + MSP_DUAL_PHASE, + MSP_PHASE2_START_MODE_FSYNC, + MSP_PHASE2_START_MODE_FSYNC, + MSP_BTF_MS_BIT_FIRST, + MSP_BTF_MS_BIT_FIRST, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_ELEM_LEN_16, + MSP_ELEM_LEN_16, + MSP_ELEM_LEN_16, + MSP_ELEM_LEN_16, + MSP_DELAY_0, + MSP_DELAY_0, + MSP_RISING_EDGE, + MSP_FALLING_EDGE, + MSP_FSYNC_POL_ACT_HI, + MSP_FSYNC_POL_ACT_HI, + MSP_SWAP_NONE, + MSP_SWAP_NONE, + MSP_COMPRESS_MODE_LINEAR, + MSP_EXPAND_MODE_LINEAR, + MSP_FSYNC_IGNORE, + 255, + 0, + 256, + }, { /* Companded PCM */ + MSP_SINGLE_PHASE, + MSP_SINGLE_PHASE, + MSP_PHASE2_START_MODE_FSYNC, + MSP_PHASE2_START_MODE_FSYNC, + MSP_BTF_MS_BIT_FIRST, + MSP_BTF_MS_BIT_FIRST, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_FRAME_LEN_1, + MSP_ELEM_LEN_8, + MSP_ELEM_LEN_8, + MSP_ELEM_LEN_8, + MSP_ELEM_LEN_8, + MSP_DELAY_0, + MSP_DELAY_0, + MSP_RISING_EDGE, + MSP_RISING_EDGE, + MSP_FSYNC_POL_ACT_HI, + MSP_FSYNC_POL_ACT_HI, + MSP_SWAP_NONE, + MSP_SWAP_NONE, + MSP_COMPRESS_MODE_LINEAR, + MSP_EXPAND_MODE_LINEAR, + MSP_FSYNC_IGNORE, + 255, + 0, + 256, + }, +}; + +static void set_prot_desc_tx(struct ux500_msp *msp, + struct msp_protdesc *protdesc, + enum msp_data_size data_size) +{ + u32 temp_reg = 0; + + temp_reg |= MSP_P2_ENABLE_BIT(protdesc->tx_phase_mode); + temp_reg |= MSP_P2_START_MODE_BIT(protdesc->tx_phase2_start_mode); + temp_reg |= MSP_P1_FRAME_LEN_BITS(protdesc->tx_frame_len_1); + temp_reg |= MSP_P2_FRAME_LEN_BITS(protdesc->tx_frame_len_2); + if (msp->def_elem_len) { + temp_reg |= MSP_P1_ELEM_LEN_BITS(protdesc->tx_elem_len_1); + temp_reg |= MSP_P2_ELEM_LEN_BITS(protdesc->tx_elem_len_2); + } else { + temp_reg |= MSP_P1_ELEM_LEN_BITS(data_size); + temp_reg |= MSP_P2_ELEM_LEN_BITS(data_size); + } + temp_reg |= MSP_DATA_DELAY_BITS(protdesc->tx_data_delay); + temp_reg |= MSP_SET_ENDIANNES_BIT(protdesc->tx_byte_order); + temp_reg |= MSP_FSYNC_POL(protdesc->tx_fsync_pol); + temp_reg |= MSP_DATA_WORD_SWAP(protdesc->tx_half_word_swap); + temp_reg |= MSP_SET_COMPANDING_MODE(protdesc->compression_mode); + temp_reg |= MSP_SET_FSYNC_IGNORE(protdesc->frame_sync_ignore); + + writel(temp_reg, msp->registers + MSP_TCF); +} + +static void set_prot_desc_rx(struct ux500_msp *msp, + struct msp_protdesc *protdesc, + enum msp_data_size data_size) +{ + u32 temp_reg = 0; + + temp_reg |= MSP_P2_ENABLE_BIT(protdesc->rx_phase_mode); + temp_reg |= MSP_P2_START_MODE_BIT(protdesc->rx_phase2_start_mode); + temp_reg |= MSP_P1_FRAME_LEN_BITS(protdesc->rx_frame_len_1); + temp_reg |= MSP_P2_FRAME_LEN_BITS(protdesc->rx_frame_len_2); + if (msp->def_elem_len) { + temp_reg |= MSP_P1_ELEM_LEN_BITS(protdesc->rx_elem_len_1); + temp_reg |= MSP_P2_ELEM_LEN_BITS(protdesc->rx_elem_len_2); + } else { + temp_reg |= MSP_P1_ELEM_LEN_BITS(data_size); + temp_reg |= MSP_P2_ELEM_LEN_BITS(data_size); + } + + temp_reg |= MSP_DATA_DELAY_BITS(protdesc->rx_data_delay); + temp_reg |= MSP_SET_ENDIANNES_BIT(protdesc->rx_byte_order); + temp_reg |= MSP_FSYNC_POL(protdesc->rx_fsync_pol); + temp_reg |= MSP_DATA_WORD_SWAP(protdesc->rx_half_word_swap); + temp_reg |= MSP_SET_COMPANDING_MODE(protdesc->expansion_mode); + temp_reg |= MSP_SET_FSYNC_IGNORE(protdesc->frame_sync_ignore); + + writel(temp_reg, msp->registers + MSP_RCF); +} + +static int configure_protocol(struct ux500_msp *msp, + struct ux500_msp_config *config) +{ + struct msp_protdesc *protdesc; + enum msp_data_size data_size; + u32 temp_reg = 0; + + data_size = config->data_size; + msp->def_elem_len = config->def_elem_len; + if (config->default_protdesc == 1) { + if (config->protocol >= MSP_INVALID_PROTOCOL) { + dev_err(msp->dev, "%s: ERROR: Invalid protocol!\n", + __func__); + return -EINVAL; + } + protdesc = + (struct msp_protdesc *)&prot_descs[config->protocol]; + } else { + protdesc = (struct msp_protdesc *)&config->protdesc; + } + + if (data_size < MSP_DATA_BITS_DEFAULT || data_size > MSP_DATA_BITS_32) { + dev_err(msp->dev, + "%s: ERROR: Invalid data-size requested (data_size = %d)!\n", + __func__, data_size); + return -EINVAL; + } + + if (config->direction & MSP_DIR_TX) + set_prot_desc_tx(msp, protdesc, data_size); + if (config->direction & MSP_DIR_RX) + set_prot_desc_rx(msp, protdesc, data_size); + + /* The code below should not be separated. */ + temp_reg = readl(msp->registers + MSP_GCR) & ~TX_CLK_POL_RISING; + temp_reg |= MSP_TX_CLKPOL_BIT(~protdesc->tx_clk_pol); + writel(temp_reg, msp->registers + MSP_GCR); + temp_reg = readl(msp->registers + MSP_GCR) & ~RX_CLK_POL_RISING; + temp_reg |= MSP_RX_CLKPOL_BIT(protdesc->rx_clk_pol); + writel(temp_reg, msp->registers + MSP_GCR); + + return 0; +} + +static int setup_bitclk(struct ux500_msp *msp, struct ux500_msp_config *config) +{ + u32 reg_val_GCR; + u32 frame_per = 0; + u32 sck_div = 0; + u32 frame_width = 0; + u32 temp_reg = 0; + struct msp_protdesc *protdesc = NULL; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR & ~SRG_ENABLE, msp->registers + MSP_GCR); + + if (config->default_protdesc) + protdesc = + (struct msp_protdesc *)&prot_descs[config->protocol]; + else + protdesc = (struct msp_protdesc *)&config->protdesc; + + switch (config->protocol) { + case MSP_PCM_PROTOCOL: + case MSP_PCM_COMPAND_PROTOCOL: + frame_width = protdesc->frame_width; + sck_div = config->f_inputclk / (config->frame_freq * + (protdesc->clocks_per_frame)); + frame_per = protdesc->frame_period; + break; + case MSP_I2S_PROTOCOL: + frame_width = protdesc->frame_width; + sck_div = config->f_inputclk / (config->frame_freq * + (protdesc->clocks_per_frame)); + frame_per = protdesc->frame_period; + break; + default: + dev_err(msp->dev, "%s: ERROR: Unknown protocol (%d)!\n", + __func__, + config->protocol); + return -EINVAL; + } + + temp_reg = (sck_div - 1) & SCK_DIV_MASK; + temp_reg |= FRAME_WIDTH_BITS(frame_width); + temp_reg |= FRAME_PERIOD_BITS(frame_per); + writel(temp_reg, msp->registers + MSP_SRG); + + msp->f_bitclk = (config->f_inputclk)/(sck_div + 1); + + /* Enable bit-clock */ + udelay(100); + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | SRG_ENABLE, msp->registers + MSP_GCR); + udelay(100); + + return 0; +} + +static int configure_multichannel(struct ux500_msp *msp, + struct ux500_msp_config *config) +{ + struct msp_protdesc *protdesc; + struct msp_multichannel_config *mcfg; + u32 reg_val_MCR; + + if (config->default_protdesc == 1) { + if (config->protocol >= MSP_INVALID_PROTOCOL) { + dev_err(msp->dev, + "%s: ERROR: Invalid protocol (%d)!\n", + __func__, config->protocol); + return -EINVAL; + } + protdesc = (struct msp_protdesc *) + &prot_descs[config->protocol]; + } else { + protdesc = (struct msp_protdesc *)&config->protdesc; + } + + mcfg = &config->multichannel_config; + if (mcfg->tx_multichannel_enable) { + if (protdesc->tx_phase_mode == MSP_SINGLE_PHASE) { + reg_val_MCR = readl(msp->registers + MSP_MCR); + writel(reg_val_MCR | (mcfg->tx_multichannel_enable ? + 1 << TMCEN_BIT : 0), + msp->registers + MSP_MCR); + writel(mcfg->tx_channel_0_enable, + msp->registers + MSP_TCE0); + writel(mcfg->tx_channel_1_enable, + msp->registers + MSP_TCE1); + writel(mcfg->tx_channel_2_enable, + msp->registers + MSP_TCE2); + writel(mcfg->tx_channel_3_enable, + msp->registers + MSP_TCE3); + } else { + dev_err(msp->dev, + "%s: ERROR: Only single-phase supported (TX-mode: %d)!\n", + __func__, protdesc->tx_phase_mode); + return -EINVAL; + } + } + if (mcfg->rx_multichannel_enable) { + if (protdesc->rx_phase_mode == MSP_SINGLE_PHASE) { + reg_val_MCR = readl(msp->registers + MSP_MCR); + writel(reg_val_MCR | (mcfg->rx_multichannel_enable ? + 1 << RMCEN_BIT : 0), + msp->registers + MSP_MCR); + writel(mcfg->rx_channel_0_enable, + msp->registers + MSP_RCE0); + writel(mcfg->rx_channel_1_enable, + msp->registers + MSP_RCE1); + writel(mcfg->rx_channel_2_enable, + msp->registers + MSP_RCE2); + writel(mcfg->rx_channel_3_enable, + msp->registers + MSP_RCE3); + } else { + dev_err(msp->dev, + "%s: ERROR: Only single-phase supported (RX-mode: %d)!\n", + __func__, protdesc->rx_phase_mode); + return -EINVAL; + } + if (mcfg->rx_comparison_enable_mode) { + reg_val_MCR = readl(msp->registers + MSP_MCR); + writel(reg_val_MCR | + (mcfg->rx_comparison_enable_mode << RCMPM_BIT), + msp->registers + MSP_MCR); + + writel(mcfg->comparison_mask, + msp->registers + MSP_RCM); + writel(mcfg->comparison_value, + msp->registers + MSP_RCV); + + } + } + + return 0; +} + +static int enable_msp(struct ux500_msp *msp, struct ux500_msp_config *config) +{ + int status = 0; + u32 reg_val_DMACR, reg_val_GCR; + + /* Check msp state whether in RUN or CONFIGURED Mode */ + if ((msp->msp_state == MSP_STATE_IDLE) && (msp->plat_init)) { + status = msp->plat_init(); + if (status) { + dev_err(msp->dev, "%s: ERROR: Failed to init MSP (%d)!\n", + __func__, status); + return status; + } + } + + /* Configure msp with protocol dependent settings */ + configure_protocol(msp, config); + setup_bitclk(msp, config); + if (config->multichannel_configured == 1) { + status = configure_multichannel(msp, config); + if (status) + dev_warn(msp->dev, + "%s: WARN: configure_multichannel failed (%d)!\n", + __func__, status); + } + + /* Make sure the correct DMA-directions are configured */ + if ((config->direction & MSP_DIR_RX) && (!msp->dma_cfg_rx)) { + dev_err(msp->dev, "%s: ERROR: MSP RX-mode is not configured!", + __func__); + return -EINVAL; + } + if ((config->direction == MSP_DIR_TX) && (!msp->dma_cfg_tx)) { + dev_err(msp->dev, "%s: ERROR: MSP TX-mode is not configured!", + __func__); + return -EINVAL; + } + + reg_val_DMACR = readl(msp->registers + MSP_DMACR); + if (config->direction & MSP_DIR_RX) + reg_val_DMACR |= RX_DMA_ENABLE; + if (config->direction & MSP_DIR_TX) + reg_val_DMACR |= TX_DMA_ENABLE; + writel(reg_val_DMACR, msp->registers + MSP_DMACR); + + writel(config->iodelay, msp->registers + MSP_IODLY); + + /* Enable frame generation logic */ + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | FRAME_GEN_ENABLE, msp->registers + MSP_GCR); + + return status; +} + +static void flush_fifo_rx(struct ux500_msp *msp) +{ + u32 reg_val_DR, reg_val_GCR, reg_val_FLR; + u32 limit = 32; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | RX_ENABLE, msp->registers + MSP_GCR); + + reg_val_FLR = readl(msp->registers + MSP_FLR); + while (!(reg_val_FLR & RX_FIFO_EMPTY) && limit--) { + reg_val_DR = readl(msp->registers + MSP_DR); + reg_val_FLR = readl(msp->registers + MSP_FLR); + } + + writel(reg_val_GCR, msp->registers + MSP_GCR); +} + +static void flush_fifo_tx(struct ux500_msp *msp) +{ + u32 reg_val_TSTDR, reg_val_GCR, reg_val_FLR; + u32 limit = 32; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | TX_ENABLE, msp->registers + MSP_GCR); + writel(MSP_ITCR_ITEN | MSP_ITCR_TESTFIFO, msp->registers + MSP_ITCR); + + reg_val_FLR = readl(msp->registers + MSP_FLR); + while (!(reg_val_FLR & TX_FIFO_EMPTY) && limit--) { + reg_val_TSTDR = readl(msp->registers + MSP_TSTDR); + reg_val_FLR = readl(msp->registers + MSP_FLR); + } + writel(0x0, msp->registers + MSP_ITCR); + writel(reg_val_GCR, msp->registers + MSP_GCR); +} + +int ux500_msp_i2s_open(struct ux500_msp *msp, + struct ux500_msp_config *config) +{ + u32 old_reg, new_reg, mask; + int res; + unsigned int tx_sel, rx_sel, tx_busy, rx_busy; + + if (in_interrupt()) { + dev_err(msp->dev, + "%s: ERROR: Open called in interrupt context!\n", + __func__); + return -1; + } + + tx_sel = (config->direction & MSP_DIR_TX) > 0; + rx_sel = (config->direction & MSP_DIR_RX) > 0; + if (!tx_sel && !rx_sel) { + dev_err(msp->dev, "%s: Error: No direction selected!\n", + __func__); + return -EINVAL; + } + + tx_busy = (msp->dir_busy & MSP_DIR_TX) > 0; + rx_busy = (msp->dir_busy & MSP_DIR_RX) > 0; + if (tx_busy && tx_sel) { + dev_err(msp->dev, "%s: Error: TX is in use!\n", __func__); + return -EBUSY; + } + if (rx_busy && rx_sel) { + dev_err(msp->dev, "%s: Error: RX is in use!\n", __func__); + return -EBUSY; + } + + msp->dir_busy |= (tx_sel ? MSP_DIR_TX : 0) | (rx_sel ? MSP_DIR_RX : 0); + + /* First do the global config register */ + mask = RX_CLK_SEL_MASK | TX_CLK_SEL_MASK | RX_FSYNC_MASK | + TX_FSYNC_MASK | RX_SYNC_SEL_MASK | TX_SYNC_SEL_MASK | + RX_FIFO_ENABLE_MASK | TX_FIFO_ENABLE_MASK | SRG_CLK_SEL_MASK | + LOOPBACK_MASK | TX_EXTRA_DELAY_MASK; + + new_reg = (config->tx_clk_sel | config->rx_clk_sel | + config->rx_fsync_pol | config->tx_fsync_pol | + config->rx_fsync_sel | config->tx_fsync_sel | + config->rx_fifo_config | config->tx_fifo_config | + config->srg_clk_sel | config->loopback_enable | + config->tx_data_enable); + + old_reg = readl(msp->registers + MSP_GCR); + old_reg &= ~mask; + new_reg |= old_reg; + writel(new_reg, msp->registers + MSP_GCR); + + res = enable_msp(msp, config); + if (res < 0) { + dev_err(msp->dev, "%s: ERROR: enable_msp failed (%d)!\n", + __func__, res); + return -EBUSY; + } + if (config->loopback_enable & 0x80) + msp->loopback_enable = 1; + + /* Flush FIFOs */ + flush_fifo_tx(msp); + flush_fifo_rx(msp); + + msp->msp_state = MSP_STATE_CONFIGURED; + return 0; +} + +static void disable_msp_rx(struct ux500_msp *msp) +{ + u32 reg_val_GCR, reg_val_DMACR, reg_val_IMSC; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR & ~RX_ENABLE, msp->registers + MSP_GCR); + reg_val_DMACR = readl(msp->registers + MSP_DMACR); + writel(reg_val_DMACR & ~RX_DMA_ENABLE, msp->registers + MSP_DMACR); + reg_val_IMSC = readl(msp->registers + MSP_IMSC); + writel(reg_val_IMSC & + ~(RX_SERVICE_INT | RX_OVERRUN_ERROR_INT), + msp->registers + MSP_IMSC); + + msp->dir_busy &= ~MSP_DIR_RX; +} + +static void disable_msp_tx(struct ux500_msp *msp) +{ + u32 reg_val_GCR, reg_val_DMACR, reg_val_IMSC; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR & ~TX_ENABLE, msp->registers + MSP_GCR); + reg_val_DMACR = readl(msp->registers + MSP_DMACR); + writel(reg_val_DMACR & ~TX_DMA_ENABLE, msp->registers + MSP_DMACR); + reg_val_IMSC = readl(msp->registers + MSP_IMSC); + writel(reg_val_IMSC & + ~(TX_SERVICE_INT | TX_UNDERRUN_ERR_INT), + msp->registers + MSP_IMSC); + + msp->dir_busy &= ~MSP_DIR_TX; +} + +static int disable_msp(struct ux500_msp *msp, unsigned int dir) +{ + u32 reg_val_GCR; + int status = 0; + unsigned int disable_tx, disable_rx; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + disable_tx = dir & MSP_DIR_TX; + disable_rx = dir & MSP_DIR_TX; + if (disable_tx && disable_rx) { + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | LOOPBACK_MASK, + msp->registers + MSP_GCR); + + /* Flush TX-FIFO */ + flush_fifo_tx(msp); + + /* Disable TX-channel */ + writel((readl(msp->registers + MSP_GCR) & + (~TX_ENABLE)), msp->registers + MSP_GCR); + + /* Flush RX-FIFO */ + flush_fifo_rx(msp); + + /* Disable Loopback and Receive channel */ + writel((readl(msp->registers + MSP_GCR) & + (~(RX_ENABLE | LOOPBACK_MASK))), + msp->registers + MSP_GCR); + + disable_msp_tx(msp); + disable_msp_rx(msp); + } else if (disable_tx) + disable_msp_tx(msp); + else if (disable_rx) + disable_msp_rx(msp); + + return status; +} + +int ux500_msp_i2s_trigger(struct ux500_msp *msp, int cmd, int direction) +{ + u32 reg_val_GCR, enable_bit; + + if (msp->msp_state == MSP_STATE_IDLE) { + dev_err(msp->dev, "%s: ERROR: MSP is not configured!\n", + __func__); + return -EINVAL; + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + enable_bit = TX_ENABLE; + else + enable_bit = RX_ENABLE; + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | enable_bit, msp->registers + MSP_GCR); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + disable_msp_tx(msp); + else + disable_msp_rx(msp); + break; + default: + return -EINVAL; + break; + } + + return 0; +} + +int ux500_msp_i2s_close(struct ux500_msp *msp, unsigned int dir) +{ + int status = 0; + + dev_dbg(msp->dev, "%s: Enter (dir = 0x%01x).\n", __func__, dir); + + status = disable_msp(msp, dir); + if (msp->dir_busy == 0) { + /* disable sample rate and frame generators */ + msp->msp_state = MSP_STATE_IDLE; + writel((readl(msp->registers + MSP_GCR) & + (~(FRAME_GEN_ENABLE | SRG_ENABLE))), + msp->registers + MSP_GCR); + if (msp->plat_exit) + status = msp->plat_exit(); + if (status) + dev_warn(msp->dev, + "%s: WARN: ux500_msp_i2s_exit failed (%d)!\n", + __func__, status); + writel(0, msp->registers + MSP_GCR); + writel(0, msp->registers + MSP_TCF); + writel(0, msp->registers + MSP_RCF); + writel(0, msp->registers + MSP_DMACR); + writel(0, msp->registers + MSP_SRG); + writel(0, msp->registers + MSP_MCR); + writel(0, msp->registers + MSP_RCM); + writel(0, msp->registers + MSP_RCV); + writel(0, msp->registers + MSP_TCE0); + writel(0, msp->registers + MSP_TCE1); + writel(0, msp->registers + MSP_TCE2); + writel(0, msp->registers + MSP_TCE3); + writel(0, msp->registers + MSP_RCE0); + writel(0, msp->registers + MSP_RCE1); + writel(0, msp->registers + MSP_RCE2); + writel(0, msp->registers + MSP_RCE3); + } + + return status; + +} + +int ux500_msp_i2s_init_msp(struct platform_device *pdev, + struct ux500_msp **msp_p, + struct msp_i2s_platform_data *platform_data) +{ + int ret = 0; + struct resource *res = NULL; + struct i2s_controller *i2s_cont; + struct ux500_msp *msp; + + dev_dbg(&pdev->dev, "%s: Enter (name: %s, id: %d).\n", __func__, + pdev->name, platform_data->id); + + *msp_p = devm_kzalloc(&pdev->dev, sizeof(struct ux500_msp), GFP_KERNEL); + msp = *msp_p; + + msp->id = platform_data->id; + msp->dev = &pdev->dev; + msp->plat_init = platform_data->msp_i2s_init; + msp->plat_exit = platform_data->msp_i2s_exit; + msp->dma_cfg_rx = platform_data->msp_i2s_dma_rx; + msp->dma_cfg_tx = platform_data->msp_i2s_dma_tx; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) { + dev_err(&pdev->dev, "%s: ERROR: Unable to get resource!\n", + __func__); + ret = -ENOMEM; + goto err_res; + } + + msp->registers = ioremap(res->start, (res->end - res->start + 1)); + if (msp->registers == NULL) { + dev_err(&pdev->dev, "%s: ERROR: ioremap failed!\n", __func__); + ret = -ENOMEM; + goto err_res; + } + + msp->msp_state = MSP_STATE_IDLE; + msp->loopback_enable = 0; + + /* I2S-controller is allocated and added in I2S controller class. */ + i2s_cont = devm_kzalloc(&pdev->dev, sizeof(*i2s_cont), GFP_KERNEL); + if (!i2s_cont) { + dev_err(&pdev->dev, + "%s: ERROR: Failed to allocate I2S-controller!\n", + __func__); + goto err_i2s_cont; + } + i2s_cont->dev.parent = &pdev->dev; + i2s_cont->data = (void *)msp; + i2s_cont->id = (s16)msp->id; + snprintf(i2s_cont->name, sizeof(i2s_cont->name), "ux500-msp-i2s.%04x", + msp->id); + dev_dbg(&pdev->dev, "I2S device-name: '%s'\n", i2s_cont->name); + msp->i2s_cont = i2s_cont; + + return 0; + +err_i2s_cont: + iounmap(msp->registers); + +err_res: + devm_kfree(&pdev->dev, msp); + + return ret; +} + +void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev, + struct ux500_msp *msp) +{ + dev_dbg(msp->dev, "%s: Enter (id = %d).\n", __func__, msp->id); + + device_unregister(&msp->i2s_cont->dev); + devm_kfree(&pdev->dev, msp->i2s_cont); + + iounmap(msp->registers); + + devm_kfree(&pdev->dev, msp); +} + +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h new file mode 100644 index 000000000000..7f71b4a0d4bc --- /dev/null +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -0,0 +1,553 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com>, + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + + +#ifndef UX500_MSP_I2S_H +#define UX500_MSP_I2S_H + +#include <linux/platform_device.h> + +#include <mach/board-mop500-msp.h> + +#define MSP_INPUT_FREQ_APB 48000000 + +/*** Stereo mode. Used for APB data accesses as 16 bits accesses (mono), + * 32 bits accesses (stereo). + ***/ +enum msp_stereo_mode { + MSP_MONO, + MSP_STEREO +}; + +/* Direction (Transmit/Receive mode) */ +enum msp_direction { + MSP_TX = 1, + MSP_RX = 2 +}; + +/* Transmit and receive configuration register */ +#define MSP_BIG_ENDIAN 0x00000000 +#define MSP_LITTLE_ENDIAN 0x00001000 +#define MSP_UNEXPECTED_FS_ABORT 0x00000000 +#define MSP_UNEXPECTED_FS_IGNORE 0x00008000 +#define MSP_NON_MODE_BIT_MASK 0x00009000 + +/* Global configuration register */ +#define RX_ENABLE 0x00000001 +#define RX_FIFO_ENABLE 0x00000002 +#define RX_SYNC_SRG 0x00000010 +#define RX_CLK_POL_RISING 0x00000020 +#define RX_CLK_SEL_SRG 0x00000040 +#define TX_ENABLE 0x00000100 +#define TX_FIFO_ENABLE 0x00000200 +#define TX_SYNC_SRG_PROG 0x00001800 +#define TX_SYNC_SRG_AUTO 0x00001000 +#define TX_CLK_POL_RISING 0x00002000 +#define TX_CLK_SEL_SRG 0x00004000 +#define TX_EXTRA_DELAY_ENABLE 0x00008000 +#define SRG_ENABLE 0x00010000 +#define FRAME_GEN_ENABLE 0x00100000 +#define SRG_CLK_SEL_APB 0x00000000 +#define RX_FIFO_SYNC_HI 0x00000000 +#define TX_FIFO_SYNC_HI 0x00000000 +#define SPI_CLK_MODE_NORMAL 0x00000000 + +#define MSP_FRAME_SIZE_AUTO -1 + +#define MSP_DR 0x00 +#define MSP_GCR 0x04 +#define MSP_TCF 0x08 +#define MSP_RCF 0x0c +#define MSP_SRG 0x10 +#define MSP_FLR 0x14 +#define MSP_DMACR 0x18 + +#define MSP_IMSC 0x20 +#define MSP_RIS 0x24 +#define MSP_MIS 0x28 +#define MSP_ICR 0x2c +#define MSP_MCR 0x30 +#define MSP_RCV 0x34 +#define MSP_RCM 0x38 + +#define MSP_TCE0 0x40 +#define MSP_TCE1 0x44 +#define MSP_TCE2 0x48 +#define MSP_TCE3 0x4c + +#define MSP_RCE0 0x60 +#define MSP_RCE1 0x64 +#define MSP_RCE2 0x68 +#define MSP_RCE3 0x6c +#define MSP_IODLY 0x70 + +#define MSP_ITCR 0x80 +#define MSP_ITIP 0x84 +#define MSP_ITOP 0x88 +#define MSP_TSTDR 0x8c + +#define MSP_PID0 0xfe0 +#define MSP_PID1 0xfe4 +#define MSP_PID2 0xfe8 +#define MSP_PID3 0xfec + +#define MSP_CID0 0xff0 +#define MSP_CID1 0xff4 +#define MSP_CID2 0xff8 +#define MSP_CID3 0xffc + +/* Protocol dependant parameters list */ +#define RX_ENABLE_MASK BIT(0) +#define RX_FIFO_ENABLE_MASK BIT(1) +#define RX_FSYNC_MASK BIT(2) +#define DIRECT_COMPANDING_MASK BIT(3) +#define RX_SYNC_SEL_MASK BIT(4) +#define RX_CLK_POL_MASK BIT(5) +#define RX_CLK_SEL_MASK BIT(6) +#define LOOPBACK_MASK BIT(7) +#define TX_ENABLE_MASK BIT(8) +#define TX_FIFO_ENABLE_MASK BIT(9) +#define TX_FSYNC_MASK BIT(10) +#define TX_MSP_TDR_TSR BIT(11) +#define TX_SYNC_SEL_MASK (BIT(12) | BIT(11)) +#define TX_CLK_POL_MASK BIT(13) +#define TX_CLK_SEL_MASK BIT(14) +#define TX_EXTRA_DELAY_MASK BIT(15) +#define SRG_ENABLE_MASK BIT(16) +#define SRG_CLK_POL_MASK BIT(17) +#define SRG_CLK_SEL_MASK (BIT(19) | BIT(18)) +#define FRAME_GEN_EN_MASK BIT(20) +#define SPI_CLK_MODE_MASK (BIT(22) | BIT(21)) +#define SPI_BURST_MODE_MASK BIT(23) + +#define RXEN_SHIFT 0 +#define RFFEN_SHIFT 1 +#define RFSPOL_SHIFT 2 +#define DCM_SHIFT 3 +#define RFSSEL_SHIFT 4 +#define RCKPOL_SHIFT 5 +#define RCKSEL_SHIFT 6 +#define LBM_SHIFT 7 +#define TXEN_SHIFT 8 +#define TFFEN_SHIFT 9 +#define TFSPOL_SHIFT 10 +#define TFSSEL_SHIFT 11 +#define TCKPOL_SHIFT 13 +#define TCKSEL_SHIFT 14 +#define TXDDL_SHIFT 15 +#define SGEN_SHIFT 16 +#define SCKPOL_SHIFT 17 +#define SCKSEL_SHIFT 18 +#define FGEN_SHIFT 20 +#define SPICKM_SHIFT 21 +#define TBSWAP_SHIFT 28 + +#define RCKPOL_MASK BIT(0) +#define TCKPOL_MASK BIT(0) +#define SPICKM_MASK (BIT(1) | BIT(0)) +#define MSP_RX_CLKPOL_BIT(n) ((n & RCKPOL_MASK) << RCKPOL_SHIFT) +#define MSP_TX_CLKPOL_BIT(n) ((n & TCKPOL_MASK) << TCKPOL_SHIFT) + +#define P1ELEN_SHIFT 0 +#define P1FLEN_SHIFT 3 +#define DTYP_SHIFT 10 +#define ENDN_SHIFT 12 +#define DDLY_SHIFT 13 +#define FSIG_SHIFT 15 +#define P2ELEN_SHIFT 16 +#define P2FLEN_SHIFT 19 +#define P2SM_SHIFT 26 +#define P2EN_SHIFT 27 +#define FSYNC_SHIFT 15 + +#define P1ELEN_MASK 0x00000007 +#define P2ELEN_MASK 0x00070000 +#define P1FLEN_MASK 0x00000378 +#define P2FLEN_MASK 0x03780000 +#define DDLY_MASK 0x00003000 +#define DTYP_MASK 0x00000600 +#define P2SM_MASK 0x04000000 +#define P2EN_MASK 0x08000000 +#define ENDN_MASK 0x00001000 +#define TFSPOL_MASK 0x00000400 +#define TBSWAP_MASK 0x30000000 +#define COMPANDING_MODE_MASK 0x00000c00 +#define FSYNC_MASK 0x00008000 + +#define MSP_P1_ELEM_LEN_BITS(n) (n & P1ELEN_MASK) +#define MSP_P2_ELEM_LEN_BITS(n) (((n) << P2ELEN_SHIFT) & P2ELEN_MASK) +#define MSP_P1_FRAME_LEN_BITS(n) (((n) << P1FLEN_SHIFT) & P1FLEN_MASK) +#define MSP_P2_FRAME_LEN_BITS(n) (((n) << P2FLEN_SHIFT) & P2FLEN_MASK) +#define MSP_DATA_DELAY_BITS(n) (((n) << DDLY_SHIFT) & DDLY_MASK) +#define MSP_DATA_TYPE_BITS(n) (((n) << DTYP_SHIFT) & DTYP_MASK) +#define MSP_P2_START_MODE_BIT(n) ((n << P2SM_SHIFT) & P2SM_MASK) +#define MSP_P2_ENABLE_BIT(n) ((n << P2EN_SHIFT) & P2EN_MASK) +#define MSP_SET_ENDIANNES_BIT(n) ((n << ENDN_SHIFT) & ENDN_MASK) +#define MSP_FSYNC_POL(n) ((n << TFSPOL_SHIFT) & TFSPOL_MASK) +#define MSP_DATA_WORD_SWAP(n) ((n << TBSWAP_SHIFT) & TBSWAP_MASK) +#define MSP_SET_COMPANDING_MODE(n) ((n << DTYP_SHIFT) & \ + COMPANDING_MODE_MASK) +#define MSP_SET_FSYNC_IGNORE(n) ((n << FSYNC_SHIFT) & FSYNC_MASK) + +/* Flag register */ +#define RX_BUSY BIT(0) +#define RX_FIFO_EMPTY BIT(1) +#define RX_FIFO_FULL BIT(2) +#define TX_BUSY BIT(3) +#define TX_FIFO_EMPTY BIT(4) +#define TX_FIFO_FULL BIT(5) + +#define RBUSY_SHIFT 0 +#define RFE_SHIFT 1 +#define RFU_SHIFT 2 +#define TBUSY_SHIFT 3 +#define TFE_SHIFT 4 +#define TFU_SHIFT 5 + +/* Multichannel control register */ +#define RMCEN_SHIFT 0 +#define RMCSF_SHIFT 1 +#define RCMPM_SHIFT 3 +#define TMCEN_SHIFT 5 +#define TNCSF_SHIFT 6 + +/* Sample rate generator register */ +#define SCKDIV_SHIFT 0 +#define FRWID_SHIFT 10 +#define FRPER_SHIFT 16 + +#define SCK_DIV_MASK 0x0000003FF +#define FRAME_WIDTH_BITS(n) (((n) << FRWID_SHIFT) & 0x0000FC00) +#define FRAME_PERIOD_BITS(n) (((n) << FRPER_SHIFT) & 0x1FFF0000) + +/* DMA controller register */ +#define RX_DMA_ENABLE BIT(0) +#define TX_DMA_ENABLE BIT(1) + +#define RDMAE_SHIFT 0 +#define TDMAE_SHIFT 1 + +/* Interrupt Register */ +#define RX_SERVICE_INT BIT(0) +#define RX_OVERRUN_ERROR_INT BIT(1) +#define RX_FSYNC_ERR_INT BIT(2) +#define RX_FSYNC_INT BIT(3) +#define TX_SERVICE_INT BIT(4) +#define TX_UNDERRUN_ERR_INT BIT(5) +#define TX_FSYNC_ERR_INT BIT(6) +#define TX_FSYNC_INT BIT(7) +#define ALL_INT 0x000000ff + +/* MSP test control register */ +#define MSP_ITCR_ITEN BIT(0) +#define MSP_ITCR_TESTFIFO BIT(1) + +#define RMCEN_BIT 0 +#define RMCSF_BIT 1 +#define RCMPM_BIT 3 +#define TMCEN_BIT 5 +#define TNCSF_BIT 6 + +/* Single or dual phase mode */ +enum msp_phase_mode { + MSP_SINGLE_PHASE, + MSP_DUAL_PHASE +}; + +/* Frame length */ +enum msp_frame_length { + MSP_FRAME_LEN_1 = 0, + MSP_FRAME_LEN_2 = 1, + MSP_FRAME_LEN_4 = 3, + MSP_FRAME_LEN_8 = 7, + MSP_FRAME_LEN_12 = 11, + MSP_FRAME_LEN_16 = 15, + MSP_FRAME_LEN_20 = 19, + MSP_FRAME_LEN_32 = 31, + MSP_FRAME_LEN_48 = 47, + MSP_FRAME_LEN_64 = 63 +}; + +/* Element length */ +enum msp_elem_length { + MSP_ELEM_LEN_8 = 0, + MSP_ELEM_LEN_10 = 1, + MSP_ELEM_LEN_12 = 2, + MSP_ELEM_LEN_14 = 3, + MSP_ELEM_LEN_16 = 4, + MSP_ELEM_LEN_20 = 5, + MSP_ELEM_LEN_24 = 6, + MSP_ELEM_LEN_32 = 7 +}; + +enum msp_data_xfer_width { + MSP_DATA_TRANSFER_WIDTH_BYTE, + MSP_DATA_TRANSFER_WIDTH_HALFWORD, + MSP_DATA_TRANSFER_WIDTH_WORD +}; + +enum msp_frame_sync { + MSP_FSYNC_UNIGNORE = 0, + MSP_FSYNC_IGNORE = 1, +}; + +enum msp_phase2_start_mode { + MSP_PHASE2_START_MODE_IMEDIATE, + MSP_PHASE2_START_MODE_FSYNC +}; + +enum msp_btf { + MSP_BTF_MS_BIT_FIRST = 0, + MSP_BTF_LS_BIT_FIRST = 1 +}; + +enum msp_fsync_pol { + MSP_FSYNC_POL_ACT_HI = 0, + MSP_FSYNC_POL_ACT_LO = 1 +}; + +/* Data delay (in bit clock cycles) */ +enum msp_delay { + MSP_DELAY_0 = 0, + MSP_DELAY_1 = 1, + MSP_DELAY_2 = 2, + MSP_DELAY_3 = 3 +}; + +/* Configurations of clocks (transmit, receive or sample rate generator) */ +enum msp_edge { + MSP_FALLING_EDGE = 0, + MSP_RISING_EDGE = 1, +}; + +enum msp_hws { + MSP_SWAP_NONE = 0, + MSP_SWAP_BYTE_PER_WORD = 1, + MSP_SWAP_BYTE_PER_HALF_WORD = 2, + MSP_SWAP_HALF_WORD_PER_WORD = 3 +}; + +enum msp_compress_mode { + MSP_COMPRESS_MODE_LINEAR = 0, + MSP_COMPRESS_MODE_MU_LAW = 2, + MSP_COMPRESS_MODE_A_LAW = 3 +}; + +enum msp_spi_burst_mode { + MSP_SPI_BURST_MODE_DISABLE = 0, + MSP_SPI_BURST_MODE_ENABLE = 1 +}; + +enum msp_expand_mode { + MSP_EXPAND_MODE_LINEAR = 0, + MSP_EXPAND_MODE_LINEAR_SIGNED = 1, + MSP_EXPAND_MODE_MU_LAW = 2, + MSP_EXPAND_MODE_A_LAW = 3 +}; + +#define MSP_FRAME_PERIOD_IN_MONO_MODE 256 +#define MSP_FRAME_PERIOD_IN_STEREO_MODE 32 +#define MSP_FRAME_WIDTH_IN_STEREO_MODE 16 + +enum msp_protocol { + MSP_I2S_PROTOCOL, + MSP_PCM_PROTOCOL, + MSP_PCM_COMPAND_PROTOCOL, + MSP_INVALID_PROTOCOL +}; + +/* + * No of registers to backup during + * suspend resume + */ +#define MAX_MSP_BACKUP_REGS 36 + +enum enum_i2s_controller { + MSP_0_I2S_CONTROLLER = 0, + MSP_1_I2S_CONTROLLER, + MSP_2_I2S_CONTROLLER, + MSP_3_I2S_CONTROLLER, +}; + +enum i2s_direction_t { + MSP_DIR_TX = 0x01, + MSP_DIR_RX = 0x02, +}; + +enum msp_data_size { + MSP_DATA_BITS_DEFAULT = -1, + MSP_DATA_BITS_8 = 0x00, + MSP_DATA_BITS_10, + MSP_DATA_BITS_12, + MSP_DATA_BITS_14, + MSP_DATA_BITS_16, + MSP_DATA_BITS_20, + MSP_DATA_BITS_24, + MSP_DATA_BITS_32, +}; + +enum msp_state { + MSP_STATE_IDLE = 0, + MSP_STATE_CONFIGURED = 1, + MSP_STATE_RUNNING = 2, +}; + +enum msp_rx_comparison_enable_mode { + MSP_COMPARISON_DISABLED = 0, + MSP_COMPARISON_NONEQUAL_ENABLED = 2, + MSP_COMPARISON_EQUAL_ENABLED = 3 +}; + +struct msp_multichannel_config { + bool rx_multichannel_enable; + bool tx_multichannel_enable; + enum msp_rx_comparison_enable_mode rx_comparison_enable_mode; + u8 padding; + u32 comparison_value; + u32 comparison_mask; + u32 rx_channel_0_enable; + u32 rx_channel_1_enable; + u32 rx_channel_2_enable; + u32 rx_channel_3_enable; + u32 tx_channel_0_enable; + u32 tx_channel_1_enable; + u32 tx_channel_2_enable; + u32 tx_channel_3_enable; +}; + +struct msp_protdesc { + u32 rx_phase_mode; + u32 tx_phase_mode; + u32 rx_phase2_start_mode; + u32 tx_phase2_start_mode; + u32 rx_byte_order; + u32 tx_byte_order; + u32 rx_frame_len_1; + u32 rx_frame_len_2; + u32 tx_frame_len_1; + u32 tx_frame_len_2; + u32 rx_elem_len_1; + u32 rx_elem_len_2; + u32 tx_elem_len_1; + u32 tx_elem_len_2; + u32 rx_data_delay; + u32 tx_data_delay; + u32 rx_clk_pol; + u32 tx_clk_pol; + u32 rx_fsync_pol; + u32 tx_fsync_pol; + u32 rx_half_word_swap; + u32 tx_half_word_swap; + u32 compression_mode; + u32 expansion_mode; + u32 frame_sync_ignore; + u32 frame_period; + u32 frame_width; + u32 clocks_per_frame; +}; + +struct i2s_message { + enum i2s_direction_t i2s_direction; + void *txdata; + void *rxdata; + size_t txbytes; + size_t rxbytes; + int dma_flag; + int tx_offset; + int rx_offset; + bool cyclic_dma; + dma_addr_t buf_addr; + size_t buf_len; + size_t period_len; +}; + +struct i2s_controller { + struct module *owner; + unsigned int id; + unsigned int class; + const struct i2s_algorithm *algo; /* the algorithm to access the bus */ + void *data; + struct mutex bus_lock; + struct device dev; /* the controller device */ + char name[48]; +}; + +struct ux500_msp_config { + unsigned int f_inputclk; + unsigned int rx_clk_sel; + unsigned int tx_clk_sel; + unsigned int srg_clk_sel; + unsigned int rx_fsync_pol; + unsigned int tx_fsync_pol; + unsigned int rx_fsync_sel; + unsigned int tx_fsync_sel; + unsigned int rx_fifo_config; + unsigned int tx_fifo_config; + unsigned int spi_clk_mode; + unsigned int spi_burst_mode; + unsigned int loopback_enable; + unsigned int tx_data_enable; + unsigned int default_protdesc; + struct msp_protdesc protdesc; + int multichannel_configured; + struct msp_multichannel_config multichannel_config; + unsigned int direction; + unsigned int protocol; + unsigned int frame_freq; + unsigned int frame_size; + enum msp_data_size data_size; + unsigned int def_elem_len; + unsigned int iodelay; + void (*handler) (void *data); + void *tx_callback_data; + void *rx_callback_data; +}; + +struct ux500_msp { + enum enum_i2s_controller id; + void __iomem *registers; + struct device *dev; + struct i2s_controller *i2s_cont; + struct stedma40_chan_cfg *dma_cfg_rx; + struct stedma40_chan_cfg *dma_cfg_tx; + struct dma_chan *tx_pipeid; + struct dma_chan *rx_pipeid; + enum msp_state msp_state; + int (*transfer) (struct ux500_msp *msp, struct i2s_message *message); + int (*plat_init) (void); + int (*plat_exit) (void); + struct timer_list notify_timer; + int def_elem_len; + unsigned int dir_busy; + int loopback_enable; + u32 backup_regs[MAX_MSP_BACKUP_REGS]; + unsigned int f_bitclk; +}; + +struct ux500_msp_dma_params { + unsigned int data_size; + struct stedma40_chan_cfg *dma_cfg; +}; + +int ux500_msp_i2s_init_msp(struct platform_device *pdev, + struct ux500_msp **msp_p, + struct msp_i2s_platform_data *platform_data); +void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev, + struct ux500_msp *msp); +int ux500_msp_i2s_open(struct ux500_msp *msp, struct ux500_msp_config *config); +int ux500_msp_i2s_close(struct ux500_msp *msp, + unsigned int dir); +int ux500_msp_i2s_trigger(struct ux500_msp *msp, int cmd, + int direction); + +#endif diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c new file mode 100644 index 000000000000..1a04e248453c --- /dev/null +++ b/sound/soc/ux500/ux500_pcm.c @@ -0,0 +1,318 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com>, + * Roger Nilsson <roger.xr.nilsson@stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <asm/page.h> + +#include <linux/module.h> +#include <linux/dma-mapping.h> +#include <linux/dmaengine.h> +#include <linux/slab.h> + +#include <plat/ste_dma40.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +#include "ux500_msp_i2s.h" +#include "ux500_pcm.h" + +static struct snd_pcm_hardware ux500_pcm_hw_playback = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_BE, + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = UX500_PLATFORM_MIN_RATE_PLAYBACK, + .rate_max = UX500_PLATFORM_MAX_RATE_PLAYBACK, + .channels_min = UX500_PLATFORM_MIN_CHANNELS, + .channels_max = UX500_PLATFORM_MAX_CHANNELS, + .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, + .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN, + .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX, + .periods_min = UX500_PLATFORM_PERIODS_MIN, + .periods_max = UX500_PLATFORM_PERIODS_MAX, +}; + +static struct snd_pcm_hardware ux500_pcm_hw_capture = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_BE, + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = UX500_PLATFORM_MIN_RATE_CAPTURE, + .rate_max = UX500_PLATFORM_MAX_RATE_CAPTURE, + .channels_min = UX500_PLATFORM_MIN_CHANNELS, + .channels_max = UX500_PLATFORM_MAX_CHANNELS, + .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, + .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN, + .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX, + .periods_min = UX500_PLATFORM_PERIODS_MIN, + .periods_max = UX500_PLATFORM_PERIODS_MAX, +}; + +static void ux500_pcm_dma_hw_free(struct device *dev, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = runtime->dma_buffer_p; + + if (runtime->dma_area == NULL) + return; + + if (buf != &substream->dma_buffer) { + dma_free_coherent(buf->dev.dev, buf->bytes, buf->area, + buf->addr); + kfree(runtime->dma_buffer_p); + } + + snd_pcm_set_runtime_buffer(substream, NULL); +} + +static int ux500_pcm_open(struct snd_pcm_substream *substream) +{ + int stream_id = substream->pstr->stream; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct device *dev = dai->dev; + int ret; + struct ux500_msp_dma_params *dma_params; + u16 per_data_width, mem_data_width; + struct stedma40_chan_cfg *dma_cfg; + + dev_dbg(dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id, + snd_pcm_stream_str(substream)); + + dev_dbg(dev, "%s: Set runtime hwparams.\n", __func__); + if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_set_runtime_hwparams(substream, + &ux500_pcm_hw_playback); + else + snd_soc_set_runtime_hwparams(substream, + &ux500_pcm_hw_capture); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(dev, "%s: Error: snd_pcm_hw_constraints failed (%d)\n", + __func__, ret); + return ret; + } + + dev_dbg(dev, "%s: Set hw-struct for %s.\n", __func__, + snd_pcm_stream_str(substream)); + runtime->hw = (stream_id == SNDRV_PCM_STREAM_PLAYBACK) ? + ux500_pcm_hw_playback : ux500_pcm_hw_capture; + + mem_data_width = STEDMA40_HALFWORD_WIDTH; + + dma_params = snd_soc_dai_get_dma_data(dai, substream); + switch (dma_params->data_size) { + case 32: + per_data_width = STEDMA40_WORD_WIDTH; + break; + case 16: + per_data_width = STEDMA40_HALFWORD_WIDTH; + break; + case 8: + per_data_width = STEDMA40_BYTE_WIDTH; + break; + default: + per_data_width = STEDMA40_WORD_WIDTH; + dev_warn(rtd->platform->dev, + "%s: Unknown data-size (%d)! Assuming 32 bits.\n", + __func__, dma_params->data_size); + } + + dma_cfg = dma_params->dma_cfg; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dma_cfg->src_info.data_width = mem_data_width; + dma_cfg->dst_info.data_width = per_data_width; + } else { + dma_cfg->src_info.data_width = per_data_width; + dma_cfg->dst_info.data_width = mem_data_width; + } + + + ret = snd_dmaengine_pcm_open(substream, stedma40_filter, dma_cfg); + if (ret) { + dev_dbg(dai->dev, + "%s: ERROR: snd_dmaengine_pcm_open failed (%d)!\n", + __func__, ret); + return ret; + } + + snd_dmaengine_pcm_set_data(substream, dma_cfg); + + return 0; +} + +static int ux500_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *dai = rtd->cpu_dai; + + dev_dbg(dai->dev, "%s: Enter\n", __func__); + + snd_dmaengine_pcm_close(substream); + + return 0; +} + +static int ux500_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = runtime->dma_buffer_p; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret = 0; + int size; + + dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__); + + size = params_buffer_bytes(hw_params); + + if (buf) { + if (buf->bytes >= size) + goto out; + ux500_pcm_dma_hw_free(NULL, substream); + } + + if (substream->dma_buffer.area != NULL && + substream->dma_buffer.bytes >= size) { + buf = &substream->dma_buffer; + } else { + buf = kmalloc(sizeof(struct snd_dma_buffer), GFP_KERNEL); + if (!buf) + goto nomem; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = NULL; + buf->area = dma_alloc_coherent(NULL, size, &buf->addr, + GFP_KERNEL); + buf->bytes = size; + buf->private_data = NULL; + + if (!buf->area) + goto free; + } + snd_pcm_set_runtime_buffer(substream, buf); + ret = 1; + out: + runtime->dma_bytes = size; + return ret; + + free: + kfree(buf); + nomem: + return -ENOMEM; +} + +static int ux500_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__); + + ux500_pcm_dma_hw_free(NULL, substream); + + return 0; +} + +static int ux500_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->platform->dev, "%s: Enter.\n", __func__); + + return dma_mmap_coherent(NULL, vma, runtime->dma_area, + runtime->dma_addr, runtime->dma_bytes); +} + +static struct snd_pcm_ops ux500_pcm_ops = { + .open = ux500_pcm_open, + .close = ux500_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = ux500_pcm_hw_params, + .hw_free = ux500_pcm_hw_free, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer_no_residue, + .mmap = ux500_pcm_mmap +}; + +int ux500_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + + dev_dbg(rtd->platform->dev, "%s: Enter (id = '%s').\n", __func__, + pcm->id); + + pcm->info_flags = 0; + + return 0; +} + +static struct snd_soc_platform_driver ux500_pcm_soc_drv = { + .ops = &ux500_pcm_ops, + .pcm_new = ux500_pcm_new, +}; + +static int __devexit ux500_pcm_drv_probe(struct platform_device *pdev) +{ + int ret; + + ret = snd_soc_register_platform(&pdev->dev, &ux500_pcm_soc_drv); + if (ret < 0) { + dev_err(&pdev->dev, + "%s: ERROR: Failed to register platform '%s' (%d)!\n", + __func__, pdev->name, ret); + return ret; + } + + return 0; +} + +static int __devinit ux500_pcm_drv_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + + return 0; +} + +static struct platform_driver ux500_pcm_driver = { + .driver = { + .name = "ux500-pcm", + .owner = THIS_MODULE, + }, + + .probe = ux500_pcm_drv_probe, + .remove = __devexit_p(ux500_pcm_drv_remove), +}; +module_platform_driver(ux500_pcm_driver); + +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ux500/ux500_pcm.h b/sound/soc/ux500/ux500_pcm.h new file mode 100644 index 000000000000..77ed44d371e9 --- /dev/null +++ b/sound/soc/ux500/ux500_pcm.h @@ -0,0 +1,35 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja@stericsson.com>, + * Roger Nilsson <roger.xr.nilsson@stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ +#ifndef UX500_PCM_H +#define UX500_PCM_H + +#include <asm/page.h> + +#include <linux/workqueue.h> + +#define UX500_PLATFORM_MIN_RATE_PLAYBACK 8000 +#define UX500_PLATFORM_MAX_RATE_PLAYBACK 48000 +#define UX500_PLATFORM_MIN_RATE_CAPTURE 8000 +#define UX500_PLATFORM_MAX_RATE_CAPTURE 48000 + +#define UX500_PLATFORM_MIN_CHANNELS 1 +#define UX500_PLATFORM_MAX_CHANNELS 8 + +#define UX500_PLATFORM_PERIODS_BYTES_MIN 128 +#define UX500_PLATFORM_PERIODS_BYTES_MAX (64 * PAGE_SIZE) +#define UX500_PLATFORM_PERIODS_MIN 2 +#define UX500_PLATFORM_PERIODS_MAX 48 +#define UX500_PLATFORM_BUFFER_BYTES_MAX (2048 * PAGE_SIZE) + +#endif diff --git a/sound/sound_core.c b/sound/sound_core.c index c6e81fb928e9..fb9255cca214 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -361,7 +361,7 @@ int register_sound_special_device(const struct file_operations *fops, int unit, struct device *dev) { const int chain = unit % SOUND_STEP; - int max_unit = 128 + chain; + int max_unit = 256; const char *name; char _name[16]; diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index 6f9715ab32fe..56ad923bf6b5 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -209,7 +209,7 @@ static int usb6fire_fw_ezusb_upload( int ret; u8 data; struct usb_device *device = interface_to_usbdev(intf); - const struct firmware *fw = 0; + const struct firmware *fw = NULL; struct ihex_record *rec = kmalloc(sizeof(struct ihex_record), GFP_KERNEL); diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 64aed432ae22..7da0d0aa72cb 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -485,7 +485,7 @@ static int __devinit snd_probe(struct usb_interface *intf, const struct usb_device_id *id) { int ret; - struct snd_card *card; + struct snd_card *card = NULL; struct usb_device *device = interface_to_usbdev(intf); ret = create_card(device, intf, &card); diff --git a/sound/usb/card.c b/sound/usb/card.c index 4a7be7b98331..d5b5c3388e28 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -131,8 +131,9 @@ static void snd_usb_stream_disconnect(struct list_head *head) subs = &as->substream[idx]; if (!subs->num_formats) continue; - snd_usb_release_substream_urbs(subs, 1); subs->interface = -1; + subs->data_endpoint = NULL; + subs->sync_endpoint = NULL; } } @@ -276,6 +277,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) static int snd_usb_audio_free(struct snd_usb_audio *chip) { + mutex_destroy(&chip->mutex); kfree(chip); return 0; } @@ -336,6 +338,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, return -ENOMEM; } + mutex_init(&chip->mutex); mutex_init(&chip->shutdown_mutex); chip->index = idx; chip->dev = dev; @@ -348,6 +351,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, chip->usb_id = USB_ID(le16_to_cpu(dev->descriptor.idVendor), le16_to_cpu(dev->descriptor.idProduct)); INIT_LIST_HEAD(&chip->pcm_list); + INIT_LIST_HEAD(&chip->ep_list); INIT_LIST_HEAD(&chip->midi_list); INIT_LIST_HEAD(&chip->mixer_list); @@ -565,6 +569,10 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, list_for_each(p, &chip->pcm_list) { snd_usb_stream_disconnect(p); } + /* release the endpoint resources */ + list_for_each(p, &chip->ep_list) { + snd_usb_endpoint_free(p); + } /* release the midi resources */ list_for_each(p, &chip->midi_list) { snd_usbmidi_disconnect(p); diff --git a/sound/usb/card.h b/sound/usb/card.h index da5fa1ac4eda..2b9fffff23b6 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -30,20 +30,71 @@ struct audioformat { }; struct snd_usb_substream; +struct snd_usb_endpoint; struct snd_urb_ctx { struct urb *urb; unsigned int buffer_size; /* size of data buffer, if data URB */ struct snd_usb_substream *subs; + struct snd_usb_endpoint *ep; int index; /* index for urb array */ int packets; /* number of packets per urb */ + int packet_size[MAX_PACKS_HS]; /* size of packets for next submission */ + struct list_head ready_list; }; -struct snd_urb_ops { - int (*prepare)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); - int (*retire)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); - int (*prepare_sync)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); - int (*retire_sync)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); +struct snd_usb_endpoint { + struct snd_usb_audio *chip; + + int use_count; + int ep_num; /* the referenced endpoint number */ + int type; /* SND_USB_ENDPOINT_TYPE_* */ + unsigned long flags; + + void (*prepare_data_urb) (struct snd_usb_substream *subs, + struct urb *urb); + void (*retire_data_urb) (struct snd_usb_substream *subs, + struct urb *urb); + + struct snd_usb_substream *data_subs; + struct snd_usb_endpoint *sync_master; + struct snd_usb_endpoint *sync_slave; + + struct snd_urb_ctx urb[MAX_URBS]; + + struct snd_usb_packet_info { + uint32_t packet_size[MAX_PACKS_HS]; + int packets; + } next_packet[MAX_URBS]; + int next_packet_read_pos, next_packet_write_pos; + struct list_head ready_playback_urbs; + + unsigned int nurbs; /* # urbs */ + unsigned long active_mask; /* bitmask of active urbs */ + unsigned long unlink_mask; /* bitmask of unlinked urbs */ + char *syncbuf; /* sync buffer for all sync URBs */ + dma_addr_t sync_dma; /* DMA address of syncbuf */ + + unsigned int pipe; /* the data i/o pipe */ + unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ + unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ + int freqshift; /* how much to shift the feedback value to get Q16.16 */ + unsigned int freqmax; /* maximum sampling rate, used for buffer management */ + unsigned int phase; /* phase accumulator */ + unsigned int maxpacksize; /* max packet size in bytes */ + unsigned int maxframesize; /* max packet size in frames */ + unsigned int curpacksize; /* current packet size in bytes (for capture) */ + unsigned int curframesize; /* current packet size in frames (for capture) */ + unsigned int syncmaxsize; /* sync endpoint packet size */ + unsigned int fill_max:1; /* fill max packet size always */ + unsigned int datainterval; /* log_2 of data packet interval */ + unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */ + unsigned char silence_value; + unsigned int stride; + int iface, alt_idx; + + spinlock_t lock; + struct list_head list; }; struct snd_usb_substream { @@ -57,21 +108,6 @@ struct snd_usb_substream { unsigned int cur_rate; /* current rate (for hw_params callback) */ unsigned int period_bytes; /* current period bytes (for hw_params callback) */ unsigned int altset_idx; /* USB data format: index of alternate setting */ - unsigned int datapipe; /* the data i/o pipe */ - unsigned int syncpipe; /* 1 - async out or adaptive in */ - unsigned int datainterval; /* log_2 of data packet interval */ - unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */ - unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ - unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ - int freqshift; /* how much to shift the feedback value to get Q16.16 */ - unsigned int freqmax; /* maximum sampling rate, used for buffer management */ - unsigned int phase; /* phase accumulator */ - unsigned int maxpacksize; /* max packet size in bytes */ - unsigned int maxframesize; /* max packet size in frames */ - unsigned int curpacksize; /* current packet size in bytes (for capture) */ - unsigned int curframesize; /* current packet size in frames (for capture) */ - unsigned int syncmaxsize; /* sync endpoint packet size */ - unsigned int fill_max: 1; /* fill max packet size always */ unsigned int txfr_quirk:1; /* allow sub-frame alignment */ unsigned int fmt_type; /* USB audio format type (1-3) */ @@ -82,11 +118,11 @@ struct snd_usb_substream { unsigned long active_mask; /* bitmask of active urbs */ unsigned long unlink_mask; /* bitmask of unlinked urbs */ - unsigned int nurbs; /* # urbs */ - struct snd_urb_ctx dataurb[MAX_URBS]; /* data urb table */ - struct snd_urb_ctx syncurb[SYNC_URBS]; /* sync urb table */ - char *syncbuf; /* sync buffer for all sync URBs */ - dma_addr_t sync_dma; /* DMA address of syncbuf */ + /* data and sync endpoints for this stream */ + unsigned int ep_num; /* the endpoint number */ + struct snd_usb_endpoint *data_endpoint; + struct snd_usb_endpoint *sync_endpoint; + unsigned long flags; u64 formats; /* format bitmasks (all or'ed) */ unsigned int num_formats; /* number of supported audio formats (list) */ @@ -94,7 +130,6 @@ struct snd_usb_substream { struct snd_pcm_hw_constraint_list rate_list; /* limited rates */ spinlock_t lock; - struct snd_urb_ops ops; /* callbacks (must be filled at init) */ int last_frame_number; /* stored frame number */ int last_delay; /* stored delay */ }; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 08dcce53720b..0f647d22cb4a 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -20,9 +20,11 @@ #include <linux/ratelimit.h> #include <linux/usb.h> #include <linux/usb/audio.h> +#include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> +#include <sound/pcm_params.h> #include "usbaudio.h" #include "helper.h" @@ -30,6 +32,36 @@ #include "endpoint.h" #include "pcm.h" +#define EP_FLAG_ACTIVATED 0 +#define EP_FLAG_RUNNING 1 + +/* + * snd_usb_endpoint is a model that abstracts everything related to an + * USB endpoint and its streaming. + * + * There are functions to activate and deactivate the streaming URBs and + * optional callbacks to let the pcm logic handle the actual content of the + * packets for playback and record. Thus, the bus streaming and the audio + * handlers are fully decoupled. + * + * There are two different types of endpoints in audio applications. + * + * SND_USB_ENDPOINT_TYPE_DATA handles full audio data payload for both + * inbound and outbound traffic. + * + * SND_USB_ENDPOINT_TYPE_SYNC endpoints are for inbound traffic only and + * expect the payload to carry Q10.14 / Q16.16 formatted sync information + * (3 or 4 bytes). + * + * Each endpoint has to be configured prior to being used by calling + * snd_usb_endpoint_set_params(). + * + * The model incorporates a reference counting, so that multiple users + * can call snd_usb_endpoint_start() and snd_usb_endpoint_stop(), and + * only the first user will effectively start the URBs, and only the last + * one to stop it will tear the URBs down again. + */ + /* * convert a sampling rate into our full speed format (fs/1000 in Q16.16) * this will overflow at approx 524 kHz @@ -49,71 +81,405 @@ static inline unsigned get_usb_high_speed_rate(unsigned int rate) } /* - * unlink active urbs. + * release a urb data */ -static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sleep) +static void release_urb_ctx(struct snd_urb_ctx *u) { - struct snd_usb_audio *chip = subs->stream->chip; - unsigned int i; - int async; + if (u->buffer_size) + usb_free_coherent(u->ep->chip->dev, u->buffer_size, + u->urb->transfer_buffer, + u->urb->transfer_dma); + usb_free_urb(u->urb); + u->urb = NULL; +} + +static const char *usb_error_string(int err) +{ + switch (err) { + case -ENODEV: + return "no device"; + case -ENOENT: + return "endpoint not enabled"; + case -EPIPE: + return "endpoint stalled"; + case -ENOSPC: + return "not enough bandwidth"; + case -ESHUTDOWN: + return "device disabled"; + case -EHOSTUNREACH: + return "device suspended"; + case -EINVAL: + case -EAGAIN: + case -EFBIG: + case -EMSGSIZE: + return "internal error"; + default: + return "unknown error"; + } +} - subs->running = 0; +/** + * snd_usb_endpoint_implicit_feedback_sink: Report endpoint usage type + * + * @ep: The snd_usb_endpoint + * + * Determine whether an endpoint is driven by an implicit feedback + * data endpoint source. + */ +int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep) +{ + return ep->sync_master && + ep->sync_master->type == SND_USB_ENDPOINT_TYPE_DATA && + ep->type == SND_USB_ENDPOINT_TYPE_DATA && + usb_pipeout(ep->pipe); +} - if (!force && subs->stream->chip->shutdown) /* to be sure... */ - return -EBADFD; +/* + * For streaming based on information derived from sync endpoints, + * prepare_outbound_urb_sizes() will call next_packet_size() to + * determine the number of samples to be sent in the next packet. + * + * For implicit feedback, next_packet_size() is unused. + */ +static int next_packet_size(struct snd_usb_endpoint *ep) +{ + unsigned long flags; + int ret; - async = !can_sleep && chip->async_unlink; + if (ep->fill_max) + return ep->maxframesize; - if (!async && in_interrupt()) - return 0; + spin_lock_irqsave(&ep->lock, flags); + ep->phase = (ep->phase & 0xffff) + + (ep->freqm << ep->datainterval); + ret = min(ep->phase >> 16, ep->maxframesize); + spin_unlock_irqrestore(&ep->lock, flags); - for (i = 0; i < subs->nurbs; i++) { - if (test_bit(i, &subs->active_mask)) { - if (!test_and_set_bit(i, &subs->unlink_mask)) { - struct urb *u = subs->dataurb[i].urb; - if (async) - usb_unlink_urb(u); - else - usb_kill_urb(u); + return ret; +} + +static void retire_outbound_urb(struct snd_usb_endpoint *ep, + struct snd_urb_ctx *urb_ctx) +{ + if (ep->retire_data_urb) + ep->retire_data_urb(ep->data_subs, urb_ctx->urb); +} + +static void retire_inbound_urb(struct snd_usb_endpoint *ep, + struct snd_urb_ctx *urb_ctx) +{ + struct urb *urb = urb_ctx->urb; + + if (ep->sync_slave) + snd_usb_handle_sync_urb(ep->sync_slave, ep, urb); + + if (ep->retire_data_urb) + ep->retire_data_urb(ep->data_subs, urb); +} + +static void prepare_outbound_urb_sizes(struct snd_usb_endpoint *ep, + struct snd_urb_ctx *ctx) +{ + int i; + + for (i = 0; i < ctx->packets; ++i) + ctx->packet_size[i] = next_packet_size(ep); +} + +/* + * Prepare a PLAYBACK urb for submission to the bus. + */ +static void prepare_outbound_urb(struct snd_usb_endpoint *ep, + struct snd_urb_ctx *ctx) +{ + int i; + struct urb *urb = ctx->urb; + unsigned char *cp = urb->transfer_buffer; + + urb->dev = ep->chip->dev; /* we need to set this at each time */ + + switch (ep->type) { + case SND_USB_ENDPOINT_TYPE_DATA: + if (ep->prepare_data_urb) { + ep->prepare_data_urb(ep->data_subs, urb); + } else { + /* no data provider, so send silence */ + unsigned int offs = 0; + for (i = 0; i < ctx->packets; ++i) { + int counts = ctx->packet_size[i]; + urb->iso_frame_desc[i].offset = offs * ep->stride; + urb->iso_frame_desc[i].length = counts * ep->stride; + offs += counts; } + + urb->number_of_packets = ctx->packets; + urb->transfer_buffer_length = offs * ep->stride; + memset(urb->transfer_buffer, ep->silence_value, + offs * ep->stride); + } + break; + + case SND_USB_ENDPOINT_TYPE_SYNC: + if (snd_usb_get_speed(ep->chip->dev) >= USB_SPEED_HIGH) { + /* + * fill the length and offset of each urb descriptor. + * the fixed 12.13 frequency is passed as 16.16 through the pipe. + */ + urb->iso_frame_desc[0].length = 4; + urb->iso_frame_desc[0].offset = 0; + cp[0] = ep->freqn; + cp[1] = ep->freqn >> 8; + cp[2] = ep->freqn >> 16; + cp[3] = ep->freqn >> 24; + } else { + /* + * fill the length and offset of each urb descriptor. + * the fixed 10.14 frequency is passed through the pipe. + */ + urb->iso_frame_desc[0].length = 3; + urb->iso_frame_desc[0].offset = 0; + cp[0] = ep->freqn >> 2; + cp[1] = ep->freqn >> 10; + cp[2] = ep->freqn >> 18; } + + break; } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - if (test_bit(i+16, &subs->active_mask)) { - if (!test_and_set_bit(i+16, &subs->unlink_mask)) { - struct urb *u = subs->syncurb[i].urb; - if (async) - usb_unlink_urb(u); - else - usb_kill_urb(u); - } - } +} + +/* + * Prepare a CAPTURE or SYNC urb for submission to the bus. + */ +static inline void prepare_inbound_urb(struct snd_usb_endpoint *ep, + struct snd_urb_ctx *urb_ctx) +{ + int i, offs; + struct urb *urb = urb_ctx->urb; + + urb->dev = ep->chip->dev; /* we need to set this at each time */ + + switch (ep->type) { + case SND_USB_ENDPOINT_TYPE_DATA: + offs = 0; + for (i = 0; i < urb_ctx->packets; i++) { + urb->iso_frame_desc[i].offset = offs; + urb->iso_frame_desc[i].length = ep->curpacksize; + offs += ep->curpacksize; } + + urb->transfer_buffer_length = offs; + urb->number_of_packets = urb_ctx->packets; + break; + + case SND_USB_ENDPOINT_TYPE_SYNC: + urb->iso_frame_desc[0].length = min(4u, ep->syncmaxsize); + urb->iso_frame_desc[0].offset = 0; + break; } - return 0; } +/* + * Send output urbs that have been prepared previously. URBs are dequeued + * from ep->ready_playback_urbs and in case there there aren't any available + * or there are no packets that have been prepared, this function does + * nothing. + * + * The reason why the functionality of sending and preparing URBs is separated + * is that host controllers don't guarantee the order in which they return + * inbound and outbound packets to their submitters. + * + * This function is only used for implicit feedback endpoints. For endpoints + * driven by dedicated sync endpoints, URBs are immediately re-submitted + * from their completion handler. + */ +static void queue_pending_output_urbs(struct snd_usb_endpoint *ep) +{ + while (test_bit(EP_FLAG_RUNNING, &ep->flags)) { + + unsigned long flags; + struct snd_usb_packet_info *uninitialized_var(packet); + struct snd_urb_ctx *ctx = NULL; + struct urb *urb; + int err, i; + + spin_lock_irqsave(&ep->lock, flags); + if (ep->next_packet_read_pos != ep->next_packet_write_pos) { + packet = ep->next_packet + ep->next_packet_read_pos; + ep->next_packet_read_pos++; + ep->next_packet_read_pos %= MAX_URBS; + + /* take URB out of FIFO */ + if (!list_empty(&ep->ready_playback_urbs)) + ctx = list_first_entry(&ep->ready_playback_urbs, + struct snd_urb_ctx, ready_list); + } + spin_unlock_irqrestore(&ep->lock, flags); + + if (ctx == NULL) + return; + + list_del_init(&ctx->ready_list); + urb = ctx->urb; + + /* copy over the length information */ + for (i = 0; i < packet->packets; i++) + ctx->packet_size[i] = packet->packet_size[i]; + + /* call the data handler to fill in playback data */ + prepare_outbound_urb(ep, ctx); + + err = usb_submit_urb(ctx->urb, GFP_ATOMIC); + if (err < 0) + snd_printk(KERN_ERR "Unable to submit urb #%d: %d (urb %p)\n", + ctx->index, err, ctx->urb); + else + set_bit(ctx->index, &ep->active_mask); + } +} /* - * release a urb data + * complete callback for urbs */ -static void release_urb_ctx(struct snd_urb_ctx *u) +static void snd_complete_urb(struct urb *urb) { - if (u->urb) { - if (u->buffer_size) - usb_free_coherent(u->subs->dev, u->buffer_size, - u->urb->transfer_buffer, - u->urb->transfer_dma); - usb_free_urb(u->urb); - u->urb = NULL; + struct snd_urb_ctx *ctx = urb->context; + struct snd_usb_endpoint *ep = ctx->ep; + int err; + + if (unlikely(urb->status == -ENOENT || /* unlinked */ + urb->status == -ENODEV || /* device removed */ + urb->status == -ECONNRESET || /* unlinked */ + urb->status == -ESHUTDOWN || /* device disabled */ + ep->chip->shutdown)) /* device disconnected */ + goto exit_clear; + + if (usb_pipeout(ep->pipe)) { + retire_outbound_urb(ep, ctx); + /* can be stopped during retire callback */ + if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags))) + goto exit_clear; + + if (snd_usb_endpoint_implict_feedback_sink(ep)) { + unsigned long flags; + + spin_lock_irqsave(&ep->lock, flags); + list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs); + spin_unlock_irqrestore(&ep->lock, flags); + queue_pending_output_urbs(ep); + + goto exit_clear; + } + + prepare_outbound_urb_sizes(ep, ctx); + prepare_outbound_urb(ep, ctx); + } else { + retire_inbound_urb(ep, ctx); + /* can be stopped during retire callback */ + if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags))) + goto exit_clear; + + prepare_inbound_urb(ep, ctx); } + + err = usb_submit_urb(urb, GFP_ATOMIC); + if (err == 0) + return; + + snd_printk(KERN_ERR "cannot submit urb (err = %d)\n", err); + //snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + +exit_clear: + clear_bit(ctx->index, &ep->active_mask); +} + +/** + * snd_usb_add_endpoint: Add an endpoint to an USB audio chip + * + * @chip: The chip + * @alts: The USB host interface + * @ep_num: The number of the endpoint to use + * @direction: SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE + * @type: SND_USB_ENDPOINT_TYPE_DATA or SND_USB_ENDPOINT_TYPE_SYNC + * + * If the requested endpoint has not been added to the given chip before, + * a new instance is created. Otherwise, a pointer to the previoulsy + * created instance is returned. In case of any error, NULL is returned. + * + * New endpoints will be added to chip->ep_list and must be freed by + * calling snd_usb_endpoint_free(). + */ +struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, + struct usb_host_interface *alts, + int ep_num, int direction, int type) +{ + struct list_head *p; + struct snd_usb_endpoint *ep; + int is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK; + + mutex_lock(&chip->mutex); + + list_for_each(p, &chip->ep_list) { + ep = list_entry(p, struct snd_usb_endpoint, list); + if (ep->ep_num == ep_num && + ep->iface == alts->desc.bInterfaceNumber && + ep->alt_idx == alts->desc.bAlternateSetting) { + snd_printdd(KERN_DEBUG "Re-using EP %x in iface %d,%d @%p\n", + ep_num, ep->iface, ep->alt_idx, ep); + goto __exit_unlock; + } + } + + snd_printdd(KERN_DEBUG "Creating new %s %s endpoint #%x\n", + is_playback ? "playback" : "capture", + type == SND_USB_ENDPOINT_TYPE_DATA ? "data" : "sync", + ep_num); + + ep = kzalloc(sizeof(*ep), GFP_KERNEL); + if (!ep) + goto __exit_unlock; + + ep->chip = chip; + spin_lock_init(&ep->lock); + ep->type = type; + ep->ep_num = ep_num; + ep->iface = alts->desc.bInterfaceNumber; + ep->alt_idx = alts->desc.bAlternateSetting; + INIT_LIST_HEAD(&ep->ready_playback_urbs); + ep_num &= USB_ENDPOINT_NUMBER_MASK; + + if (is_playback) + ep->pipe = usb_sndisocpipe(chip->dev, ep_num); + else + ep->pipe = usb_rcvisocpipe(chip->dev, ep_num); + + if (type == SND_USB_ENDPOINT_TYPE_SYNC) { + if (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && + get_endpoint(alts, 1)->bRefresh >= 1 && + get_endpoint(alts, 1)->bRefresh <= 9) + ep->syncinterval = get_endpoint(alts, 1)->bRefresh; + else if (snd_usb_get_speed(chip->dev) == USB_SPEED_FULL) + ep->syncinterval = 1; + else if (get_endpoint(alts, 1)->bInterval >= 1 && + get_endpoint(alts, 1)->bInterval <= 16) + ep->syncinterval = get_endpoint(alts, 1)->bInterval - 1; + else + ep->syncinterval = 3; + + ep->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize); + } + + list_add_tail(&ep->list, &chip->ep_list); + +__exit_unlock: + mutex_unlock(&chip->mutex); + + return ep; } /* * wait until all urbs are processed. */ -static int wait_clear_urbs(struct snd_usb_substream *subs) +static int wait_clear_urbs(struct snd_usb_endpoint *ep) { unsigned long end_time = jiffies + msecs_to_jiffies(1000); unsigned int i; @@ -121,153 +487,148 @@ static int wait_clear_urbs(struct snd_usb_substream *subs) do { alive = 0; - for (i = 0; i < subs->nurbs; i++) { - if (test_bit(i, &subs->active_mask)) + for (i = 0; i < ep->nurbs; i++) + if (test_bit(i, &ep->active_mask)) alive++; - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - if (test_bit(i + 16, &subs->active_mask)) - alive++; - } - } - if (! alive) + + if (!alive) break; + schedule_timeout_uninterruptible(1); } while (time_before(jiffies, end_time)); + if (alive) - snd_printk(KERN_ERR "timeout: still %d active urbs..\n", alive); + snd_printk(KERN_ERR "timeout: still %d active urbs on EP #%x\n", + alive, ep->ep_num); + return 0; } /* - * release a substream + * unlink active urbs. */ -void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force) +static int deactivate_urbs(struct snd_usb_endpoint *ep, int force, int can_sleep) { - int i; + unsigned int i; + int async; - /* stop urbs (to be sure) */ - deactivate_urbs(subs, force, 1); - wait_clear_urbs(subs); - - for (i = 0; i < MAX_URBS; i++) - release_urb_ctx(&subs->dataurb[i]); - for (i = 0; i < SYNC_URBS; i++) - release_urb_ctx(&subs->syncurb[i]); - usb_free_coherent(subs->dev, SYNC_URBS * 4, - subs->syncbuf, subs->sync_dma); - subs->syncbuf = NULL; - subs->nurbs = 0; -} + if (!force && ep->chip->shutdown) /* to be sure... */ + return -EBADFD; -/* - * complete callback from data urb - */ -static void snd_complete_urb(struct urb *urb) -{ - struct snd_urb_ctx *ctx = urb->context; - struct snd_usb_substream *subs = ctx->subs; - struct snd_pcm_substream *substream = ctx->subs->pcm_substream; - int err = 0; - - if ((subs->running && subs->ops.retire(subs, substream->runtime, urb)) || - !subs->running || /* can be stopped during retire callback */ - (err = subs->ops.prepare(subs, substream->runtime, urb)) < 0 || - (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { - clear_bit(ctx->index, &subs->active_mask); - if (err < 0) { - snd_printd(KERN_ERR "cannot submit urb (err = %d)\n", err); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + async = !can_sleep && ep->chip->async_unlink; + + clear_bit(EP_FLAG_RUNNING, &ep->flags); + + INIT_LIST_HEAD(&ep->ready_playback_urbs); + ep->next_packet_read_pos = 0; + ep->next_packet_write_pos = 0; + + if (!async && in_interrupt()) + return 0; + + for (i = 0; i < ep->nurbs; i++) { + if (test_bit(i, &ep->active_mask)) { + if (!test_and_set_bit(i, &ep->unlink_mask)) { + struct urb *u = ep->urb[i].urb; + if (async) + usb_unlink_urb(u); + else + usb_kill_urb(u); + } } } -} + return 0; +} /* - * complete callback from sync urb + * release an endpoint's urbs */ -static void snd_complete_sync_urb(struct urb *urb) +static void release_urbs(struct snd_usb_endpoint *ep, int force) { - struct snd_urb_ctx *ctx = urb->context; - struct snd_usb_substream *subs = ctx->subs; - struct snd_pcm_substream *substream = ctx->subs->pcm_substream; - int err = 0; - - if ((subs->running && subs->ops.retire_sync(subs, substream->runtime, urb)) || - !subs->running || /* can be stopped during retire callback */ - (err = subs->ops.prepare_sync(subs, substream->runtime, urb)) < 0 || - (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { - clear_bit(ctx->index + 16, &subs->active_mask); - if (err < 0) { - snd_printd(KERN_ERR "cannot submit sync urb (err = %d)\n", err); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - } - } -} + int i; + + /* route incoming urbs to nirvana */ + ep->retire_data_urb = NULL; + ep->prepare_data_urb = NULL; + + /* stop urbs */ + deactivate_urbs(ep, force, 1); + wait_clear_urbs(ep); + + for (i = 0; i < ep->nurbs; i++) + release_urb_ctx(&ep->urb[i]); + if (ep->syncbuf) + usb_free_coherent(ep->chip->dev, SYNC_URBS * 4, + ep->syncbuf, ep->sync_dma); + + ep->syncbuf = NULL; + ep->nurbs = 0; +} /* - * initialize a substream for plaback/capture + * configure a data endpoint */ -int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, - unsigned int period_bytes, - unsigned int rate, - unsigned int frame_bits) +static int data_ep_set_params(struct snd_usb_endpoint *ep, + struct snd_pcm_hw_params *hw_params, + struct audioformat *fmt, + struct snd_usb_endpoint *sync_ep) { - unsigned int maxsize, i; - int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; - unsigned int urb_packs, total_packs, packs_per_ms; - struct snd_usb_audio *chip = subs->stream->chip; + unsigned int maxsize, i, urb_packs, total_packs, packs_per_ms; + int period_bytes = params_period_bytes(hw_params); + int format = params_format(hw_params); + int is_playback = usb_pipeout(ep->pipe); + int frame_bits = snd_pcm_format_physical_width(params_format(hw_params)) * + params_channels(hw_params); + + ep->datainterval = fmt->datainterval; + ep->stride = frame_bits >> 3; + ep->silence_value = format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0; - /* calculate the frequency in 16.16 format */ - if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) - subs->freqn = get_usb_full_speed_rate(rate); - else - subs->freqn = get_usb_high_speed_rate(rate); - subs->freqm = subs->freqn; - subs->freqshift = INT_MIN; /* calculate max. frequency */ - if (subs->maxpacksize) { + if (ep->maxpacksize) { /* whatever fits into a max. size packet */ - maxsize = subs->maxpacksize; - subs->freqmax = (maxsize / (frame_bits >> 3)) - << (16 - subs->datainterval); + maxsize = ep->maxpacksize; + ep->freqmax = (maxsize / (frame_bits >> 3)) + << (16 - ep->datainterval); } else { /* no max. packet size: just take 25% higher than nominal */ - subs->freqmax = subs->freqn + (subs->freqn >> 2); - maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3)) - >> (16 - subs->datainterval); + ep->freqmax = ep->freqn + (ep->freqn >> 2); + maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) + >> (16 - ep->datainterval); } - subs->phase = 0; - if (subs->fill_max) - subs->curpacksize = subs->maxpacksize; + if (ep->fill_max) + ep->curpacksize = ep->maxpacksize; else - subs->curpacksize = maxsize; + ep->curpacksize = maxsize; - if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL) - packs_per_ms = 8 >> subs->datainterval; + if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) + packs_per_ms = 8 >> ep->datainterval; else packs_per_ms = 1; - if (is_playback) { - urb_packs = max(chip->nrpacks, 1); - urb_packs = min(urb_packs, (unsigned int)MAX_PACKS); - } else + if (is_playback && !snd_usb_endpoint_implict_feedback_sink(ep)) { + urb_packs = max(ep->chip->nrpacks, 1); + urb_packs = min(urb_packs, (unsigned int) MAX_PACKS); + } else { urb_packs = 1; + } + urb_packs *= packs_per_ms; - if (subs->syncpipe) - urb_packs = min(urb_packs, 1U << subs->syncinterval); + + if (sync_ep && !snd_usb_endpoint_implict_feedback_sink(ep)) + urb_packs = min(urb_packs, 1U << sync_ep->syncinterval); /* decide how many packets to be used */ - if (is_playback) { + if (is_playback && !snd_usb_endpoint_implict_feedback_sink(ep)) { unsigned int minsize, maxpacks; /* determine how small a packet can be */ - minsize = (subs->freqn >> (16 - subs->datainterval)) + minsize = (ep->freqn >> (16 - ep->datainterval)) * (frame_bits >> 3); /* with sync from device, assume it can be 12% lower */ - if (subs->syncpipe) + if (sync_ep) minsize -= minsize >> 3; minsize = max(minsize, 1u); total_packs = (period_bytes + minsize - 1) / minsize; @@ -284,284 +645,421 @@ int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, urb_packs >>= 1; total_packs = MAX_URBS * urb_packs; } - subs->nurbs = (total_packs + urb_packs - 1) / urb_packs; - if (subs->nurbs > MAX_URBS) { + + ep->nurbs = (total_packs + urb_packs - 1) / urb_packs; + if (ep->nurbs > MAX_URBS) { /* too much... */ - subs->nurbs = MAX_URBS; + ep->nurbs = MAX_URBS; total_packs = MAX_URBS * urb_packs; - } else if (subs->nurbs < 2) { + } else if (ep->nurbs < 2) { /* too little - we need at least two packets * to ensure contiguous playback/capture */ - subs->nurbs = 2; + ep->nurbs = 2; } /* allocate and initialize data urbs */ - for (i = 0; i < subs->nurbs; i++) { - struct snd_urb_ctx *u = &subs->dataurb[i]; + for (i = 0; i < ep->nurbs; i++) { + struct snd_urb_ctx *u = &ep->urb[i]; u->index = i; - u->subs = subs; - u->packets = (i + 1) * total_packs / subs->nurbs - - i * total_packs / subs->nurbs; + u->ep = ep; + u->packets = (i + 1) * total_packs / ep->nurbs + - i * total_packs / ep->nurbs; u->buffer_size = maxsize * u->packets; - if (subs->fmt_type == UAC_FORMAT_TYPE_II) + + if (fmt->fmt_type == UAC_FORMAT_TYPE_II) u->packets++; /* for transfer delimiter */ u->urb = usb_alloc_urb(u->packets, GFP_KERNEL); if (!u->urb) goto out_of_memory; + u->urb->transfer_buffer = - usb_alloc_coherent(subs->dev, u->buffer_size, + usb_alloc_coherent(ep->chip->dev, u->buffer_size, GFP_KERNEL, &u->urb->transfer_dma); if (!u->urb->transfer_buffer) goto out_of_memory; - u->urb->pipe = subs->datapipe; + u->urb->pipe = ep->pipe; u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP; - u->urb->interval = 1 << subs->datainterval; + u->urb->interval = 1 << ep->datainterval; u->urb->context = u; u->urb->complete = snd_complete_urb; + INIT_LIST_HEAD(&u->ready_list); } - if (subs->syncpipe) { - /* allocate and initialize sync urbs */ - subs->syncbuf = usb_alloc_coherent(subs->dev, SYNC_URBS * 4, - GFP_KERNEL, &subs->sync_dma); - if (!subs->syncbuf) - goto out_of_memory; - for (i = 0; i < SYNC_URBS; i++) { - struct snd_urb_ctx *u = &subs->syncurb[i]; - u->index = i; - u->subs = subs; - u->packets = 1; - u->urb = usb_alloc_urb(1, GFP_KERNEL); - if (!u->urb) - goto out_of_memory; - u->urb->transfer_buffer = subs->syncbuf + i * 4; - u->urb->transfer_dma = subs->sync_dma + i * 4; - u->urb->transfer_buffer_length = 4; - u->urb->pipe = subs->syncpipe; - u->urb->transfer_flags = URB_ISO_ASAP | - URB_NO_TRANSFER_DMA_MAP; - u->urb->number_of_packets = 1; - u->urb->interval = 1 << subs->syncinterval; - u->urb->context = u; - u->urb->complete = snd_complete_sync_urb; - } - } return 0; out_of_memory: - snd_usb_release_substream_urbs(subs, 0); + release_urbs(ep, 0); return -ENOMEM; } /* - * prepare urb for full speed capture sync pipe - * - * fill the length and offset of each urb descriptor. - * the fixed 10.14 frequency is passed through the pipe. + * configure a sync endpoint */ -static int prepare_capture_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) +static int sync_ep_set_params(struct snd_usb_endpoint *ep, + struct snd_pcm_hw_params *hw_params, + struct audioformat *fmt) { - unsigned char *cp = urb->transfer_buffer; - struct snd_urb_ctx *ctx = urb->context; + int i; + + ep->syncbuf = usb_alloc_coherent(ep->chip->dev, SYNC_URBS * 4, + GFP_KERNEL, &ep->sync_dma); + if (!ep->syncbuf) + return -ENOMEM; + + for (i = 0; i < SYNC_URBS; i++) { + struct snd_urb_ctx *u = &ep->urb[i]; + u->index = i; + u->ep = ep; + u->packets = 1; + u->urb = usb_alloc_urb(1, GFP_KERNEL); + if (!u->urb) + goto out_of_memory; + u->urb->transfer_buffer = ep->syncbuf + i * 4; + u->urb->transfer_dma = ep->sync_dma + i * 4; + u->urb->transfer_buffer_length = 4; + u->urb->pipe = ep->pipe; + u->urb->transfer_flags = URB_ISO_ASAP | + URB_NO_TRANSFER_DMA_MAP; + u->urb->number_of_packets = 1; + u->urb->interval = 1 << ep->syncinterval; + u->urb->context = u; + u->urb->complete = snd_complete_urb; + } + + ep->nurbs = SYNC_URBS; - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 3; - urb->iso_frame_desc[0].offset = 0; - cp[0] = subs->freqn >> 2; - cp[1] = subs->freqn >> 10; - cp[2] = subs->freqn >> 18; return 0; + +out_of_memory: + release_urbs(ep, 0); + return -ENOMEM; } -/* - * prepare urb for high speed capture sync pipe +/** + * snd_usb_endpoint_set_params: configure an snd_usb_endpoint + * + * @ep: the snd_usb_endpoint to configure + * @hw_params: the hardware parameters + * @fmt: the USB audio format information + * @sync_ep: the sync endpoint to use, if any * - * fill the length and offset of each urb descriptor. - * the fixed 12.13 frequency is passed as 16.16 through the pipe. + * Determine the number of URBs to be used on this endpoint. + * An endpoint must be configured before it can be started. + * An endpoint that is already running can not be reconfigured. */ -static int prepare_capture_sync_urb_hs(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) +int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, + struct snd_pcm_hw_params *hw_params, + struct audioformat *fmt, + struct snd_usb_endpoint *sync_ep) { - unsigned char *cp = urb->transfer_buffer; - struct snd_urb_ctx *ctx = urb->context; + int err; - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 4; - urb->iso_frame_desc[0].offset = 0; - cp[0] = subs->freqn; - cp[1] = subs->freqn >> 8; - cp[2] = subs->freqn >> 16; - cp[3] = subs->freqn >> 24; - return 0; -} + if (ep->use_count != 0) { + snd_printk(KERN_WARNING "Unable to change format on ep #%x: already in use\n", + ep->ep_num); + return -EBUSY; + } -/* - * process after capture sync complete - * - nothing to do - */ -static int retire_capture_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - return 0; + /* release old buffers, if any */ + release_urbs(ep, 0); + + ep->datainterval = fmt->datainterval; + ep->maxpacksize = fmt->maxpacksize; + ep->fill_max = !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX); + + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) + ep->freqn = get_usb_full_speed_rate(params_rate(hw_params)); + else + ep->freqn = get_usb_high_speed_rate(params_rate(hw_params)); + + /* calculate the frequency in 16.16 format */ + ep->freqm = ep->freqn; + ep->freqshift = INT_MIN; + + ep->phase = 0; + + switch (ep->type) { + case SND_USB_ENDPOINT_TYPE_DATA: + err = data_ep_set_params(ep, hw_params, fmt, sync_ep); + break; + case SND_USB_ENDPOINT_TYPE_SYNC: + err = sync_ep_set_params(ep, hw_params, fmt); + break; + default: + err = -EINVAL; + } + + snd_printdd(KERN_DEBUG "Setting params for ep #%x (type %d, %d urbs), ret=%d\n", + ep->ep_num, ep->type, ep->nurbs, err); + + return err; } -/* - * prepare urb for capture data pipe +/** + * snd_usb_endpoint_start: start an snd_usb_endpoint * - * fill the offset and length of each descriptor. + * @ep: the endpoint to start * - * we use a temporary buffer to write the captured data. - * since the length of written data is determined by host, we cannot - * write onto the pcm buffer directly... the data is thus copied - * later at complete callback to the global buffer. + * A call to this function will increment the use count of the endpoint. + * In case it is not already running, the URBs for this endpoint will be + * submitted. Otherwise, this function does nothing. + * + * Must be balanced to calls of snd_usb_endpoint_stop(). + * + * Returns an error if the URB submission failed, 0 in all other cases. */ -static int prepare_capture_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) { - int i, offs; - struct snd_urb_ctx *ctx = urb->context; + int err; + unsigned int i; + + if (ep->chip->shutdown) + return -EBADFD; + + /* already running? */ + if (++ep->use_count != 1) + return 0; + + /* just to be sure */ + deactivate_urbs(ep, 0, 1); + wait_clear_urbs(ep); + + ep->active_mask = 0; + ep->unlink_mask = 0; + ep->phase = 0; + + /* + * If this endpoint has a data endpoint as implicit feedback source, + * don't start the urbs here. Instead, mark them all as available, + * wait for the record urbs to return and queue the playback urbs + * from that context. + */ + + set_bit(EP_FLAG_RUNNING, &ep->flags); + + if (snd_usb_endpoint_implict_feedback_sink(ep)) { + for (i = 0; i < ep->nurbs; i++) { + struct snd_urb_ctx *ctx = ep->urb + i; + list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs); + } + + return 0; + } - offs = 0; - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - for (i = 0; i < ctx->packets; i++) { - urb->iso_frame_desc[i].offset = offs; - urb->iso_frame_desc[i].length = subs->curpacksize; - offs += subs->curpacksize; + for (i = 0; i < ep->nurbs; i++) { + struct urb *urb = ep->urb[i].urb; + + if (snd_BUG_ON(!urb)) + goto __error; + + if (usb_pipeout(ep->pipe)) { + prepare_outbound_urb_sizes(ep, urb->context); + prepare_outbound_urb(ep, urb->context); + } else { + prepare_inbound_urb(ep, urb->context); + } + + err = usb_submit_urb(urb, GFP_ATOMIC); + if (err < 0) { + snd_printk(KERN_ERR "cannot submit urb %d, error %d: %s\n", + i, err, usb_error_string(err)); + goto __error; + } + set_bit(i, &ep->active_mask); } - urb->transfer_buffer_length = offs; - urb->number_of_packets = ctx->packets; + return 0; + +__error: + clear_bit(EP_FLAG_RUNNING, &ep->flags); + ep->use_count--; + deactivate_urbs(ep, 0, 0); + return -EPIPE; } -/* - * process after capture complete +/** + * snd_usb_endpoint_stop: stop an snd_usb_endpoint + * + * @ep: the endpoint to stop (may be NULL) * - * copy the data from each desctiptor to the pcm buffer, and - * update the current position. + * A call to this function will decrement the use count of the endpoint. + * In case the last user has requested the endpoint stop, the URBs will + * actually be deactivated. + * + * Must be balanced to calls of snd_usb_endpoint_start(). */ -static int retire_capture_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) +void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, + int force, int can_sleep, int wait) { - unsigned long flags; - unsigned char *cp; - int i; - unsigned int stride, frames, bytes, oldptr; - int period_elapsed = 0; + if (!ep) + return; - stride = runtime->frame_bits >> 3; + if (snd_BUG_ON(ep->use_count == 0)) + return; - for (i = 0; i < urb->number_of_packets; i++) { - cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset; - if (urb->iso_frame_desc[i].status && printk_ratelimit()) { - snd_printdd("frame %d active: %d\n", i, urb->iso_frame_desc[i].status); - // continue; - } - bytes = urb->iso_frame_desc[i].actual_length; - frames = bytes / stride; - if (!subs->txfr_quirk) - bytes = frames * stride; - if (bytes % (runtime->sample_bits >> 3) != 0) { -#ifdef CONFIG_SND_DEBUG_VERBOSE - int oldbytes = bytes; -#endif - bytes = frames * stride; - snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n", - oldbytes, bytes); - } - /* update the current pointer */ - spin_lock_irqsave(&subs->lock, flags); - oldptr = subs->hwptr_done; - subs->hwptr_done += bytes; - if (subs->hwptr_done >= runtime->buffer_size * stride) - subs->hwptr_done -= runtime->buffer_size * stride; - frames = (bytes + (oldptr % stride)) / stride; - subs->transfer_done += frames; - if (subs->transfer_done >= runtime->period_size) { - subs->transfer_done -= runtime->period_size; - period_elapsed = 1; - } - spin_unlock_irqrestore(&subs->lock, flags); - /* copy a data chunk */ - if (oldptr + bytes > runtime->buffer_size * stride) { - unsigned int bytes1 = - runtime->buffer_size * stride - oldptr; - memcpy(runtime->dma_area + oldptr, cp, bytes1); - memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1); - } else { - memcpy(runtime->dma_area + oldptr, cp, bytes); - } + if (--ep->use_count == 0) { + deactivate_urbs(ep, force, can_sleep); + ep->data_subs = NULL; + ep->sync_slave = NULL; + ep->retire_data_urb = NULL; + ep->prepare_data_urb = NULL; + + if (wait) + wait_clear_urbs(ep); } - if (period_elapsed) - snd_pcm_period_elapsed(subs->pcm_substream); - return 0; } -/* - * Process after capture complete when paused. Nothing to do. +/** + * snd_usb_endpoint_deactivate: deactivate an snd_usb_endpoint + * + * @ep: the endpoint to deactivate + * + * If the endpoint is not currently in use, this functions will select the + * alternate interface setting 0 for the interface of this endpoint. + * + * In case of any active users, this functions does nothing. + * + * Returns an error if usb_set_interface() failed, 0 in all other + * cases. */ -static int retire_paused_capture_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) +int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep) { + if (!ep) + return -EINVAL; + + deactivate_urbs(ep, 1, 1); + wait_clear_urbs(ep); + + if (ep->use_count != 0) + return 0; + + clear_bit(EP_FLAG_ACTIVATED, &ep->flags); + return 0; } - -/* - * prepare urb for playback sync pipe +/** + * snd_usb_endpoint_free: Free the resources of an snd_usb_endpoint * - * set up the offset and length to receive the current frequency. + * @ep: the list header of the endpoint to free + * + * This function does not care for the endpoint's use count but will tear + * down all the streaming URBs immediately and free all resources. */ -static int prepare_playback_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) +void snd_usb_endpoint_free(struct list_head *head) { - struct snd_urb_ctx *ctx = urb->context; + struct snd_usb_endpoint *ep; - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = min(4u, ctx->subs->syncmaxsize); - urb->iso_frame_desc[0].offset = 0; - return 0; + ep = list_entry(head, struct snd_usb_endpoint, list); + release_urbs(ep, 1); + kfree(ep); } -/* - * process after playback sync complete - * - * Full speed devices report feedback values in 10.14 format as samples per - * frame, high speed devices in 16.16 format as samples per microframe. - * Because the Audio Class 1 spec was written before USB 2.0, many high speed - * devices use a wrong interpretation, some others use an entirely different - * format. Therefore, we cannot predict what format any particular device uses - * and must detect it automatically. +/** + * snd_usb_handle_sync_urb: parse an USB sync packet + * + * @ep: the endpoint to handle the packet + * @sender: the sending endpoint + * @urb: the received packet + * + * This function is called from the context of an endpoint that received + * the packet and is used to let another endpoint object handle the payload. */ -static int retire_playback_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) +void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, + struct snd_usb_endpoint *sender, + const struct urb *urb) { - unsigned int f; int shift; + unsigned int f; unsigned long flags; + snd_BUG_ON(ep == sender); + + /* + * In case the endpoint is operating in implicit feedback mode, prepare + * a new outbound URB that has the same layout as the received packet + * and add it to the list of pending urbs. queue_pending_output_urbs() + * will take care of them later. + */ + if (snd_usb_endpoint_implict_feedback_sink(ep) && + ep->use_count != 0) { + + /* implicit feedback case */ + int i, bytes = 0; + struct snd_urb_ctx *in_ctx; + struct snd_usb_packet_info *out_packet; + + in_ctx = urb->context; + + /* Count overall packet size */ + for (i = 0; i < in_ctx->packets; i++) + if (urb->iso_frame_desc[i].status == 0) + bytes += urb->iso_frame_desc[i].actual_length; + + /* + * skip empty packets. At least M-Audio's Fast Track Ultra stops + * streaming once it received a 0-byte OUT URB + */ + if (bytes == 0) + return; + + spin_lock_irqsave(&ep->lock, flags); + out_packet = ep->next_packet + ep->next_packet_write_pos; + + /* + * Iterate through the inbound packet and prepare the lengths + * for the output packet. The OUT packet we are about to send + * will have the same amount of payload bytes than the IN + * packet we just received. + */ + + out_packet->packets = in_ctx->packets; + for (i = 0; i < in_ctx->packets; i++) { + if (urb->iso_frame_desc[i].status == 0) + out_packet->packet_size[i] = + urb->iso_frame_desc[i].actual_length / ep->stride; + else + out_packet->packet_size[i] = 0; + } + + ep->next_packet_write_pos++; + ep->next_packet_write_pos %= MAX_URBS; + spin_unlock_irqrestore(&ep->lock, flags); + queue_pending_output_urbs(ep); + + return; + } + + /* + * process after playback sync complete + * + * Full speed devices report feedback values in 10.14 format as samples + * per frame, high speed devices in 16.16 format as samples per + * microframe. + * + * Because the Audio Class 1 spec was written before USB 2.0, many high + * speed devices use a wrong interpretation, some others use an + * entirely different format. + * + * Therefore, we cannot predict what format any particular device uses + * and must detect it automatically. + */ + if (urb->iso_frame_desc[0].status != 0 || urb->iso_frame_desc[0].actual_length < 3) - return 0; + return; f = le32_to_cpup(urb->transfer_buffer); if (urb->iso_frame_desc[0].actual_length == 3) f &= 0x00ffffff; else f &= 0x0fffffff; + if (f == 0) - return 0; + return; - if (unlikely(subs->freqshift == INT_MIN)) { + if (unlikely(ep->freqshift == INT_MIN)) { /* * The first time we see a feedback value, determine its format * by shifting it left or right until it matches the nominal @@ -569,398 +1067,34 @@ static int retire_playback_sync_urb(struct snd_usb_substream *subs, * differ from the nominal value more than +50% or -25%. */ shift = 0; - while (f < subs->freqn - subs->freqn / 4) { + while (f < ep->freqn - ep->freqn / 4) { f <<= 1; shift++; } - while (f > subs->freqn + subs->freqn / 2) { + while (f > ep->freqn + ep->freqn / 2) { f >>= 1; shift--; } - subs->freqshift = shift; - } - else if (subs->freqshift >= 0) - f <<= subs->freqshift; + ep->freqshift = shift; + } else if (ep->freqshift >= 0) + f <<= ep->freqshift; else - f >>= -subs->freqshift; + f >>= -ep->freqshift; - if (likely(f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax)) { + if (likely(f >= ep->freqn - ep->freqn / 8 && f <= ep->freqmax)) { /* * If the frequency looks valid, set it. * This value is referred to in prepare_playback_urb(). */ - spin_lock_irqsave(&subs->lock, flags); - subs->freqm = f; - spin_unlock_irqrestore(&subs->lock, flags); + spin_lock_irqsave(&ep->lock, flags); + ep->freqm = f; + spin_unlock_irqrestore(&ep->lock, flags); } else { /* * Out of range; maybe the shift value is wrong. * Reset it so that we autodetect again the next time. */ - subs->freqshift = INT_MIN; - } - - return 0; -} - -/* determine the number of frames in the next packet */ -static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs) -{ - if (subs->fill_max) - return subs->maxframesize; - else { - subs->phase = (subs->phase & 0xffff) - + (subs->freqm << subs->datainterval); - return min(subs->phase >> 16, subs->maxframesize); - } -} - -/* - * Prepare urb for streaming before playback starts or when paused. - * - * We don't have any data, so we send silence. - */ -static int prepare_nodata_playback_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned int i, offs, counts; - struct snd_urb_ctx *ctx = urb->context; - int stride = runtime->frame_bits >> 3; - - offs = 0; - urb->dev = ctx->subs->dev; - for (i = 0; i < ctx->packets; ++i) { - counts = snd_usb_audio_next_packet_size(subs); - urb->iso_frame_desc[i].offset = offs * stride; - urb->iso_frame_desc[i].length = counts * stride; - offs += counts; + ep->freqshift = INT_MIN; } - urb->number_of_packets = ctx->packets; - urb->transfer_buffer_length = offs * stride; - memset(urb->transfer_buffer, - runtime->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0, - offs * stride); - return 0; -} - -/* - * prepare urb for playback data pipe - * - * Since a URB can handle only a single linear buffer, we must use double - * buffering when the data to be transferred overflows the buffer boundary. - * To avoid inconsistencies when updating hwptr_done, we use double buffering - * for all URBs. - */ -static int prepare_playback_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - int i, stride; - unsigned int counts, frames, bytes; - unsigned long flags; - int period_elapsed = 0; - struct snd_urb_ctx *ctx = urb->context; - - stride = runtime->frame_bits >> 3; - - frames = 0; - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->number_of_packets = 0; - spin_lock_irqsave(&subs->lock, flags); - for (i = 0; i < ctx->packets; i++) { - counts = snd_usb_audio_next_packet_size(subs); - /* set up descriptor */ - urb->iso_frame_desc[i].offset = frames * stride; - urb->iso_frame_desc[i].length = counts * stride; - frames += counts; - urb->number_of_packets++; - subs->transfer_done += counts; - if (subs->transfer_done >= runtime->period_size) { - subs->transfer_done -= runtime->period_size; - period_elapsed = 1; - if (subs->fmt_type == UAC_FORMAT_TYPE_II) { - if (subs->transfer_done > 0) { - /* FIXME: fill-max mode is not - * supported yet */ - frames -= subs->transfer_done; - counts -= subs->transfer_done; - urb->iso_frame_desc[i].length = - counts * stride; - subs->transfer_done = 0; - } - i++; - if (i < ctx->packets) { - /* add a transfer delimiter */ - urb->iso_frame_desc[i].offset = - frames * stride; - urb->iso_frame_desc[i].length = 0; - urb->number_of_packets++; - } - break; - } - } - if (period_elapsed) /* finish at the period boundary */ - break; - } - bytes = frames * stride; - if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { - /* err, the transferred area goes over buffer boundary. */ - unsigned int bytes1 = - runtime->buffer_size * stride - subs->hwptr_done; - memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done, bytes1); - memcpy(urb->transfer_buffer + bytes1, - runtime->dma_area, bytes - bytes1); - } else { - memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done, bytes); - } - subs->hwptr_done += bytes; - if (subs->hwptr_done >= runtime->buffer_size * stride) - subs->hwptr_done -= runtime->buffer_size * stride; - - /* update delay with exact number of samples queued */ - runtime->delay = subs->last_delay; - runtime->delay += frames; - subs->last_delay = runtime->delay; - - /* realign last_frame_number */ - subs->last_frame_number = usb_get_current_frame_number(subs->dev); - subs->last_frame_number &= 0xFF; /* keep 8 LSBs */ - - spin_unlock_irqrestore(&subs->lock, flags); - urb->transfer_buffer_length = bytes; - if (period_elapsed) - snd_pcm_period_elapsed(subs->pcm_substream); - return 0; -} - -/* - * process after playback data complete - * - decrease the delay count again - */ -static int retire_playback_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned long flags; - int stride = runtime->frame_bits >> 3; - int processed = urb->transfer_buffer_length / stride; - int est_delay; - - spin_lock_irqsave(&subs->lock, flags); - - est_delay = snd_usb_pcm_delay(subs, runtime->rate); - /* update delay with exact number of samples played */ - if (processed > subs->last_delay) - subs->last_delay = 0; - else - subs->last_delay -= processed; - runtime->delay = subs->last_delay; - - /* - * Report when delay estimate is off by more than 2ms. - * The error should be lower than 2ms since the estimate relies - * on two reads of a counter updated every ms. - */ - if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2) - snd_printk(KERN_DEBUG "delay: estimated %d, actual %d\n", - est_delay, subs->last_delay); - - spin_unlock_irqrestore(&subs->lock, flags); - return 0; -} - -static const char *usb_error_string(int err) -{ - switch (err) { - case -ENODEV: - return "no device"; - case -ENOENT: - return "endpoint not enabled"; - case -EPIPE: - return "endpoint stalled"; - case -ENOSPC: - return "not enough bandwidth"; - case -ESHUTDOWN: - return "device disabled"; - case -EHOSTUNREACH: - return "device suspended"; - case -EINVAL: - case -EAGAIN: - case -EFBIG: - case -EMSGSIZE: - return "internal error"; - default: - return "unknown error"; - } -} - -/* - * set up and start data/sync urbs - */ -static int start_urbs(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime) -{ - unsigned int i; - int err; - - if (subs->stream->chip->shutdown) - return -EBADFD; - - for (i = 0; i < subs->nurbs; i++) { - if (snd_BUG_ON(!subs->dataurb[i].urb)) - return -EINVAL; - if (subs->ops.prepare(subs, runtime, subs->dataurb[i].urb) < 0) { - snd_printk(KERN_ERR "cannot prepare datapipe for urb %d\n", i); - goto __error; - } - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - if (snd_BUG_ON(!subs->syncurb[i].urb)) - return -EINVAL; - if (subs->ops.prepare_sync(subs, runtime, subs->syncurb[i].urb) < 0) { - snd_printk(KERN_ERR "cannot prepare syncpipe for urb %d\n", i); - goto __error; - } - } - } - - subs->active_mask = 0; - subs->unlink_mask = 0; - subs->running = 1; - for (i = 0; i < subs->nurbs; i++) { - err = usb_submit_urb(subs->dataurb[i].urb, GFP_ATOMIC); - if (err < 0) { - snd_printk(KERN_ERR "cannot submit datapipe " - "for urb %d, error %d: %s\n", - i, err, usb_error_string(err)); - goto __error; - } - set_bit(i, &subs->active_mask); - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - err = usb_submit_urb(subs->syncurb[i].urb, GFP_ATOMIC); - if (err < 0) { - snd_printk(KERN_ERR "cannot submit syncpipe " - "for urb %d, error %d: %s\n", - i, err, usb_error_string(err)); - goto __error; - } - set_bit(i + 16, &subs->active_mask); - } - } - return 0; - - __error: - // snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); - deactivate_urbs(subs, 0, 0); - return -EPIPE; -} - - -/* - */ -static struct snd_urb_ops audio_urb_ops[2] = { - { - .prepare = prepare_nodata_playback_urb, - .retire = retire_playback_urb, - .prepare_sync = prepare_playback_sync_urb, - .retire_sync = retire_playback_sync_urb, - }, - { - .prepare = prepare_capture_urb, - .retire = retire_capture_urb, - .prepare_sync = prepare_capture_sync_urb, - .retire_sync = retire_capture_sync_urb, - }, -}; - -/* - * initialize the substream instance. - */ - -void snd_usb_init_substream(struct snd_usb_stream *as, - int stream, struct audioformat *fp) -{ - struct snd_usb_substream *subs = &as->substream[stream]; - - INIT_LIST_HEAD(&subs->fmt_list); - spin_lock_init(&subs->lock); - - subs->stream = as; - subs->direction = stream; - subs->dev = as->chip->dev; - subs->txfr_quirk = as->chip->txfr_quirk; - subs->ops = audio_urb_ops[stream]; - if (snd_usb_get_speed(subs->dev) >= USB_SPEED_HIGH) - subs->ops.prepare_sync = prepare_capture_sync_urb_hs; - - snd_usb_set_pcm_ops(as->pcm, stream); - - list_add_tail(&fp->list, &subs->fmt_list); - subs->formats |= fp->formats; - subs->endpoint = fp->endpoint; - subs->num_formats++; - subs->fmt_type = fp->fmt_type; -} - -int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_usb_substream *subs = substream->runtime->private_data; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - subs->ops.prepare = prepare_playback_urb; - return 0; - case SNDRV_PCM_TRIGGER_STOP: - return deactivate_urbs(subs, 0, 0); - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - subs->ops.prepare = prepare_nodata_playback_urb; - return 0; - } - - return -EINVAL; -} - -int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_usb_substream *subs = substream->runtime->private_data; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - subs->ops.retire = retire_capture_urb; - return start_urbs(subs, substream->runtime); - case SNDRV_PCM_TRIGGER_STOP: - return deactivate_urbs(subs, 0, 0); - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - subs->ops.retire = retire_paused_capture_urb; - return 0; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - subs->ops.retire = retire_capture_urb; - return 0; - } - - return -EINVAL; -} - -int snd_usb_substream_prepare(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime) -{ - /* clear urbs (to be sure) */ - deactivate_urbs(subs, 0, 1); - wait_clear_urbs(subs); - - /* for playback, submit the URBs now; otherwise, the first hwptr_done - * updates for all URBs would happen at the same time when starting */ - if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) { - subs->ops.prepare = prepare_nodata_playback_urb; - return start_urbs(subs, runtime); - } - - return 0; } diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 88eb63a636eb..ee2723fb174f 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -1,21 +1,29 @@ #ifndef __USBAUDIO_ENDPOINT_H #define __USBAUDIO_ENDPOINT_H -void snd_usb_init_substream(struct snd_usb_stream *as, - int stream, - struct audioformat *fp); +#define SND_USB_ENDPOINT_TYPE_DATA 0 +#define SND_USB_ENDPOINT_TYPE_SYNC 1 -int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, - unsigned int period_bytes, - unsigned int rate, - unsigned int frame_bits); +struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, + struct usb_host_interface *alts, + int ep_num, int direction, int type); -void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force); +int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, + struct snd_pcm_hw_params *hw_params, + struct audioformat *fmt, + struct snd_usb_endpoint *sync_ep); -int snd_usb_substream_prepare(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime); +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep); +void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, + int force, int can_sleep, int wait); +int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep); +void snd_usb_endpoint_free(struct list_head *head); -int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd); -int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd); +int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep); + +void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, + struct snd_usb_endpoint *sender, + const struct urb *urb); #endif /* __USBAUDIO_ENDPOINT_H */ diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ab23869c01bb..4f40ba823163 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -486,7 +486,7 @@ static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, /* * TLV callback for mixer volume controls */ -static int mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, +int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *_tlv) { struct usb_mixer_elem_info *cval = kcontrol->private_data; @@ -770,6 +770,26 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, struct snd_kcontrol *kctl) { switch (cval->mixer->chip->usb_id) { + case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ + case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra */ + if (strcmp(kctl->id.name, "Effect Duration") == 0) { + snd_printk(KERN_INFO + "usb-audio: set quirk for FTU Effect Duration\n"); + cval->min = 0x0000; + cval->max = 0x7f00; + cval->res = 0x0100; + break; + } + if (strcmp(kctl->id.name, "Effect Volume") == 0 || + strcmp(kctl->id.name, "Effect Feedback Volume") == 0) { + snd_printk(KERN_INFO + "usb-audio: set quirks for FTU Effect Feedback/Volume\n"); + cval->min = 0x00; + cval->max = 0x7f; + break; + } + break; + case USB_ID(0x0471, 0x0101): case USB_ID(0x0471, 0x0104): case USB_ID(0x0471, 0x0105): @@ -1121,9 +1141,6 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, len = snd_usb_copy_string_desc(state, nameid, kctl->id.name, sizeof(kctl->id.name)); - /* get min/max values */ - get_min_max_with_quirks(cval, 0, kctl); - switch (control) { case UAC_FU_MUTE: case UAC_FU_VOLUME: @@ -1155,17 +1172,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, } append_ctl_name(kctl, control == UAC_FU_MUTE ? " Switch" : " Volume"); - if (control == UAC_FU_VOLUME) { - check_mapped_dB(map, cval); - if (cval->dBmin < cval->dBmax || !cval->initialized) { - kctl->tlv.c = mixer_vol_tlv; - kctl->vd[0].access |= - SNDRV_CTL_ELEM_ACCESS_TLV_READ | - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; - } - } break; - default: if (! len) strlcpy(kctl->id.name, audio_feature_info[control-1].name, @@ -1173,6 +1180,19 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, break; } + /* get min/max values */ + get_min_max_with_quirks(cval, 0, kctl); + + if (control == UAC_FU_VOLUME) { + check_mapped_dB(map, cval); + if (cval->dBmin < cval->dBmax || !cval->initialized) { + kctl->tlv.c = snd_usb_mixer_vol_tlv; + kctl->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + } + } + range = (cval->max - cval->min) / cval->res; /* Are there devices with volume range more than 255? I use a bit more * to be sure. 384 is a resolution magic number found on Logitech @@ -1388,7 +1408,7 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, void *r for (pin = 0; pin < input_pins; pin++) { err = parse_audio_unit(state, desc->baSourceID[pin]); if (err < 0) - return err; + continue; err = check_input_term(state, desc->baSourceID[pin], &iterm); if (err < 0) return err; diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 81b2d8a32fb0..a7f3d45a8acf 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -68,4 +68,7 @@ int snd_usb_mixer_activate(struct usb_mixer_interface *mixer); int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer, struct snd_kcontrol *kctl); +int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *_tlv); + #endif /* __USBMIXER_H */ diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index f1324c423835..e71fe55cebef 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -288,6 +288,15 @@ static struct usbmix_name_map scratch_live_map[] = { { 0 } /* terminator */ }; +static struct usbmix_name_map ebox44_map[] = { + { 4, NULL }, /* FU */ + { 6, NULL }, /* MU */ + { 7, NULL }, /* FU */ + { 10, NULL }, /* FU */ + { 11, NULL }, /* MU */ + { 0 } +}; + /* "Gamesurround Muse Pocket LT" looks same like "Sound Blaster MP3+" * most importand difference is SU[8], it should be set to "Capture Source" * to make alsamixer and PA working properly. @@ -332,6 +341,14 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = audigy2nx_map, .selector_map = audigy2nx_selectors, }, + { /* Logitech, Inc. QuickCam Pro for Notebooks */ + .id = USB_ID(0x046d, 0x0991), + .ignore_ctl_error = 1, + }, + { /* Logitech, Inc. QuickCam E 3500 */ + .id = USB_ID(0x046d, 0x09a4), + .ignore_ctl_error = 1, + }, { /* Hercules DJ Console (Windows Edition) */ .id = USB_ID(0x06f8, 0xb000), @@ -371,6 +388,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = scratch_live_map, .ignore_ctl_error = 1, }, + { + .id = USB_ID(0x200c, 0x1018), + .map = ebox44_map, + }, { 0 } /* terminator */ }; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index ab125ee0b0f0..690000db0ec0 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -42,6 +42,103 @@ extern struct snd_kcontrol_new *snd_usb_feature_unit_ctl; +struct std_mono_table { + unsigned int unitid, control, cmask; + int val_type; + const char *name; + snd_kcontrol_tlv_rw_t *tlv_callback; +}; + +/* private_free callback */ +static void usb_mixer_elem_free(struct snd_kcontrol *kctl) +{ + kfree(kctl->private_data); + kctl->private_data = NULL; +} + +/* This function allows for the creation of standard UAC controls. + * See the quirks for M-Audio FTUs or Ebox-44. + * If you don't want to set a TLV callback pass NULL. + * + * Since there doesn't seem to be a devices that needs a multichannel + * version, we keep it mono for simplicity. + */ +static int snd_create_std_mono_ctl(struct usb_mixer_interface *mixer, + unsigned int unitid, + unsigned int control, + unsigned int cmask, + int val_type, + const char *name, + snd_kcontrol_tlv_rw_t *tlv_callback) +{ + int err; + struct usb_mixer_elem_info *cval; + struct snd_kcontrol *kctl; + + cval = kzalloc(sizeof(*cval), GFP_KERNEL); + if (!cval) + return -ENOMEM; + + cval->id = unitid; + cval->mixer = mixer; + cval->val_type = val_type; + cval->channels = 1; + cval->control = control; + cval->cmask = cmask; + + /* get_min_max() is called only for integer volumes later, + * so provide a short-cut for booleans */ + cval->min = 0; + cval->max = 1; + cval->res = 0; + cval->dBmin = 0; + cval->dBmax = 0; + + /* Create control */ + kctl = snd_ctl_new1(snd_usb_feature_unit_ctl, cval); + if (!kctl) { + kfree(cval); + return -ENOMEM; + } + + /* Set name */ + snprintf(kctl->id.name, sizeof(kctl->id.name), name); + kctl->private_free = usb_mixer_elem_free; + + /* set TLV */ + if (tlv_callback) { + kctl->tlv.c = tlv_callback; + kctl->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + } + /* Add control to mixer */ + err = snd_usb_mixer_add_control(mixer, kctl); + if (err < 0) + return err; + + return 0; +} + +/* + * Create a set of standard UAC controls from a table + */ +static int snd_create_std_mono_table(struct usb_mixer_interface *mixer, + struct std_mono_table *t) +{ + int err; + + while (t->name != NULL) { + err = snd_create_std_mono_ctl(mixer, t->unitid, t->control, + t->cmask, t->val_type, t->name, t->tlv_callback); + if (err < 0) + return err; + t++; + } + + return 0; +} + /* * Sound Blaster remote control configuration * @@ -495,60 +592,218 @@ static int snd_nativeinstruments_create_mixer(struct usb_mixer_interface *mixer, } /* M-Audio FastTrack Ultra quirks */ +/* FTU Effect switch */ +struct snd_ftu_eff_switch_priv_val { + struct usb_mixer_interface *mixer; + int cached_value; + int is_cached; +}; -/* private_free callback */ -static void usb_mixer_elem_free(struct snd_kcontrol *kctl) +static int snd_ftu_eff_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { - kfree(kctl->private_data); - kctl->private_data = NULL; + static const char *texts[8] = {"Room 1", + "Room 2", + "Room 3", + "Hall 1", + "Hall 2", + "Plate", + "Delay", + "Echo" + }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 8; + if (uinfo->value.enumerated.item > 7) + uinfo->value.enumerated.item = 7; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; } -static int snd_maudio_ftu_create_ctl(struct usb_mixer_interface *mixer, - int in, int out, const char *name) +static int snd_ftu_eff_switch_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) { - struct usb_mixer_elem_info *cval; + struct snd_usb_audio *chip; + struct usb_mixer_interface *mixer; + struct snd_ftu_eff_switch_priv_val *pval; + int err; + unsigned char value[2]; + + const int id = 6; + const int validx = 1; + const int val_len = 2; + + value[0] = 0x00; + value[1] = 0x00; + + pval = (struct snd_ftu_eff_switch_priv_val *) + kctl->private_value; + + if (pval->is_cached) { + ucontrol->value.enumerated.item[0] = pval->cached_value; + return 0; + } + + mixer = (struct usb_mixer_interface *) pval->mixer; + if (snd_BUG_ON(!mixer)) + return -EINVAL; + + chip = (struct snd_usb_audio *) mixer->chip; + if (snd_BUG_ON(!chip)) + return -EINVAL; + + + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), UAC_GET_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + validx << 8, snd_usb_ctrl_intf(chip) | (id << 8), + value, val_len); + if (err < 0) + return err; + + ucontrol->value.enumerated.item[0] = value[0]; + pval->cached_value = value[0]; + pval->is_cached = 1; + + return 0; +} + +static int snd_ftu_eff_switch_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_usb_audio *chip; + struct snd_ftu_eff_switch_priv_val *pval; + + struct usb_mixer_interface *mixer; + int changed, cur_val, err, new_val; + unsigned char value[2]; + + + const int id = 6; + const int validx = 1; + const int val_len = 2; + + changed = 0; + + pval = (struct snd_ftu_eff_switch_priv_val *) + kctl->private_value; + cur_val = pval->cached_value; + new_val = ucontrol->value.enumerated.item[0]; + + mixer = (struct usb_mixer_interface *) pval->mixer; + if (snd_BUG_ON(!mixer)) + return -EINVAL; + + chip = (struct snd_usb_audio *) mixer->chip; + if (snd_BUG_ON(!chip)) + return -EINVAL; + + if (!pval->is_cached) { + /* Read current value */ + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), UAC_GET_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + validx << 8, snd_usb_ctrl_intf(chip) | (id << 8), + value, val_len); + if (err < 0) + return err; + + cur_val = value[0]; + pval->cached_value = cur_val; + pval->is_cached = 1; + } + /* update value if needed */ + if (cur_val != new_val) { + value[0] = new_val; + value[1] = 0; + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), UAC_SET_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + validx << 8, snd_usb_ctrl_intf(chip) | (id << 8), + value, val_len); + if (err < 0) + return err; + + pval->cached_value = new_val; + pval->is_cached = 1; + changed = 1; + } + + return changed; +} + +static int snd_ftu_create_effect_switch(struct usb_mixer_interface *mixer) +{ + static struct snd_kcontrol_new template = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Effect Program Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ftu_eff_switch_info, + .get = snd_ftu_eff_switch_get, + .put = snd_ftu_eff_switch_put + }; + + int err; struct snd_kcontrol *kctl; + struct snd_ftu_eff_switch_priv_val *pval; - cval = kzalloc(sizeof(*cval), GFP_KERNEL); - if (!cval) + pval = kzalloc(sizeof(*pval), GFP_KERNEL); + if (!pval) return -ENOMEM; - cval->id = 5; - cval->mixer = mixer; - cval->val_type = USB_MIXER_S16; - cval->channels = 1; - cval->control = out + 1; - cval->cmask = 1 << in; + pval->cached_value = 0; + pval->is_cached = 0; + pval->mixer = mixer; - kctl = snd_ctl_new1(snd_usb_feature_unit_ctl, cval); + template.private_value = (unsigned long) pval; + kctl = snd_ctl_new1(&template, mixer->chip); if (!kctl) { - kfree(cval); + kfree(pval); return -ENOMEM; } - snprintf(kctl->id.name, sizeof(kctl->id.name), name); - kctl->private_free = usb_mixer_elem_free; - return snd_usb_mixer_add_control(mixer, kctl); + err = snd_ctl_add(mixer->chip->card, kctl); + if (err < 0) + return err; + + return 0; } -static int snd_maudio_ftu_create_mixer(struct usb_mixer_interface *mixer) +/* Create volume controls for FTU devices*/ +static int snd_ftu_create_volume_ctls(struct usb_mixer_interface *mixer) { char name[64]; + unsigned int control, cmask; int in, out, err; + const unsigned int id = 5; + const int val_type = USB_MIXER_S16; + for (out = 0; out < 8; out++) { + control = out + 1; for (in = 0; in < 8; in++) { + cmask = 1 << in; snprintf(name, sizeof(name), - "AIn%d - Out%d Capture Volume", in + 1, out + 1); - err = snd_maudio_ftu_create_ctl(mixer, in, out, name); + "AIn%d - Out%d Capture Volume", + in + 1, out + 1); + err = snd_create_std_mono_ctl(mixer, id, control, + cmask, val_type, name, + &snd_usb_mixer_vol_tlv); if (err < 0) return err; } - for (in = 8; in < 16; in++) { + cmask = 1 << in; snprintf(name, sizeof(name), - "DIn%d - Out%d Playback Volume", in - 7, out + 1); - err = snd_maudio_ftu_create_ctl(mixer, in, out, name); + "DIn%d - Out%d Playback Volume", + in - 7, out + 1); + err = snd_create_std_mono_ctl(mixer, id, control, + cmask, val_type, name, + &snd_usb_mixer_vol_tlv); if (err < 0) return err; } @@ -557,6 +812,136 @@ static int snd_maudio_ftu_create_mixer(struct usb_mixer_interface *mixer) return 0; } +/* This control needs a volume quirk, see mixer.c */ +static int snd_ftu_create_effect_volume_ctl(struct usb_mixer_interface *mixer) +{ + static const char name[] = "Effect Volume"; + const unsigned int id = 6; + const int val_type = USB_MIXER_U8; + const unsigned int control = 2; + const unsigned int cmask = 0; + + return snd_create_std_mono_ctl(mixer, id, control, cmask, val_type, + name, snd_usb_mixer_vol_tlv); +} + +/* This control needs a volume quirk, see mixer.c */ +static int snd_ftu_create_effect_duration_ctl(struct usb_mixer_interface *mixer) +{ + static const char name[] = "Effect Duration"; + const unsigned int id = 6; + const int val_type = USB_MIXER_S16; + const unsigned int control = 3; + const unsigned int cmask = 0; + + return snd_create_std_mono_ctl(mixer, id, control, cmask, val_type, + name, snd_usb_mixer_vol_tlv); +} + +/* This control needs a volume quirk, see mixer.c */ +static int snd_ftu_create_effect_feedback_ctl(struct usb_mixer_interface *mixer) +{ + static const char name[] = "Effect Feedback Volume"; + const unsigned int id = 6; + const int val_type = USB_MIXER_U8; + const unsigned int control = 4; + const unsigned int cmask = 0; + + return snd_create_std_mono_ctl(mixer, id, control, cmask, val_type, + name, NULL); +} + +static int snd_ftu_create_effect_return_ctls(struct usb_mixer_interface *mixer) +{ + unsigned int cmask; + int err, ch; + char name[48]; + + const unsigned int id = 7; + const int val_type = USB_MIXER_S16; + const unsigned int control = 7; + + for (ch = 0; ch < 4; ++ch) { + cmask = 1 << ch; + snprintf(name, sizeof(name), + "Effect Return %d Volume", ch + 1); + err = snd_create_std_mono_ctl(mixer, id, control, + cmask, val_type, name, + snd_usb_mixer_vol_tlv); + if (err < 0) + return err; + } + + return 0; +} + +static int snd_ftu_create_effect_send_ctls(struct usb_mixer_interface *mixer) +{ + unsigned int cmask; + int err, ch; + char name[48]; + + const unsigned int id = 5; + const int val_type = USB_MIXER_S16; + const unsigned int control = 9; + + for (ch = 0; ch < 8; ++ch) { + cmask = 1 << ch; + snprintf(name, sizeof(name), + "Effect Send AIn%d Volume", ch + 1); + err = snd_create_std_mono_ctl(mixer, id, control, cmask, + val_type, name, + snd_usb_mixer_vol_tlv); + if (err < 0) + return err; + } + for (ch = 8; ch < 16; ++ch) { + cmask = 1 << ch; + snprintf(name, sizeof(name), + "Effect Send DIn%d Volume", ch - 7); + err = snd_create_std_mono_ctl(mixer, id, control, cmask, + val_type, name, + snd_usb_mixer_vol_tlv); + if (err < 0) + return err; + } + return 0; +} + +static int snd_ftu_create_mixer(struct usb_mixer_interface *mixer) +{ + int err; + + err = snd_ftu_create_volume_ctls(mixer); + if (err < 0) + return err; + + err = snd_ftu_create_effect_switch(mixer); + if (err < 0) + return err; + err = snd_ftu_create_effect_volume_ctl(mixer); + if (err < 0) + return err; + + err = snd_ftu_create_effect_duration_ctl(mixer); + if (err < 0) + return err; + + err = snd_ftu_create_effect_feedback_ctl(mixer); + if (err < 0) + return err; + + err = snd_ftu_create_effect_return_ctls(mixer); + if (err < 0) + return err; + + err = snd_ftu_create_effect_send_ctls(mixer); + if (err < 0) + return err; + + return 0; +} + void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, unsigned char samplerate_id) { @@ -576,6 +961,81 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, } } +/* + * The mixer units for Ebox-44 are corrupt, and even where they + * are valid they presents mono controls as L and R channels of + * stereo. So we provide a good mixer here. + */ +struct std_mono_table ebox44_table[] = { + { + .unitid = 4, + .control = 1, + .cmask = 0x0, + .val_type = USB_MIXER_INV_BOOLEAN, + .name = "Headphone Playback Switch" + }, + { + .unitid = 4, + .control = 2, + .cmask = 0x1, + .val_type = USB_MIXER_S16, + .name = "Headphone A Mix Playback Volume" + }, + { + .unitid = 4, + .control = 2, + .cmask = 0x2, + .val_type = USB_MIXER_S16, + .name = "Headphone B Mix Playback Volume" + }, + + { + .unitid = 7, + .control = 1, + .cmask = 0x0, + .val_type = USB_MIXER_INV_BOOLEAN, + .name = "Output Playback Switch" + }, + { + .unitid = 7, + .control = 2, + .cmask = 0x1, + .val_type = USB_MIXER_S16, + .name = "Output A Playback Volume" + }, + { + .unitid = 7, + .control = 2, + .cmask = 0x2, + .val_type = USB_MIXER_S16, + .name = "Output B Playback Volume" + }, + + { + .unitid = 10, + .control = 1, + .cmask = 0x0, + .val_type = USB_MIXER_INV_BOOLEAN, + .name = "Input Capture Switch" + }, + { + .unitid = 10, + .control = 2, + .cmask = 0x1, + .val_type = USB_MIXER_S16, + .name = "Input A Capture Volume" + }, + { + .unitid = 10, + .control = 2, + .cmask = 0x2, + .val_type = USB_MIXER_S16, + .name = "Input B Capture Volume" + }, + + {} +}; + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -600,7 +1060,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra */ case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ - err = snd_maudio_ftu_create_mixer(mixer); + err = snd_ftu_create_mixer(mixer); break; case USB_ID(0x0b05, 0x1739): @@ -619,6 +1079,11 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) snd_nativeinstruments_ta10_mixers, ARRAY_SIZE(snd_nativeinstruments_ta10_mixers)); break; + + case USB_ID(0x200c, 0x1018): /* Electrix Ebox-44 */ + /* detection is disabled in mixer_maps.c */ + err = snd_create_std_mono_table(mixer, ebox44_table); + break; } return err; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 0eed6115c2d4..a1298f379428 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -16,6 +16,7 @@ #include <linux/init.h> #include <linux/slab.h> +#include <linux/ratelimit.h> #include <linux/usb.h> #include <linux/usb/audio.h> #include <linux/usb/audio-v2.h> @@ -34,6 +35,9 @@ #include "clock.h" #include "power.h" +#define SUBSTREAM_FLAG_DATA_EP_STARTED 0 +#define SUBSTREAM_FLAG_SYNC_EP_STARTED 1 + /* return the estimated delay based on USB frame counters */ snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, unsigned int rate) @@ -208,6 +212,71 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, } } +static int start_endpoints(struct snd_usb_substream *subs) +{ + int err; + + if (!subs->data_endpoint) + return -EINVAL; + + if (!test_and_set_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags)) { + struct snd_usb_endpoint *ep = subs->data_endpoint; + + snd_printdd(KERN_DEBUG "Starting data EP @%p\n", ep); + + ep->data_subs = subs; + err = snd_usb_endpoint_start(ep); + if (err < 0) { + clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags); + return err; + } + } + + if (subs->sync_endpoint && + !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) { + struct snd_usb_endpoint *ep = subs->sync_endpoint; + + snd_printdd(KERN_DEBUG "Starting sync EP @%p\n", ep); + + ep->sync_slave = subs->data_endpoint; + err = snd_usb_endpoint_start(ep); + if (err < 0) { + clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); + return err; + } + } + + return 0; +} + +static void stop_endpoints(struct snd_usb_substream *subs, + int force, int can_sleep, int wait) +{ + if (test_and_clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) + snd_usb_endpoint_stop(subs->sync_endpoint, + force, can_sleep, wait); + + if (test_and_clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags)) + snd_usb_endpoint_stop(subs->data_endpoint, + force, can_sleep, wait); +} + +static int deactivate_endpoints(struct snd_usb_substream *subs) +{ + int reta, retb; + + reta = snd_usb_endpoint_deactivate(subs->sync_endpoint); + retb = snd_usb_endpoint_deactivate(subs->data_endpoint); + + if (reta < 0) + return reta; + + if (retb < 0) + return retb; + + return 0; +} + /* * find a matching format and set up the interface */ @@ -219,7 +288,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) struct usb_interface *iface; unsigned int ep, attr; int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; - int err; + int err, implicit_fb = 0; iface = usb_ifnum_to_if(dev, fmt->iface); if (WARN_ON(!iface)) @@ -234,9 +303,10 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) /* close the old interface */ if (subs->interface >= 0 && subs->interface != fmt->iface) { - if (usb_set_interface(subs->dev, subs->interface, 0) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: return to setting 0 failed\n", - dev->devnum, fmt->iface, fmt->altsetting); + err = usb_set_interface(subs->dev, subs->interface, 0); + if (err < 0) { + snd_printk(KERN_ERR "%d:%d:%d: return to setting 0 failed (%d)\n", + dev->devnum, fmt->iface, fmt->altsetting, err); return -EIO; } subs->interface = -1; @@ -244,28 +314,25 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) } /* set interface */ - if (subs->interface != fmt->iface || subs->altset_idx != fmt->altset_idx) { - if (usb_set_interface(dev, fmt->iface, fmt->altsetting) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: usb_set_interface failed\n", - dev->devnum, fmt->iface, fmt->altsetting); + if (subs->interface != fmt->iface || + subs->altset_idx != fmt->altset_idx) { + err = usb_set_interface(dev, fmt->iface, fmt->altsetting); + if (err < 0) { + snd_printk(KERN_ERR "%d:%d:%d: usb_set_interface failed (%d)\n", + dev->devnum, fmt->iface, fmt->altsetting, err); return -EIO; } - snd_printdd(KERN_INFO "setting usb interface %d:%d\n", fmt->iface, fmt->altsetting); + snd_printdd(KERN_INFO "setting usb interface %d:%d\n", + fmt->iface, fmt->altsetting); subs->interface = fmt->iface; subs->altset_idx = fmt->altset_idx; } - /* create a data pipe */ - ep = fmt->endpoint & USB_ENDPOINT_NUMBER_MASK; - if (is_playback) - subs->datapipe = usb_sndisocpipe(dev, ep); - else - subs->datapipe = usb_rcvisocpipe(dev, ep); - subs->datainterval = fmt->datainterval; - subs->syncpipe = subs->syncinterval = 0; - subs->maxpacksize = fmt->maxpacksize; - subs->syncmaxsize = 0; - subs->fill_max = 0; + subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip, + alts, fmt->endpoint, subs->direction, + SND_USB_ENDPOINT_TYPE_DATA); + if (!subs->data_endpoint) + return -EINVAL; /* we need a sync pipe in async OUT or adaptive IN mode */ /* check the number of EP, since some devices have broken @@ -273,8 +340,25 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) * assume it as adaptive-out or sync-in. */ attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE; + + switch (subs->stream->chip->usb_id) { + case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */ + case USB_ID(0x0763, 0x2081): + if (is_playback) { + implicit_fb = 1; + ep = 0x81; + iface = usb_ifnum_to_if(dev, 2); + + if (!iface || iface->num_altsetting == 0) + return -EINVAL; + + alts = &iface->altsetting[1]; + goto add_sync_ep; + } + } + if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) || - (! is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) && + (!is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) && altsd->bNumEndpoints >= 2) { /* check sync-pipe endpoint */ /* ... and check descriptor size before accessing bSynchAddress @@ -282,43 +366,42 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) the audio fields in the endpoint descriptors */ if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != 0x01 || (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && - get_endpoint(alts, 1)->bSynchAddress != 0)) { - snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", - dev->devnum, fmt->iface, fmt->altsetting); + get_endpoint(alts, 1)->bSynchAddress != 0 && + !implicit_fb)) { + snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n", + dev->devnum, fmt->iface, fmt->altsetting, + get_endpoint(alts, 1)->bmAttributes, + get_endpoint(alts, 1)->bLength, + get_endpoint(alts, 1)->bSynchAddress); return -EINVAL; } ep = get_endpoint(alts, 1)->bEndpointAddress; - if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && + if (!implicit_fb && + get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) || (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) { - snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", - dev->devnum, fmt->iface, fmt->altsetting); + snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n", + dev->devnum, fmt->iface, fmt->altsetting, + is_playback, ep, get_endpoint(alts, 0)->bSynchAddress); return -EINVAL; } - ep &= USB_ENDPOINT_NUMBER_MASK; - if (is_playback) - subs->syncpipe = usb_rcvisocpipe(dev, ep); - else - subs->syncpipe = usb_sndisocpipe(dev, ep); - if (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && - get_endpoint(alts, 1)->bRefresh >= 1 && - get_endpoint(alts, 1)->bRefresh <= 9) - subs->syncinterval = get_endpoint(alts, 1)->bRefresh; - else if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) - subs->syncinterval = 1; - else if (get_endpoint(alts, 1)->bInterval >= 1 && - get_endpoint(alts, 1)->bInterval <= 16) - subs->syncinterval = get_endpoint(alts, 1)->bInterval - 1; - else - subs->syncinterval = 3; - subs->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize); - } - - /* always fill max packet size */ - if (fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX) - subs->fill_max = 1; - - if ((err = snd_usb_init_pitch(subs->stream->chip, subs->interface, alts, fmt)) < 0) + + implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK) + == USB_ENDPOINT_USAGE_IMPLICIT_FB; + +add_sync_ep: + subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip, + alts, ep, !subs->direction, + implicit_fb ? + SND_USB_ENDPOINT_TYPE_DATA : + SND_USB_ENDPOINT_TYPE_SYNC); + if (!subs->sync_endpoint) + return -EINVAL; + + subs->data_endpoint->sync_master = subs->sync_endpoint; + } + + if ((err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt)) < 0) return err; subs->cur_audiofmt = fmt; @@ -381,7 +464,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, struct usb_interface *iface; iface = usb_ifnum_to_if(subs->dev, fmt->iface); alts = &iface->altsetting[fmt->altset_idx]; - ret = snd_usb_init_sample_rate(subs->stream->chip, subs->interface, alts, fmt, rate); + ret = snd_usb_init_sample_rate(subs->stream->chip, fmt->iface, alts, fmt, rate); if (ret < 0) return ret; subs->cur_rate = rate; @@ -390,15 +473,24 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, if (changed) { mutex_lock(&subs->stream->chip->shutdown_mutex); /* format changed */ - snd_usb_release_substream_urbs(subs, 0); - /* influenced: period_bytes, channels, rate, format, */ - ret = snd_usb_init_substream_urbs(subs, params_period_bytes(hw_params), - params_rate(hw_params), - snd_pcm_format_physical_width(params_format(hw_params)) * - params_channels(hw_params)); + stop_endpoints(subs, 0, 0, 0); + ret = snd_usb_endpoint_set_params(subs->data_endpoint, hw_params, fmt, + subs->sync_endpoint); + if (ret < 0) + goto unlock; + + if (subs->sync_endpoint) + ret = snd_usb_endpoint_set_params(subs->sync_endpoint, + hw_params, fmt, NULL); +unlock: mutex_unlock(&subs->stream->chip->shutdown_mutex); } + if (ret == 0) { + subs->interface = fmt->iface; + subs->altset_idx = fmt->altset_idx; + } + return ret; } @@ -415,7 +507,8 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) subs->cur_rate = 0; subs->period_bytes = 0; mutex_lock(&subs->stream->chip->shutdown_mutex); - snd_usb_release_substream_urbs(subs, 0); + stop_endpoints(subs, 0, 1, 1); + deactivate_endpoints(subs); mutex_unlock(&subs->stream->chip->shutdown_mutex); return snd_pcm_lib_free_vmalloc_buffer(substream); } @@ -435,19 +528,28 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) return -ENXIO; } + if (snd_BUG_ON(!subs->data_endpoint)) + return -EIO; + /* some unit conversions in runtime */ - subs->maxframesize = bytes_to_frames(runtime, subs->maxpacksize); - subs->curframesize = bytes_to_frames(runtime, subs->curpacksize); + subs->data_endpoint->maxframesize = + bytes_to_frames(runtime, subs->data_endpoint->maxpacksize); + subs->data_endpoint->curframesize = + bytes_to_frames(runtime, subs->data_endpoint->curpacksize); /* reset the pointer */ subs->hwptr_done = 0; subs->transfer_done = 0; - subs->phase = 0; subs->last_delay = 0; subs->last_frame_number = 0; runtime->delay = 0; - return snd_usb_substream_prepare(subs, runtime); + /* for playback, submit the URBs now; otherwise, the first hwptr_done + * updates for all URBs would happen at the same time when starting */ + if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) + return start_endpoints(subs); + + return 0; } static struct snd_pcm_hardware snd_usb_hardware = @@ -699,6 +801,9 @@ static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, int count = 0, needs_knot = 0; int err; + kfree(subs->rate_list.list); + subs->rate_list.list = NULL; + list_for_each_entry(fp, &subs->fmt_list, list) { if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS) return 0; @@ -845,15 +950,174 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_usb_substream *subs = &as->substream[direction]; + stop_endpoints(subs, 0, 0, 0); + if (!as->chip->shutdown && subs->interface >= 0) { usb_set_interface(subs->dev, subs->interface, 0); subs->interface = -1; } + subs->pcm_substream = NULL; snd_usb_autosuspend(subs->stream->chip); + return 0; } +/* Since a URB can handle only a single linear buffer, we must use double + * buffering when the data to be transferred overflows the buffer boundary. + * To avoid inconsistencies when updating hwptr_done, we use double buffering + * for all URBs. + */ +static void retire_capture_urb(struct snd_usb_substream *subs, + struct urb *urb) +{ + struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; + unsigned int stride, frames, bytes, oldptr; + int i, period_elapsed = 0; + unsigned long flags; + unsigned char *cp; + + stride = runtime->frame_bits >> 3; + + for (i = 0; i < urb->number_of_packets; i++) { + cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset; + if (urb->iso_frame_desc[i].status && printk_ratelimit()) { + snd_printdd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); + // continue; + } + bytes = urb->iso_frame_desc[i].actual_length; + frames = bytes / stride; + if (!subs->txfr_quirk) + bytes = frames * stride; + if (bytes % (runtime->sample_bits >> 3) != 0) { +#ifdef CONFIG_SND_DEBUG_VERBOSE + int oldbytes = bytes; +#endif + bytes = frames * stride; + snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n", + oldbytes, bytes); + } + /* update the current pointer */ + spin_lock_irqsave(&subs->lock, flags); + oldptr = subs->hwptr_done; + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + frames = (bytes + (oldptr % stride)) / stride; + subs->transfer_done += frames; + if (subs->transfer_done >= runtime->period_size) { + subs->transfer_done -= runtime->period_size; + period_elapsed = 1; + } + spin_unlock_irqrestore(&subs->lock, flags); + /* copy a data chunk */ + if (oldptr + bytes > runtime->buffer_size * stride) { + unsigned int bytes1 = + runtime->buffer_size * stride - oldptr; + memcpy(runtime->dma_area + oldptr, cp, bytes1); + memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1); + } else { + memcpy(runtime->dma_area + oldptr, cp, bytes); + } + } + + if (period_elapsed) + snd_pcm_period_elapsed(subs->pcm_substream); +} + +static void prepare_playback_urb(struct snd_usb_substream *subs, + struct urb *urb) +{ + struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; + struct snd_urb_ctx *ctx = urb->context; + unsigned int counts, frames, bytes; + int i, stride, period_elapsed = 0; + unsigned long flags; + + stride = runtime->frame_bits >> 3; + + frames = 0; + urb->number_of_packets = 0; + spin_lock_irqsave(&subs->lock, flags); + for (i = 0; i < ctx->packets; i++) { + counts = ctx->packet_size[i]; + /* set up descriptor */ + urb->iso_frame_desc[i].offset = frames * stride; + urb->iso_frame_desc[i].length = counts * stride; + frames += counts; + urb->number_of_packets++; + subs->transfer_done += counts; + if (subs->transfer_done >= runtime->period_size) { + subs->transfer_done -= runtime->period_size; + period_elapsed = 1; + if (subs->fmt_type == UAC_FORMAT_TYPE_II) { + if (subs->transfer_done > 0) { + /* FIXME: fill-max mode is not + * supported yet */ + frames -= subs->transfer_done; + counts -= subs->transfer_done; + urb->iso_frame_desc[i].length = + counts * stride; + subs->transfer_done = 0; + } + i++; + if (i < ctx->packets) { + /* add a transfer delimiter */ + urb->iso_frame_desc[i].offset = + frames * stride; + urb->iso_frame_desc[i].length = 0; + urb->number_of_packets++; + } + break; + } + } + if (period_elapsed && + !snd_usb_endpoint_implict_feedback_sink(subs->data_endpoint)) /* finish at the period boundary */ + break; + } + bytes = frames * stride; + if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { + /* err, the transferred area goes over buffer boundary. */ + unsigned int bytes1 = + runtime->buffer_size * stride - subs->hwptr_done; + memcpy(urb->transfer_buffer, + runtime->dma_area + subs->hwptr_done, bytes1); + memcpy(urb->transfer_buffer + bytes1, + runtime->dma_area, bytes - bytes1); + } else { + memcpy(urb->transfer_buffer, + runtime->dma_area + subs->hwptr_done, bytes); + } + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + runtime->delay += frames; + spin_unlock_irqrestore(&subs->lock, flags); + urb->transfer_buffer_length = bytes; + if (period_elapsed) + snd_pcm_period_elapsed(subs->pcm_substream); +} + +/* + * process after playback data complete + * - decrease the delay count again + */ +static void retire_playback_urb(struct snd_usb_substream *subs, + struct urb *urb) +{ + unsigned long flags; + struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; + int stride = runtime->frame_bits >> 3; + int processed = urb->transfer_buffer_length / stride; + + spin_lock_irqsave(&subs->lock, flags); + if (processed > runtime->delay) + runtime->delay = 0; + else + runtime->delay -= processed; + spin_unlock_irqrestore(&subs->lock, flags); +} + static int snd_usb_playback_open(struct snd_pcm_substream *substream) { return snd_usb_pcm_open(substream, SNDRV_PCM_STREAM_PLAYBACK); @@ -874,6 +1138,64 @@ static int snd_usb_capture_close(struct snd_pcm_substream *substream) return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_CAPTURE); } +static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_usb_substream *subs = substream->runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + subs->data_endpoint->prepare_data_urb = prepare_playback_urb; + subs->data_endpoint->retire_data_urb = retire_playback_urb; + subs->running = 1; + return 0; + case SNDRV_PCM_TRIGGER_STOP: + stop_endpoints(subs, 0, 0, 0); + subs->running = 0; + return 0; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + subs->data_endpoint->prepare_data_urb = NULL; + subs->data_endpoint->retire_data_urb = NULL; + subs->running = 0; + return 0; + } + + return -EINVAL; +} + +static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + int err; + struct snd_usb_substream *subs = substream->runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + err = start_endpoints(subs); + if (err < 0) + return err; + + subs->data_endpoint->retire_data_urb = retire_capture_urb; + subs->running = 1; + return 0; + case SNDRV_PCM_TRIGGER_STOP: + stop_endpoints(subs, 0, 0, 0); + subs->running = 0; + return 0; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + subs->data_endpoint->retire_data_urb = NULL; + subs->running = 0; + return 0; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + subs->data_endpoint->retire_data_urb = retire_capture_urb; + subs->running = 1; + return 0; + } + + return -EINVAL; +} + static struct snd_pcm_ops snd_usb_playback_ops = { .open = snd_usb_playback_open, .close = snd_usb_playback_close, diff --git a/sound/usb/proc.c b/sound/usb/proc.c index 961c9a250686..ebc1a5b5b3f1 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -25,6 +25,7 @@ #include "usbaudio.h" #include "helper.h" #include "card.h" +#include "endpoint.h" #include "proc.h" /* convert our full speed USB rate into sampling rate in Hz */ @@ -115,28 +116,33 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s } } +static void proc_dump_ep_status(struct snd_usb_substream *subs, + struct snd_usb_endpoint *ep, + struct snd_info_buffer *buffer) +{ + if (!ep) + return; + snd_iprintf(buffer, " Packet Size = %d\n", ep->curpacksize); + snd_iprintf(buffer, " Momentary freq = %u Hz (%#x.%04x)\n", + snd_usb_get_speed(subs->dev) == USB_SPEED_FULL + ? get_full_speed_hz(ep->freqm) + : get_high_speed_hz(ep->freqm), + ep->freqm >> 16, ep->freqm & 0xffff); + if (ep->freqshift != INT_MIN) { + int res = 16 - ep->freqshift; + snd_iprintf(buffer, " Feedback Format = %d.%d\n", + (ep->syncmaxsize > 3 ? 32 : 24) - res, res); + } +} + static void proc_dump_substream_status(struct snd_usb_substream *subs, struct snd_info_buffer *buffer) { if (subs->running) { - unsigned int i; snd_iprintf(buffer, " Status: Running\n"); snd_iprintf(buffer, " Interface = %d\n", subs->interface); snd_iprintf(buffer, " Altset = %d\n", subs->altset_idx); - snd_iprintf(buffer, " URBs = %d [ ", subs->nurbs); - for (i = 0; i < subs->nurbs; i++) - snd_iprintf(buffer, "%d ", subs->dataurb[i].packets); - snd_iprintf(buffer, "]\n"); - snd_iprintf(buffer, " Packet Size = %d\n", subs->curpacksize); - snd_iprintf(buffer, " Momentary freq = %u Hz (%#x.%04x)\n", - snd_usb_get_speed(subs->dev) == USB_SPEED_FULL - ? get_full_speed_hz(subs->freqm) - : get_high_speed_hz(subs->freqm), - subs->freqm >> 16, subs->freqm & 0xffff); - if (subs->freqshift != INT_MIN) - snd_iprintf(buffer, " Feedback Format = %d.%d\n", - (subs->syncmaxsize > 3 ? 32 : 24) - - (16 - subs->freqshift), - 16 - subs->freqshift); + proc_dump_ep_status(subs, subs->data_endpoint, buffer); + proc_dump_ep_status(subs, subs->sync_endpoint, buffer); } else { snd_iprintf(buffer, " Status: Stop\n"); } diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d89ab4c7d44b..79780fa57a43 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1831,6 +1831,36 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE(0x0582, 0x014d), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "BOSS", */ + /* .product_name = "GT-100", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* Guillemot devices */ { diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 5ff8010b2d6f..083ed81160e5 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -73,6 +73,32 @@ static void snd_usb_audio_pcm_free(struct snd_pcm *pcm) } } +/* + * initialize the substream instance. + */ + +static void snd_usb_init_substream(struct snd_usb_stream *as, + int stream, + struct audioformat *fp) +{ + struct snd_usb_substream *subs = &as->substream[stream]; + + INIT_LIST_HEAD(&subs->fmt_list); + spin_lock_init(&subs->lock); + + subs->stream = as; + subs->direction = stream; + subs->dev = as->chip->dev; + subs->txfr_quirk = as->chip->txfr_quirk; + + snd_usb_set_pcm_ops(as->pcm, stream); + + list_add_tail(&fp->list, &subs->fmt_list); + subs->formats |= fp->formats; + subs->num_formats++; + subs->fmt_type = fp->fmt_type; + subs->ep_num = fp->endpoint; +} /* * add this endpoint to the chip instance. @@ -94,9 +120,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, if (as->fmt_type != fp->fmt_type) continue; subs = &as->substream[stream]; - if (!subs->endpoint) - continue; - if (subs->endpoint == fp->endpoint) { + if (subs->ep_num == fp->endpoint) { list_add_tail(&fp->list, &subs->fmt_list); subs->num_formats++; subs->formats |= fp->formats; @@ -109,7 +133,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, if (as->fmt_type != fp->fmt_type) continue; subs = &as->substream[stream]; - if (subs->endpoint) + if (subs->ep_num) continue; err = snd_pcm_new_stream(as->pcm, stream, 1); if (err < 0) diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 3e2b03577936..b8233ebe250f 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -36,6 +36,7 @@ struct snd_usb_audio { struct snd_card *card; struct usb_interface *pm_intf; u32 usb_id; + struct mutex mutex; struct mutex shutdown_mutex; unsigned int shutdown:1; unsigned int probing:1; @@ -46,6 +47,7 @@ struct snd_usb_audio { int num_suspended_intf; struct list_head pcm_list; /* list of pcm streams */ + struct list_head ep_list; /* list of audio-related endpoints */ int pcm_devs; struct list_head midi_list; /* list of midi interfaces */ |