diff options
| author |   Mauro Carvalho Chehab <mchehab@redhat.com> | 2012-10-05 09:36:26 -0300 |
|---|---|---|
| committer | Mauro Carvalho Chehab <mchehab@redhat.com> | 2012-10-05 09:36:26 -0300 |
| commit | bd0d10498826ed150da5e4c45baf8b9c7088fb71 (patch) | |
| tree | cdee4371121a355d627a655c4eef5c0047b0462a /sound | |
| parent | [media] v4l2-ioctl.c: fix overlay support (diff) | |
| parent | em28xx: regression fix: use DRX-K sync firmware requests on em28xx (diff) | |
| download | linux-dev-bd0d10498826ed150da5e4c45baf8b9c7088fb71.tar.xz linux-dev-bd0d10498826ed150da5e4c45baf8b9c7088fb71.zip | |
Merge branch 'staging/for_v3.7' into v4l_for_linus
* staging/for_v3.7: (2891 commits)
em28xx: regression fix: use DRX-K sync firmware requests on em28xx
drxk: allow loading firmware synchrousnously
em28xx: Make all em28xx extensions to be initialized asynchronously
[media] tda18271: properly report read errors in tda18271_get_id
[media] tda18271: delay IR & RF calibration until init() if delay_cal is set
[media] MAINTAINERS: add Michael Krufky as tda827x maintainer
[media] MAINTAINERS: add Michael Krufky as tda8290 maintainer
[media] MAINTAINERS: add Michael Krufky as cxusb maintainer
[media] MAINTAINERS: add Michael Krufky as lg2160 maintainer
[media] MAINTAINERS: add Michael Krufky as lgdt3305 maintainer
[media] MAINTAINERS: add Michael Krufky as mxl111sf maintainer
[media] MAINTAINERS: add Michael Krufky as mxl5007t maintainer
[media] MAINTAINERS: add Michael Krufky as tda18271 maintainer
[media] s5p-tv: Report only multi-plane capabilities in vidioc_querycap
[media] s5p-mfc: Fix misplaced return statement in s5p_mfc_suspend()
[media] exynos-gsc: Add missing static storage class specifiers
[media] exynos-gsc: Remove <linux/version.h> header file inclusion
[media] s5p-fimc: Fix incorrect condition in fimc_lite_reqbufs()
[media] s5p-tv: Fix potential NULL pointer dereference error
[media] s5k6aa: Fix possible NULL pointer dereference
...
Conflicts:
drivers/media/platform/s5p-fimc/fimc-capture.c
drivers/media/platform/s5p-fimc/fimc-lite.c
Diffstat (limited to 'sound')
76 files changed, 696 insertions, 395 deletions
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 0d7b25e81643..4e1fda75c1c9 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -106,7 +106,7 @@ static struct pxa2xx_pcm_client pxa2xx_ac97_pcm_client = { .prepare = pxa2xx_ac97_pcm_prepare, }; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int pxa2xx_ac97_do_suspend(struct snd_card *card) { @@ -243,7 +243,7 @@ static struct platform_driver pxa2xx_ac97_driver = { .driver = { .name = "pxa2xx-ac97", .owner = THIS_MODULE, -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP .pm = &pxa2xx_ac97_pm_ops, #endif }, diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index eb4ceb71123e..277ebce23a45 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -452,6 +452,7 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev) dac->regs = ioremap(regs->start, resource_size(regs)); if (!dac->regs) { dev_dbg(&pdev->dev, "could not remap register memory\n"); + retval = -ENOMEM; goto out_free_card; } @@ -534,7 +535,7 @@ out_put_pclk: return retval; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int atmel_abdac_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index bf47025bdf45..9052aff37f64 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -278,14 +278,9 @@ static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream, if (retval < 0) return retval; /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ - if (cpu_is_at32ap7000()) { - if (retval < 0) - return retval; - /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ - if (retval == 1) - if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.rx_chan); - } + if (cpu_is_at32ap7000() && retval == 1) + if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.rx_chan); /* Set restrictions to params. */ mutex_lock(&opened_mutex); @@ -980,6 +975,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) if (!chip->regs) { dev_dbg(&pdev->dev, "could not remap register memory\n"); + retval = -ENOMEM; goto err_ioremap; } @@ -1134,7 +1130,7 @@ err_snd_card_new: return retval; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int atmel_ac97c_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ec2118d0e27a..eb60cb8dbb8a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -80,14 +80,12 @@ static int snd_compr_open(struct inode *inode, struct file *f) int maj = imajor(inode); int ret; - if (f->f_flags & O_WRONLY) + if ((f->f_flags & O_ACCMODE) == O_WRONLY) dirn = SND_COMPRESS_PLAYBACK; - else if (f->f_flags & O_RDONLY) + else if ((f->f_flags & O_ACCMODE) == O_RDONLY) dirn = SND_COMPRESS_CAPTURE; - else { - pr_err("invalid direction\n"); + else return -EINVAL; - } if (maj == snd_major) compr = snd_lookup_minor_data(iminor(inode), diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index 4e7ec2b49873..d0f00356fc11 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -101,7 +101,7 @@ void *snd_malloc_sgbuf_pages(struct device *device, if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, device, chunk, &tmpb) < 0) { if (!sgbuf->pages) - return NULL; + goto _failed; if (!res_size) goto _failed; size = sgbuf->pages * PAGE_SIZE; diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 1128b35b2b05..5a34355e78e8 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -1176,7 +1176,7 @@ static int __devexit loopback_remove(struct platform_device *devptr) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int loopback_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index f7d3bfc6bca8..54bb6644a598 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -1064,7 +1064,7 @@ static int __devexit snd_dummy_remove(struct platform_device *devptr) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_dummy_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index 1cff331a228e..4608c2ca43f8 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -554,6 +554,7 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, spin_lock_init(&mpu->output_lock); spin_lock_init(&mpu->timer_lock); mpu->hardware = hardware; + mpu->irq = -1; if (! (info_flags & MPU401_INFO_INTEGRATED)) { int res_size = hardware == MPU401_HW_PC98II ? 4 : 2; mpu->res = request_region(port, res_size, "MPU401 UART"); diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 6ca59fc6dcb9..ef171295f6d4 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -199,7 +199,7 @@ static void pcsp_stop_beep(struct snd_pcsp *chip) pcspkr_stop_sound(); } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int pcsp_suspend(struct device *dev) { struct snd_pcsp *chip = dev_get_drvdata(dev); @@ -212,7 +212,7 @@ static SIMPLE_DEV_PM_OPS(pcsp_pm, pcsp_suspend, NULL); #define PCSP_PM_OPS &pcsp_pm #else #define PCSP_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static void pcsp_shutdown(struct platform_device *dev) { diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index d14edb7d6484..3c6c1e3226f3 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -37,9 +37,6 @@ MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Routines for control of TEA5757/5759 Philips AM/FM radio tuner chips"); MODULE_LICENSE("GPL"); -#define FREQ_LO ((tea->tea5759 ? 760 : 875) * 1600U) -#define FREQ_HI ((tea->tea5759 ? 910 : 1080) * 1600U) - /* * definitions */ @@ -50,8 +47,8 @@ MODULE_LICENSE("GPL"); #define TEA575X_BIT_BAND_MASK (3<<20) #define TEA575X_BIT_BAND_FM (0<<20) #define TEA575X_BIT_BAND_MW (1<<20) -#define TEA575X_BIT_BAND_LW (1<<21) -#define TEA575X_BIT_BAND_SW (1<<22) +#define TEA575X_BIT_BAND_LW (2<<20) +#define TEA575X_BIT_BAND_SW (3<<20) #define TEA575X_BIT_PORT_0 (1<<19) /* user bit */ #define TEA575X_BIT_PORT_1 (1<<18) /* user bit */ #define TEA575X_BIT_SEARCH_MASK (3<<16) /* search level */ @@ -62,6 +59,37 @@ MODULE_LICENSE("GPL"); #define TEA575X_BIT_DUMMY (1<<15) /* buffer */ #define TEA575X_BIT_FREQ_MASK 0x7fff +enum { BAND_FM, BAND_FM_JAPAN, BAND_AM }; + +static const struct v4l2_frequency_band bands[] = { + { + .type = V4L2_TUNER_RADIO, + .index = 0, + .capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO | + V4L2_TUNER_CAP_FREQ_BANDS, + .rangelow = 87500 * 16, + .rangehigh = 108000 * 16, + .modulation = V4L2_BAND_MODULATION_FM, + }, + { + .type = V4L2_TUNER_RADIO, + .index = 0, + .capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO | + V4L2_TUNER_CAP_FREQ_BANDS, + .rangelow = 76000 * 16, + .rangehigh = 91000 * 16, + .modulation = V4L2_BAND_MODULATION_FM, + }, + { + .type = V4L2_TUNER_RADIO, + .index = 1, + .capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_FREQ_BANDS, + .rangelow = 530 * 16, + .rangehigh = 1710 * 16, + .modulation = V4L2_BAND_MODULATION_AM, + }, +}; + /* * lowlevel part */ @@ -133,16 +161,29 @@ static u32 snd_tea575x_val_to_freq(struct snd_tea575x *tea, u32 val) if (freq == 0) return freq; - /* freq *= 12.5 */ - freq *= 125; - freq /= 10; - /* crystal fixup */ - if (tea->tea5759) - freq += TEA575X_FMIF; - else + switch (tea->band) { + case BAND_FM: + /* freq *= 12.5 */ + freq *= 125; + freq /= 10; + /* crystal fixup */ freq -= TEA575X_FMIF; + break; + case BAND_FM_JAPAN: + /* freq *= 12.5 */ + freq *= 125; + freq /= 10; + /* crystal fixup */ + freq += TEA575X_FMIF; + break; + case BAND_AM: + /* crystal fixup */ + freq -= TEA575X_AMIF; + break; + } - return clamp(freq * 16, FREQ_LO, FREQ_HI); /* from kHz */ + return clamp(freq * 16, bands[tea->band].rangelow, + bands[tea->band].rangehigh); /* from kHz */ } static u32 snd_tea575x_get_freq(struct snd_tea575x *tea) @@ -150,21 +191,37 @@ static u32 snd_tea575x_get_freq(struct snd_tea575x *tea) return snd_tea575x_val_to_freq(tea, snd_tea575x_read(tea)); } -static void snd_tea575x_set_freq(struct snd_tea575x *tea) +void snd_tea575x_set_freq(struct snd_tea575x *tea) { - u32 freq = tea->freq; + u32 freq = tea->freq / 16; /* to kHz */ + u32 band = 0; - freq /= 16; /* to kHz */ - /* crystal fixup */ - if (tea->tea5759) - freq -= TEA575X_FMIF; - else + switch (tea->band) { + case BAND_FM: + band = TEA575X_BIT_BAND_FM; + /* crystal fixup */ freq += TEA575X_FMIF; - /* freq /= 12.5 */ - freq *= 10; - freq /= 125; + /* freq /= 12.5 */ + freq *= 10; + freq /= 125; + break; + case BAND_FM_JAPAN: + band = TEA575X_BIT_BAND_FM; + /* crystal fixup */ + freq -= TEA575X_FMIF; + /* freq /= 12.5 */ + freq *= 10; + freq /= 125; + break; + case BAND_AM: + band = TEA575X_BIT_BAND_MW; + /* crystal fixup */ + freq += TEA575X_AMIF; + break; + } - tea->val &= ~TEA575X_BIT_FREQ_MASK; + tea->val &= ~(TEA575X_BIT_FREQ_MASK | TEA575X_BIT_BAND_MASK); + tea->val |= band; tea->val |= freq & TEA575X_BIT_FREQ_MASK; snd_tea575x_write(tea, tea->val); tea->freq = snd_tea575x_val_to_freq(tea, tea->val); @@ -190,23 +247,57 @@ static int vidioc_querycap(struct file *file, void *priv, return 0; } +static int vidioc_enum_freq_bands(struct file *file, void *priv, + struct v4l2_frequency_band *band) +{ + struct snd_tea575x *tea = video_drvdata(file); + int index; + + if (band->tuner != 0) + return -EINVAL; + + switch (band->index) { + case 0: + if (tea->tea5759) + index = BAND_FM_JAPAN; + else + index = BAND_FM; + break; + case 1: + if (tea->has_am) { + index = BAND_AM; + break; + } + /* Fall through */ + default: + return -EINVAL; + } + + *band = bands[index]; + if (!tea->cannot_read_data) + band->capability |= V4L2_TUNER_CAP_HWSEEK_BOUNDED; + + return 0; +} + static int vidioc_g_tuner(struct file *file, void *priv, struct v4l2_tuner *v) { struct snd_tea575x *tea = video_drvdata(file); + struct v4l2_frequency_band band_fm = { 0, }; if (v->index > 0) return -EINVAL; snd_tea575x_read(tea); + vidioc_enum_freq_bands(file, priv, &band_fm); - strcpy(v->name, "FM"); + memset(v, 0, sizeof(*v)); + strlcpy(v->name, tea->has_am ? "FM/AM" : "FM", sizeof(v->name)); v->type = V4L2_TUNER_RADIO; - v->capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO; - if (!tea->cannot_read_data) - v->capability |= V4L2_TUNER_CAP_HWSEEK_BOUNDED; - v->rangelow = FREQ_LO; - v->rangehigh = FREQ_HI; + v->capability = band_fm.capability; + v->rangelow = tea->has_am ? bands[BAND_AM].rangelow : band_fm.rangelow; + v->rangehigh = band_fm.rangehigh; v->rxsubchans = tea->stereo ? V4L2_TUNER_SUB_STEREO : V4L2_TUNER_SUB_MONO; v->audmode = (tea->val & TEA575X_BIT_MONO) ? V4L2_TUNER_MODE_MONO : V4L2_TUNER_MODE_STEREO; @@ -218,13 +309,17 @@ static int vidioc_s_tuner(struct file *file, void *priv, struct v4l2_tuner *v) { struct snd_tea575x *tea = video_drvdata(file); + u32 orig_val = tea->val; if (v->index) return -EINVAL; tea->val &= ~TEA575X_BIT_MONO; if (v->audmode == V4L2_TUNER_MODE_MONO) tea->val |= TEA575X_BIT_MONO; - snd_tea575x_write(tea, tea->val); + /* Only apply changes if currently tuning FM */ + if (tea->band != BAND_AM && tea->val != orig_val) + snd_tea575x_set_freq(tea); + return 0; } @@ -248,24 +343,56 @@ static int vidioc_s_frequency(struct file *file, void *priv, if (f->tuner != 0 || f->type != V4L2_TUNER_RADIO) return -EINVAL; - tea->val &= ~TEA575X_BIT_SEARCH; - tea->freq = clamp(f->frequency, FREQ_LO, FREQ_HI); + if (tea->has_am && f->frequency < (20000 * 16)) + tea->band = BAND_AM; + else if (tea->tea5759) + tea->band = BAND_FM_JAPAN; + else + tea->band = BAND_FM; + + tea->freq = clamp(f->frequency, bands[tea->band].rangelow, + bands[tea->band].rangehigh); snd_tea575x_set_freq(tea); return 0; } static int vidioc_s_hw_freq_seek(struct file *file, void *fh, - struct v4l2_hw_freq_seek *a) + const struct v4l2_hw_freq_seek *a) { struct snd_tea575x *tea = video_drvdata(file); unsigned long timeout; - int i; + int i, spacing; if (tea->cannot_read_data) return -ENOTTY; if (a->tuner || a->wrap_around) return -EINVAL; + if (file->f_flags & O_NONBLOCK) + return -EWOULDBLOCK; + + if (a->rangelow || a->rangehigh) { + for (i = 0; i < ARRAY_SIZE(bands); i++) { + if ((i == BAND_FM && tea->tea5759) || + (i == BAND_FM_JAPAN && !tea->tea5759) || + (i == BAND_AM && !tea->has_am)) + continue; + if (bands[i].rangelow == a->rangelow && + bands[i].rangehigh == a->rangehigh) + break; + } + if (i == ARRAY_SIZE(bands)) + return -EINVAL; /* No matching band found */ + if (i != tea->band) { + tea->band = i; + tea->freq = clamp(tea->freq, bands[i].rangelow, + bands[i].rangehigh); + snd_tea575x_set_freq(tea); + } + } + + spacing = (tea->band == BAND_AM) ? 5 : 50; /* kHz */ + /* clear the frequency, HW will fill it in */ tea->val &= ~TEA575X_BIT_FREQ_MASK; tea->val |= TEA575X_BIT_SEARCH; @@ -297,10 +424,10 @@ static int vidioc_s_hw_freq_seek(struct file *file, void *fh, if (freq == 0) /* shouldn't happen */ break; /* - * if we moved by less than 50 kHz, or in the wrong - * direction, continue seeking + * if we moved by less than the spacing, or in the + * wrong direction, continue seeking */ - if (abs(tea->freq - freq) < 16 * 50 || + if (abs(tea->freq - freq) < 16 * spacing || (a->seek_upward && freq < tea->freq) || (!a->seek_upward && freq > tea->freq)) { snd_tea575x_write(tea, tea->val); @@ -344,6 +471,7 @@ static const struct v4l2_ioctl_ops tea575x_ioctl_ops = { .vidioc_g_frequency = vidioc_g_frequency, .vidioc_s_frequency = vidioc_s_frequency, .vidioc_s_hw_freq_seek = vidioc_s_hw_freq_seek, + .vidioc_enum_freq_bands = vidioc_enum_freq_bands, .vidioc_log_status = v4l2_ctrl_log_status, .vidioc_subscribe_event = v4l2_ctrl_subscribe_event, .vidioc_unsubscribe_event = v4l2_event_unsubscribe, @@ -446,3 +574,4 @@ module_exit(alsa_tea575x_module_exit) EXPORT_SYMBOL(snd_tea575x_init); EXPORT_SYMBOL(snd_tea575x_exit); +EXPORT_SYMBOL(snd_tea575x_set_freq); diff --git a/sound/isa/als100.c b/sound/isa/als100.c index 2d67c78c9f4b..f7cdaf51512d 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -233,7 +233,7 @@ static int __devinit snd_card_als100_probe(int dev, irq[dev], dma8[dev], dma16[dev]); } - if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) { + if ((error = snd_sb16dsp_pcm(chip, 0, &chip->pcm)) < 0) { snd_card_free(card); return error; } diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index 1d47be8170b5..b3b4f15e45ba 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -612,10 +612,10 @@ static int snd_es1688_capture_close(struct snd_pcm_substream *substream) static int snd_es1688_free(struct snd_es1688 *chip) { - if (chip->res_port) { + if (chip->hardware != ES1688_HW_UNDEF) snd_es1688_init(chip, 0); + if (chip->res_port) release_and_free_resource(chip->res_port); - } if (chip->irq >= 0) free_irq(chip->irq, (void *) chip); if (chip->dma8 >= 0) { @@ -657,19 +657,27 @@ int snd_es1688_create(struct snd_card *card, return -ENOMEM; chip->irq = -1; chip->dma8 = -1; + chip->hardware = ES1688_HW_UNDEF; - if ((chip->res_port = request_region(port + 4, 12, "ES1688")) == NULL) { + chip->res_port = request_region(port + 4, 12, "ES1688"); + if (chip->res_port == NULL) { snd_printk(KERN_ERR "es1688: can't grab port 0x%lx\n", port + 4); - return -EBUSY; + err = -EBUSY; + goto exit; } - if (request_irq(irq, snd_es1688_interrupt, 0, "ES1688", (void *) chip)) { + + err = request_irq(irq, snd_es1688_interrupt, 0, "ES1688", (void *) chip); + if (err < 0) { snd_printk(KERN_ERR "es1688: can't grab IRQ %d\n", irq); - return -EBUSY; + goto exit; } + chip->irq = irq; - if (request_dma(dma8, "ES1688")) { + err = request_dma(dma8, "ES1688"); + + if (err < 0) { snd_printk(KERN_ERR "es1688: can't grab DMA8 %d\n", dma8); - return -EBUSY; + goto exit; } chip->dma8 = dma8; @@ -685,14 +693,18 @@ int snd_es1688_create(struct snd_card *card, err = snd_es1688_probe(chip); if (err < 0) - return err; + goto exit; err = snd_es1688_init(chip, 1); if (err < 0) - return err; + goto exit; /* Register device */ - return snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); +exit: + if (err) + snd_es1688_free(chip); + return err; } static struct snd_pcm_ops snd_es1688_playback_ops = { diff --git a/sound/oss/sb_audio.c b/sound/oss/sb_audio.c index 733b014ec7d1..b2b3c014221a 100644 --- a/sound/oss/sb_audio.c +++ b/sound/oss/sb_audio.c @@ -575,13 +575,15 @@ static int jazz16_audio_set_speed(int dev, int speed) if (speed > 0) { int tmp; - int s = speed * devc->channels; + int s; if (speed < 5000) speed = 5000; if (speed > 44100) speed = 44100; + s = speed * devc->channels; + devc->tconst = (256 - ((1000000 + s / 2) / s)) & 0xff; tmp = 256 - devc->tconst; diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index f75f5ffdfdfb..a71d1c14a0f6 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -94,7 +94,7 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip, if (snd_BUG_ON(codec_index != CS46XX_PRIMARY_CODEC_INDEX && codec_index != CS46XX_SECONDARY_CODEC_INDEX)) - return -EINVAL; + return 0xffff; chip->active_ctrl(chip, 1); diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 8e40262d4117..2f6e9c762d3f 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1725,8 +1725,10 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, atc_connect_resources(atc); atc->timer = ct_timer_new(atc); - if (!atc->timer) + if (!atc->timer) { + err = -ENOMEM; goto error1; + } err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, atc, &ops); if (err < 0) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 4f502a2bdc3c..0a436626182b 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -326,7 +326,10 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst for (page = blk->first_page; page <= blk->last_page; page++, idx++) { unsigned long ofs = idx << PAGE_SHIFT; dma_addr_t addr; - addr = snd_pcm_sgbuf_get_addr(substream, ofs); + if (ofs >= runtime->dma_bytes) + addr = emu->silent_page.addr; + else + addr = snd_pcm_sgbuf_get_addr(substream, ofs); if (! is_valid_page(emu, addr)) { printk(KERN_ERR "emu: failure page = %d\n", idx); mutex_unlock(&hdr->block_mutex); diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 647218d69f68..4f7d2dfcef7b 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -332,13 +332,12 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, 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:"); + snd_printd(" inputs:\n"); for (i = 0; i < cfg->num_inputs; i++) { - snd_printd(" %s=0x%x", + snd_printd(" %s=0x%x\n", 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); diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 0bc2315b181d..0849aac449f2 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -231,16 +231,22 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) } EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device); +static bool ctl_has_mute(struct snd_kcontrol *kcontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + return query_amp_caps(codec, get_amp_nid(kcontrol), + get_amp_direction(kcontrol)) & AC_AMPCAP_MUTE; +} + /* 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) { + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { ucontrol->value.integer.value[0] = - ucontrol->value.integer.value[1] = - beep->enabled; + ucontrol->value.integer.value[1] = beep->enabled; return 0; } return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); @@ -252,9 +258,20 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, { 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); + if (beep) { + u8 chs = get_amp_channels(kcontrol); + int enable = 0; + long *valp = ucontrol->value.integer.value; + if (chs & 1) { + enable |= *valp; + valp++; + } + if (chs & 2) + enable |= *valp; + snd_hda_enable_beep_device(codec, enable); + } + if (!ctl_has_mute(kcontrol)) + return 0; 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_codec.c b/sound/pci/hda/hda_codec.c index 88a9c20eb7a2..1c65cc5e3a31 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1209,6 +1209,9 @@ static void snd_hda_codec_free(struct hda_codec *codec) kfree(codec); } +static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, + hda_nid_t fg, unsigned int power_state); + static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); @@ -1317,6 +1320,10 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, AC_VERB_GET_SUBSYSTEM_ID, 0); } + codec->epss = snd_hda_codec_get_supported_ps(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_EPSS); + /* power-up all before initialization */ hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, @@ -1386,6 +1393,44 @@ int snd_hda_codec_configure(struct hda_codec *codec) } EXPORT_SYMBOL_HDA(snd_hda_codec_configure); +/* update the stream-id if changed */ +static void update_pcm_stream_id(struct hda_codec *codec, + struct hda_cvt_setup *p, hda_nid_t nid, + u32 stream_tag, int channel_id) +{ + unsigned int oldval, newval; + + if (p->stream_tag != stream_tag || p->channel_id != channel_id) { + oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); + newval = (stream_tag << 4) | channel_id; + if (oldval != newval) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CHANNEL_STREAMID, + newval); + p->stream_tag = stream_tag; + p->channel_id = channel_id; + } +} + +/* update the format-id if changed */ +static void update_pcm_format(struct hda_codec *codec, struct hda_cvt_setup *p, + hda_nid_t nid, int format) +{ + unsigned int oldval; + + if (p->format_id != format) { + oldval = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_STREAM_FORMAT, 0); + if (oldval != format) { + msleep(1); + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_STREAM_FORMAT, + format); + } + p->format_id = format; + } +} + /** * snd_hda_codec_setup_stream - set up the codec for streaming * @codec: the CODEC to set up @@ -1400,7 +1445,6 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, { struct hda_codec *c; struct hda_cvt_setup *p; - unsigned int oldval, newval; int type; int i; @@ -1413,29 +1457,13 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, p = get_hda_cvt_setup(codec, nid); if (!p) return; - /* update the stream-id if changed */ - if (p->stream_tag != stream_tag || p->channel_id != channel_id) { - oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - newval = (stream_tag << 4) | channel_id; - if (oldval != newval) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - newval); - p->stream_tag = stream_tag; - p->channel_id = channel_id; - } - /* update the format-id if changed */ - if (p->format_id != format) { - oldval = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_STREAM_FORMAT, 0); - if (oldval != format) { - msleep(1); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, - format); - } - p->format_id = format; - } + + if (codec->pcm_format_first) + update_pcm_format(codec, p, nid, format); + update_pcm_stream_id(codec, p, nid, stream_tag, channel_id); + if (!codec->pcm_format_first) + update_pcm_format(codec, p, nid, format); + p->active = 1; p->dirty = 0; @@ -2325,6 +2353,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) } if (codec->patch_ops.free) codec->patch_ops.free(codec); + memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); snd_hda_jack_tbl_clear(codec); codec->proc_widget_hook = NULL; codec->spec = NULL; @@ -2340,7 +2369,6 @@ int snd_hda_codec_reset(struct hda_codec *codec) codec->num_pcms = 0; codec->pcm_info = NULL; codec->preset = NULL; - memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); codec->slave_dig_outs = NULL; codec->spdif_status_reset = 0; module_put(codec->owner); @@ -3497,7 +3525,7 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg { int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE); - if (sup < 0) + if (sup == -1) return false; if (sup & power_state) return true; @@ -3522,8 +3550,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, /* this delay seems necessary to avoid click noise at power-down */ 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); + msleep(codec->epss ? 10 : 100); } /* repeat power states setting at most 10 times*/ @@ -4433,6 +4460,8 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) * then there is no need to go through power up here. */ if (codec->power_on) { + if (codec->power_transition < 0) + codec->power_transition = 0; spin_unlock(&codec->power_lock); return; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c422d330ca54..e5a7e19a8071 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -861,6 +861,8 @@ struct hda_codec { unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */ unsigned int no_jack_detect:1; /* Machine has no jack-detection */ + unsigned int pcm_format_first:1; /* PCM format must be set first */ + unsigned int epss:1; /* supporting EPSS? */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ int power_transition; /* power-state in transition */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c8aced182fd1..c4763c52eaf6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -151,6 +151,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, CPT}," "{Intel, PPT}," "{Intel, LPT}," + "{Intel, LPT_LP}," "{Intel, HPT}," "{Intel, PBG}," "{Intel, SCH}," @@ -2700,6 +2701,8 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x1ac3, "ASUS X53S", POS_FIX_POSBUF), + SND_PCI_QUIRK(0x1043, 0x1b43, "ASUS K53E", POS_FIX_POSBUF), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x10de, 0xcb89, "Macbook Pro 7,1", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), @@ -3270,6 +3273,14 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + /* Lynx Point-LP */ + { PCI_DEVICE(0x8086, 0x9c20), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + /* Lynx Point-LP */ + { PCI_DEVICE(0x8086, 0x9c21), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0c0c), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 7e46258fc700..6894ec66258c 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -412,7 +412,7 @@ static void print_digital_conv(struct snd_info_buffer *buffer, if (digi1 & AC_DIG1_EMPHASIS) snd_iprintf(buffer, " Preemphasis"); if (digi1 & AC_DIG1_COPYRIGHT) - snd_iprintf(buffer, " Copyright"); + snd_iprintf(buffer, " Non-Copyright"); if (digi1 & AC_DIG1_NONAUDIO) snd_iprintf(buffer, " Non-Audio"); if (digi1 & AC_DIG1_PROFESSIONAL) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index d0d3540e39e7..49750a96d649 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -246,7 +246,7 @@ static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); } - if (dac) + if (dac && (get_wcaps(codec, dac) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, dac, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO); } @@ -261,7 +261,7 @@ static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc) AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)); } - if (adc) + if (adc && (get_wcaps(codec, adc) & AC_WCAP_IN_AMP)) snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)); } @@ -275,6 +275,10 @@ static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx, int type = dir ? HDA_INPUT : HDA_OUTPUT; struct snd_kcontrol_new knew = HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type); + if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_MUTE) == 0) { + snd_printdd("Skipping '%s %s Switch' (no mute on node 0x%x)\n", pfx, dirstr[dir], nid); + return 0; + } sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } @@ -286,6 +290,10 @@ static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx, int type = dir ? HDA_INPUT : HDA_OUTPUT; struct snd_kcontrol_new knew = HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type); + if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_NUM_STEPS) == 0) { + snd_printdd("Skipping '%s %s Volume' (no amp on node 0x%x)\n", pfx, dirstr[dir], nid); + return 0; + } sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]); return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } @@ -464,50 +472,17 @@ exit: } /* - * PCM stuffs + * PCM callbacks */ -static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid, - u32 stream_tag, - int channel_id, int format) +static int ca0132_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { - unsigned int oldval, newval; - - if (!nid) - return; - - snd_printdd("ca0132_setup_stream: " - "NID=0x%x, stream=0x%x, channel=%d, format=0x%x\n", - nid, stream_tag, channel_id, format); - - /* update the format-id if changed */ - oldval = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_STREAM_FORMAT, - 0); - if (oldval != format) { - msleep(20); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, - format); - } - - oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - newval = (stream_tag << 4) | channel_id; - if (oldval != newval) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - newval); - } -} - -static void ca0132_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) -{ - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); + struct ca0132_spec *spec = codec->spec; + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); } -/* - * PCM callbacks - */ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -515,10 +490,8 @@ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format); - - return 0; + return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, + stream_tag, format, substream); } static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, @@ -526,92 +499,45 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->dacs[0]); - - return 0; + return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); } /* * Digital out */ -static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) +static int ca0132_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->dig_out, stream_tag, 0, format); - - return 0; + return snd_hda_multi_out_dig_open(codec, &spec->multiout); } -static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->dig_out); - - return 0; -} - -/* - * Analog capture - */ -static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo, +static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->adcs[substream->number], - stream_tag, 0, format); - - return 0; + return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, + stream_tag, format, substream); } -static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, +static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->adcs[substream->number]); - - return 0; + return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); } -/* - * Digital capture - */ -static int ca0132_dig_capture_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) +static int ca0132_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->dig_in, stream_tag, 0, format); - - return 0; -} - -static int ca0132_dig_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->dig_in); - - return 0; + return snd_hda_multi_out_dig_close(codec, &spec->multiout); } /* @@ -621,6 +547,7 @@ static struct hda_pcm_stream ca0132_pcm_analog_playback = { .channels_min = 2, .channels_max = 2, .ops = { + .open = ca0132_playback_pcm_open, .prepare = ca0132_playback_pcm_prepare, .cleanup = ca0132_playback_pcm_cleanup }, @@ -630,10 +557,6 @@ static struct hda_pcm_stream ca0132_pcm_analog_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, - .ops = { - .prepare = ca0132_capture_pcm_prepare, - .cleanup = ca0132_capture_pcm_cleanup - }, }; static struct hda_pcm_stream ca0132_pcm_digital_playback = { @@ -641,6 +564,8 @@ static struct hda_pcm_stream ca0132_pcm_digital_playback = { .channels_min = 2, .channels_max = 2, .ops = { + .open = ca0132_dig_playback_pcm_open, + .close = ca0132_dig_playback_pcm_close, .prepare = ca0132_dig_playback_pcm_prepare, .cleanup = ca0132_dig_playback_pcm_cleanup }, @@ -650,10 +575,6 @@ static struct hda_pcm_stream ca0132_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, - .ops = { - .prepare = ca0132_dig_capture_pcm_prepare, - .cleanup = ca0132_dig_capture_pcm_cleanup - }, }; static int ca0132_build_pcms(struct hda_codec *codec) @@ -928,18 +849,16 @@ static int ca0132_build_controls(struct hda_codec *codec) spec->dig_out); if (err < 0) return err; - err = add_out_volume(codec, spec->dig_out, "IEC958"); + err = snd_hda_create_spdif_share_sw(codec, &spec->multiout); if (err < 0) return err; + /* spec->multiout.share_spdif = 1; */ } if (spec->dig_in) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in); if (err < 0) return err; - err = add_in_volume(codec, spec->dig_in, "IEC958"); - if (err < 0) - return err; } return 0; } @@ -961,6 +880,9 @@ static void ca0132_config(struct hda_codec *codec) struct ca0132_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; + codec->pcm_format_first = 1; + codec->no_sticky_stream = 1; + /* line-outs */ cfg->line_outs = 1; cfg->line_out_pins[0] = 0x0b; /* front */ @@ -988,14 +910,24 @@ static void ca0132_config(struct hda_codec *codec) /* Mic-in */ spec->input_pins[0] = 0x12; - spec->input_labels[0] = "Mic-In"; + spec->input_labels[0] = "Mic"; spec->adcs[0] = 0x07; /* Line-In */ spec->input_pins[1] = 0x11; - spec->input_labels[1] = "Line-In"; + spec->input_labels[1] = "Line"; spec->adcs[1] = 0x08; spec->num_inputs = 2; + + /* SPDIF I/O */ + spec->dig_out = 0x05; + spec->multiout.dig_out_nid = spec->dig_out; + cfg->dig_out_pins[0] = 0x0c; + cfg->dig_outs = 1; + cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF; + spec->dig_in = 0x09; + cfg->dig_in_pin = 0x0e; + cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; } static void ca0132_init_chip(struct hda_codec *codec) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 14361184ae1e..5e22a8f43d2e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2967,12 +2967,10 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { }; static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { - SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT5066_AUTO), SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO), - SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD), @@ -2988,14 +2986,10 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T510", CXT5066_AUTO), - SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520 & W520", CXT5066_AUTO), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO), - SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO), {} }; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 641408dc28c0..8f23374fa642 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1165,14 +1165,20 @@ static int generic_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) { + snd_hda_codec_cleanup_stream(codec, hinfo->nid); + return 0; +} + +static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ struct hdmi_spec *spec = codec->spec; int cvt_idx, pin_idx; struct hdmi_spec_per_cvt *per_cvt; struct hdmi_spec_per_pin *per_pin; int pinctl; - snd_hda_codec_cleanup_stream(codec, hinfo->nid); - if (hinfo->nid) { cvt_idx = cvt_nid_to_cvt_index(spec, hinfo->nid); if (snd_BUG_ON(cvt_idx < 0)) @@ -1195,12 +1201,12 @@ static int generic_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, pinctl & ~PIN_OUT); snd_hda_spdif_ctls_unassign(codec, pin_idx); } - return 0; } static const struct hda_pcm_ops generic_ops = { .open = hdmi_pcm_open, + .close = hdmi_pcm_close, .prepare = generic_hdmi_playback_pcm_prepare, .cleanup = generic_hdmi_playback_pcm_cleanup, }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f141395dfee6..4f81dd44c837 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -203,6 +203,7 @@ struct alc_spec { 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? */ + unsigned int no_primary_hp:1; /* Don't prefer HP pins to speaker pins */ /* auto-mute control */ int automute_mode; @@ -4323,7 +4324,8 @@ static int alc_parse_auto_config(struct hda_codec *codec, return 0; /* can't find valid BIOS pin config */ } - if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT && + if (!spec->no_primary_hp && + cfg->line_out_type == AUTO_PIN_SPEAKER_OUT && cfg->line_outs <= cfg->hp_outs) { /* use HP as primary out */ cfg->speaker_outs = cfg->line_outs; @@ -5050,6 +5052,7 @@ enum { ALC889_FIXUP_MBP_VREF, ALC889_FIXUP_IMAC91_VREF, ALC882_FIXUP_INV_DMIC, + ALC882_FIXUP_NO_PRIMARY_HP, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -5171,6 +5174,17 @@ static void alc889_fixup_imac91_vref(struct hda_codec *codec, spec->keep_vref_in_automute = 1; } +/* Don't take HP output as primary + * strangely, the speaker output doesn't work on VAIO Z through DAC 0x05 + */ +static void alc882_fixup_no_primary_hp(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->no_primary_hp = 1; +} + static const struct alc_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { .type = ALC_FIXUP_PINS, @@ -5357,6 +5371,10 @@ static const struct alc_fixup alc882_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, }, + [ALC882_FIXUP_NO_PRIMARY_HP] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc882_fixup_no_primary_hp, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -5391,6 +5409,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), + SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), /* All Apple entries are in codec SSIDs */ SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), @@ -5432,6 +5451,7 @@ static const struct alc_model_fixup alc882_fixup_models[] = { {.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"}, + {.id = ALC882_FIXUP_NO_PRIMARY_HP, .name = "no-primary-hp"}, {} }; @@ -6079,6 +6099,8 @@ static const struct alc_fixup alc269_fixups[] = { [ALC269_FIXUP_PCM_44K] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_pcm_44k, + .chained = true, + .chain_id = ALC269_FIXUP_QUANTA_MUTE }, [ALC269_FIXUP_STEREO_DMIC] = { .type = ALC_FIXUP_FUNC, @@ -6186,9 +6208,11 @@ 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, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), 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, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), #if 0 diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a1596a3b171c..3d4722f0a1ca 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -101,6 +101,8 @@ enum { STAC_92HD83XXX_HP_cNB11_INTQUAD, STAC_HP_DV7_4000, STAC_HP_ZEPHYR, + STAC_92HD83XXX_HP_LED, + STAC_92HD83XXX_HP_INV_LED, STAC_92HD83XXX_MODELS }; @@ -1073,7 +1075,7 @@ static struct snd_kcontrol_new stac_smux_mixer = { static const char * const slave_pfxs[] = { "Front", "Surround", "Center", "LFE", "Side", - "Headphone", "Speaker", "IEC958", + "Headphone", "Speaker", "IEC958", "PCM", NULL }; @@ -1675,6 +1677,8 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_HP_cNB11_INTQUAD] = "hp_cNB11_intquad", [STAC_HP_DV7_4000] = "hp-dv7-4000", [STAC_HP_ZEPHYR] = "hp-zephyr", + [STAC_92HD83XXX_HP_LED] = "hp-led", + [STAC_92HD83XXX_HP_INV_LED] = "hp-inv-led", }; static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { @@ -1729,6 +1733,8 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3561, "HP", STAC_HP_ZEPHYR), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3660, + "HP Mini", STAC_92HD83XXX_HP_LED), {} /* terminator */ }; @@ -4266,7 +4272,8 @@ static int stac92xx_init(struct hda_codec *codec) unsigned int gpio; int i; - snd_hda_sequence_write(codec, spec->init); + if (spec->init) + snd_hda_sequence_write(codec, spec->init); /* power down adcs initially */ if (spec->powerdown_adcs) @@ -4414,7 +4421,12 @@ static int stac92xx_init(struct hda_codec *codec) snd_hda_jack_report_sync(codec); /* sync mute LED */ - snd_hda_sync_vmaster_hook(&spec->vmaster_mute); + if (spec->gpio_led) { + if (spec->vmaster_mute.hook) + snd_hda_sync_vmaster_hook(&spec->vmaster_mute); + else /* the very first init call doesn't have vmaster yet */ + stac92xx_update_led_status(codec, false); + } /* sync the power-map */ if (spec->num_pwrs) @@ -4531,6 +4543,9 @@ static void stac92xx_line_out_detect(struct hda_codec *codec, struct auto_pin_cfg *cfg = &spec->autocfg; int i; + if (cfg->speaker_outs == 0) + return; + for (i = 0; i < cfg->line_outs; i++) { if (presence) break; @@ -5507,6 +5522,7 @@ static void stac92hd8x_fill_auto_spec(struct hda_codec *codec) static int patch_stac92hd83xxx(struct hda_codec *codec) { struct sigmatel_spec *spec; + int default_polarity = -1; /* no default cfg */ int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -5518,6 +5534,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) snd_hda_codec_set_pincfg(codec, 0xf, 0x2181205e); } + codec->epss = 0; /* longer delay needed for D3 */ codec->no_trigger_sense = 1; codec->spec = spec; @@ -5555,9 +5572,15 @@ again: case STAC_HP_ZEPHYR: spec->init = stac92hd83xxx_hp_zephyr_init; break; + case STAC_92HD83XXX_HP_LED: + default_polarity = 0; + break; + case STAC_92HD83XXX_HP_INV_LED: + default_polarity = 1; + break; } - if (find_mute_led_cfg(codec, -1/*no default cfg*/)) + if (find_mute_led_cfg(codec, default_polarity)) snd_printd("mute LED gpio %d polarity %d\n", spec->gpio_led, spec->gpio_led_polarity); @@ -5730,7 +5753,6 @@ again: /* fallthru */ case 0x111d76b4: /* 6 Port without Analog Mixer */ case 0x111d76b5: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5755,7 +5777,6 @@ again: spec->stream_delay = 40; /* 40 milliseconds */ /* disable VSW */ - spec->init = stac92hd71bxx_core_init; unmute_init++; snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); @@ -5770,7 +5791,6 @@ again: /* fallthru */ default: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5778,6 +5798,9 @@ again: break; } + if (get_wcaps_type(get_wcaps(codec, 0x28)) == AC_WID_VOL_KNB) + spec->init = stac92hd71bxx_core_init; + if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) snd_hda_sequence_write_cache(codec, unmute_init); diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 90645560ed39..430771776915 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1752,6 +1752,14 @@ static int via_suspend(struct hda_codec *codec) { struct via_spec *spec = codec->spec; vt1708_stop_hp_work(spec); + + if (spec->codec_type == VT1802) { + /* Fix pop noise on headphones */ + int i; + for (i = 0; i < spec->autocfg.hp_outs; i++) + snd_hda_set_pin_ctl(codec, spec->autocfg.hp_pins[i], 0); + } + return 0; } #endif @@ -3226,7 +3234,7 @@ static void set_widgets_power_state_vt1718S(struct hda_codec *codec) { struct via_spec *spec = codec->spec; int imux_is_smixer; - unsigned int parm; + unsigned int parm, parm2; /* MUX6 (1eh) = stereo mixer */ imux_is_smixer = snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; @@ -3249,7 +3257,7 @@ static void set_widgets_power_state_vt1718S(struct hda_codec *codec) parm = AC_PWRST_D3; set_pin_power_state(codec, 0x27, &parm); update_power_state(codec, 0x1a, parm); - update_power_state(codec, 0xb, parm); + parm2 = parm; /* for pin 0x0b */ /* PW2 (26h), AOW2 (ah) */ parm = AC_PWRST_D3; @@ -3264,6 +3272,9 @@ static void set_widgets_power_state_vt1718S(struct hda_codec *codec) if (!spec->hp_independent_mode) /* check for redirected HP */ set_pin_power_state(codec, 0x28, &parm); update_power_state(codec, 0x8, parm); + if (!spec->hp_independent_mode && parm2 != AC_PWRST_D3) + parm = parm2; + update_power_state(codec, 0xb, parm); /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ update_power_state(codec, 0x21, imux_is_smixer ? AC_PWRST_D0 : parm); diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 764cc93dbca4..075d5aa1fee0 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -297,6 +297,7 @@ static int ak4396_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem } static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1); +static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = { { @@ -307,7 +308,7 @@ static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = { .info = ak4396_dac_vol_info, .get = ak4396_dac_vol_get, .put = ak4396_dac_vol_put, - .tlv = { .p = db_scale_wm_dac }, + .tlv = { .p = ak4396_db_scale }, }, }; diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index d1ab43706735..5579b08bb35b 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -851,6 +851,8 @@ static int __devinit lx_pcm_create(struct lx6464es *chip) /* hardcoded device name & channel count */ err = snd_pcm_new(chip->card, (char *)card_name, 0, 1, 1, &pcm); + if (err < 0) + return err; pcm->private_data = chip; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index b8ac8710f47f..b12308b5ba2a 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6585,7 +6585,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card, snd_printk(KERN_ERR "HDSPM: " "unable to kmalloc Mixer memory of %d Bytes\n", (int)sizeof(struct hdspm_mixer)); - return err; + return -ENOMEM; } hdspm->port_names_in = NULL; diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 512434efcc31..805ab6e9a78f 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1377,8 +1377,9 @@ static int __devinit sis_chip_create(struct snd_card *card, if (rc) goto error_out_cleanup; - if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME, - sis)) { + rc = request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME, + sis); + if (rc) { dev_err(&pci->dev, "unable to allocate irq %d\n", sis->irq); goto error_out_cleanup; } diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index f5ceb6f282de..210cafe04890 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -143,7 +143,7 @@ static int __devexit snd_pmac_remove(struct platform_device *devptr) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_pmac_driver_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 1aa52eff526a..9b18b5243a56 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -1040,6 +1040,7 @@ static int __devinit snd_ps3_driver_probe(struct ps3_system_bus_device *dev) GFP_KERNEL); if (!the_card.null_buffer_start_vaddr) { pr_info("%s: nullbuffer alloc failed\n", __func__); + ret = -ENOMEM; goto clean_preallocate; } pr_debug("%s: null vaddr=%p dma=%#llx\n", __func__, diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c index 318c5ba5360f..dfb744381c42 100644 --- a/sound/soc/blackfin/bf6xx-sport.c +++ b/sound/soc/blackfin/bf6xx-sport.c @@ -413,7 +413,14 @@ EXPORT_SYMBOL(sport_create); void sport_delete(struct sport_device *sport) { + if (sport->tx_desc) + dma_free_coherent(NULL, sport->tx_desc_size, + sport->tx_desc, 0); + if (sport->rx_desc) + dma_free_coherent(NULL, sport->rx_desc_size, + sport->rx_desc, 0); sport_free_resource(sport); + kfree(sport); } EXPORT_SYMBOL(sport_delete); diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 3c795921c5f6..23b40186f9b8 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2406,6 +2406,10 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) /* Setup AB8500 according to board-settings */ pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent); + + /* Inform SoC Core that we have our own I/O arrangements. */ + codec->control_data = (void *)true; + status = ab8500_audio_setup_mics(codec, &pdata->codec->amics); if (status < 0) { pr_err("%s: Failed to setup mics (%d)!\n", __func__, status); diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 8c39dddd7d00..11b1b714b8b5 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -186,6 +186,7 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) printk(KERN_INFO "AD1980 SoC Audio Codec\n"); + codec->control_data = codec; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "ad1980: failed to register AC97 codec\n"); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 5c9cacaf2d52..1cf7a32d1b21 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -426,7 +426,7 @@ static const int arizona_44k1_bclk_rates[] = { 940800, 1411200, 1881600, - 2882400, + 2822400, 3763200, 5644800, 7526400, diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 6276e352125f..115a40301810 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -581,6 +581,8 @@ static int mc13783_probe(struct snd_soc_codec *codec) { struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + codec->control_data = priv->mc13xxx; + mc13xxx_lock(priv->mc13xxx); /* these are the reset values */ @@ -657,7 +659,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = { .id = MC13783_ID_STEREO_DAC, .playback = { .stream_name = "Playback", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = MC13783_FORMATS, @@ -668,7 +670,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = { .id = MC13783_ID_STEREO_CODEC, .capture = { .stream_name = "Capture", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = MC13783_RATES_RECORD, .formats = MC13783_FORMATS, @@ -690,14 +692,14 @@ static struct snd_soc_dai_driver mc13783_dai_sync[] = { .id = MC13783_ID_SYNC, .playback = { .stream_name = "Playback", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = MC13783_FORMATS, }, .capture = { .stream_name = "Capture", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = MC13783_RATES_RECORD, .formats = MC13783_FORMATS, diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 8af6a5245b18..df2f99d1d428 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -239,6 +239,7 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ + {"LINE_IN", NULL, "VAG_POWER"}, {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */ {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */ @@ -1357,8 +1358,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) if (ret) goto err; - snd_soc_dapm_new_widgets(&codec->dapm); - return 0; err: diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 982e437799a8..33c0f3d39c87 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -340,6 +340,7 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); + codec->control_data = codec; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) goto codec_err; diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 3fd5b29dc933..a3acb7a85f6a 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -702,7 +702,7 @@ static bool wm2000_readable_reg(struct device *dev, unsigned int reg) } static const struct regmap_config wm2000_regmap = { - .reg_bits = 8, + .reg_bits = 16, .val_bits = 8, .max_register = WM2000_REG_IF_CTL, diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 6537f16d383e..e33d327396ad 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -128,13 +128,9 @@ SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, 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), @@ -236,8 +232,6 @@ 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); @@ -349,10 +343,6 @@ 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), @@ -466,8 +456,6 @@ 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"), @@ -553,8 +541,6 @@ SND_SOC_DAPM_OUTPUT("SPKDAT1R"), { name, "EQ4", "EQ4" }, \ { name, "DRC1L", "DRC1L" }, \ { name, "DRC1R", "DRC1R" }, \ - { name, "DRC2L", "DRC2L" }, \ - { name, "DRC2R", "DRC2R" }, \ { name, "LHPF1", "LHPF1" }, \ { name, "LHPF2", "LHPF2" }, \ { name, "LHPF3", "LHPF3" }, \ @@ -639,6 +625,15 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "AIF2 Capture", NULL, "SYSCLK" }, { "AIF3 Capture", NULL, "SYSCLK" }, + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + { "IN3L PGA", NULL, "IN3L" }, + { "IN3R PGA", NULL, "IN3R" }, + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), @@ -675,8 +670,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { 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"), diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 8033f7065189..01ebbcc5c6a4 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -681,6 +681,18 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "AIF2 Capture", NULL, "SYSCLK" }, { "AIF3 Capture", NULL, "SYSCLK" }, + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + { "IN3L PGA", NULL, "IN3L" }, + { "IN3R PGA", NULL, "IN3R" }, + + { "IN4L PGA", NULL, "IN4L" }, + { "IN4R PGA", NULL, "IN4R" }, + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 0013afe48e66..dc4262eea4b7 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -100,7 +100,7 @@ static const struct reg_default wm8904_reg_defaults[] = { { 14, 0x0000 }, /* R14 - Power Management 2 */ { 15, 0x0000 }, /* R15 - Power Management 3 */ { 18, 0x0000 }, /* R18 - Power Management 6 */ - { 19, 0x945E }, /* R20 - Clock Rates 0 */ + { 20, 0x945E }, /* R20 - Clock Rates 0 */ { 21, 0x0C05 }, /* R21 - Clock Rates 1 */ { 22, 0x0006 }, /* R22 - Clock Rates 2 */ { 24, 0x0050 }, /* R24 - Audio Interface 0 */ diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index eaf65863ec21..ce6720073798 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2501,6 +2501,9 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec, /* VMID 2*250k */ snd_soc_update_bits(codec, WM8962_PWR_MGMT_1, WM8962_VMID_SEL_MASK, 0x100); + + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + msleep(100); break; case SND_SOC_BIAS_OFF: @@ -3730,21 +3733,6 @@ static int wm8962_runtime_resume(struct device *dev) regcache_sync(wm8962->regmap); - regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, - WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA, - WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA); - - /* Bias enable at 2*50k for ramp */ - regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, - WM8962_VMID_SEL_MASK | WM8962_BIAS_ENA, - WM8962_BIAS_ENA | 0x180); - - msleep(5); - - /* VMID back to 2x250k for standby */ - regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, - WM8962_VMID_SEL_MASK, 0x100); - return 0; } diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index bb62f4b3d563..6c9eeca85b95 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2649,7 +2649,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - bclk_rate = params_rate(params) * 2; + bclk_rate = params_rate(params) * 4; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: bclk_rate *= 16; @@ -3253,10 +3253,13 @@ static void wm8994_mic_work(struct work_struct *work) int ret; int report; + pm_runtime_get_sync(dev); + ret = regmap_read(regmap, WM8994_INTERRUPT_RAW_STATUS_2, ®); if (ret < 0) { dev_err(dev, "Failed to read microphone status: %d\n", ret); + pm_runtime_put(dev); return; } @@ -3299,6 +3302,8 @@ static void wm8994_mic_work(struct work_struct *work) snd_soc_jack_report(priv->micdet[1].jack, report, SND_JACK_HEADSET | SND_JACK_BTN_0); + + pm_runtime_put(dev); } static irqreturn_t wm8994_mic_irq(int irq, void *data) @@ -3421,12 +3426,15 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) int reg; bool present; + pm_runtime_get_sync(codec->dev); + mutex_lock(&wm8994->accdet_lock); reg = snd_soc_read(codec, WM1811_JACKDET_CTRL); if (reg < 0) { dev_err(codec->dev, "Failed to read jack status: %d\n", reg); mutex_unlock(&wm8994->accdet_lock); + pm_runtime_put(codec->dev); return IRQ_NONE; } @@ -3491,6 +3499,7 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) SND_JACK_MECHANICAL | SND_JACK_HEADSET | wm8994->btn_mask); + pm_runtime_put(codec->dev); return IRQ_HANDLED; } @@ -3602,6 +3611,8 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) if (!(snd_soc_read(codec, WM8958_MIC_DETECT_1) & WM8958_MICD_ENA)) return IRQ_HANDLED; + pm_runtime_get_sync(codec->dev); + /* We may occasionally read a detection without an impedence * range being provided - if that happens loop again. */ @@ -3612,6 +3623,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) dev_err(codec->dev, "Failed to read mic detect status: %d\n", reg); + pm_runtime_put(codec->dev); return IRQ_NONE; } @@ -3639,6 +3651,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) dev_warn(codec->dev, "Accessory detection with no callback\n"); out: + pm_runtime_put(codec->dev); return IRQ_HANDLED; } @@ -4025,6 +4038,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; case WM8958: if (wm8994->revision < 1) { + snd_soc_dapm_add_routes(dapm, wm8994_intercon, + ARRAY_SIZE(wm8994_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, ARRAY_SIZE(wm8994_revd_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon, diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 099e6ec32125..c6d2076a796b 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -148,7 +148,7 @@ SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1), SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1), SOC_ENUM("Capture Volume Steps", wm9712_enum[6]), -SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1), +SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 0), SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0), SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv), @@ -272,7 +272,7 @@ SOC_DAPM_ENUM("Route", wm9712_enum[9]); /* Mic select */ static const struct snd_kcontrol_new wm9712_mic_src_controls = -SOC_DAPM_ENUM("Route", wm9712_enum[7]); +SOC_DAPM_ENUM("Mic Source Select", wm9712_enum[7]); /* diff select */ static const struct snd_kcontrol_new wm9712_diff_sel_controls = @@ -291,7 +291,9 @@ SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0, &wm9712_capture_selectl_controls), SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0, &wm9712_capture_selectr_controls), -SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0, +SND_SOC_DAPM_MUX("Left Mic Select Source", SND_SOC_NOPM, 0, 0, + &wm9712_mic_src_controls), +SND_SOC_DAPM_MUX("Right Mic Select Source", SND_SOC_NOPM, 0, 0, &wm9712_mic_src_controls), SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0, &wm9712_diff_sel_controls), @@ -319,6 +321,7 @@ SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0), SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0), SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0), SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0), +SND_SOC_DAPM_PGA("Differential Mic", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1), SND_SOC_DAPM_OUTPUT("MONOOUT"), SND_SOC_DAPM_OUTPUT("HPOUTL"), @@ -379,6 +382,18 @@ static const struct snd_soc_dapm_route wm9712_audio_map[] = { {"Mic PGA", NULL, "MIC1"}, {"Mic PGA", NULL, "MIC2"}, + /* microphones */ + {"Differential Mic", NULL, "MIC1"}, + {"Differential Mic", NULL, "MIC2"}, + {"Left Mic Select Source", "Mic 1", "MIC1"}, + {"Left Mic Select Source", "Mic 2", "MIC2"}, + {"Left Mic Select Source", "Stereo", "MIC1"}, + {"Left Mic Select Source", "Differential", "Differential Mic"}, + {"Right Mic Select Source", "Mic 1", "MIC1"}, + {"Right Mic Select Source", "Mic 2", "MIC2"}, + {"Right Mic Select Source", "Stereo", "MIC2"}, + {"Right Mic Select Source", "Differential", "Differential Mic"}, + /* left capture selector */ {"Left Capture Select", "Mic", "MIC1"}, {"Left Capture Select", "Speaker Mixer", "Speaker Mixer"}, @@ -619,6 +634,7 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) { int ret = 0; + codec->control_data = codec; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9712: failed to register AC97 codec\n"); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 3eb19fb71d17..d0b8a3287a85 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1196,6 +1196,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) if (wm9713 == NULL) return -ENOMEM; snd_soc_codec_set_drvdata(codec, wm9713); + codec->control_data = wm9713; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 95441bfc8190..ce5e5cd254dd 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -380,14 +380,20 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (dev->txnumevt) /* enable FIFO */ + if (dev->txnumevt) { /* enable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + FIFO_ENABLE); mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } mcasp_start_tx(dev); } else { - if (dev->rxnumevt) /* enable FIFO */ + if (dev->rxnumevt) { /* enable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + FIFO_ENABLE); mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } mcasp_start_rx(dev); } } diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index fb21b17f17f5..199408ec4261 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -94,7 +94,7 @@ static int __devinit imx_sgtl5000_probe(struct platform_device *pdev) dev_err(&pdev->dev, "audmux internal port setup failed\n"); return ret; } - imx_audmux_v2_configure_port(ext_port, + ret = imx_audmux_v2_configure_port(ext_port, IMX_AUDMUX_V2_PTCR_SYN, IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); if (ret) { diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 28dd76c7cb1c..81d7728cf67f 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -380,13 +380,14 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver imx_ssi_dai = { .probe = imx_ssi_dai_probe, .playback = { - .channels_min = 1, + /* The SSI does not support monaural audio. */ + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index 99a997f19bb9..b6fa77678d97 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -10,7 +10,7 @@ menuconfig SND_MXS_SOC if SND_MXS_SOC config SND_SOC_MXS_SGTL5000 - tristate "SoC Audio support for i.MX boards with sgtl5000" + tristate "SoC Audio support for MXS boards with sgtl5000" depends on I2C select SND_SOC_SGTL5000 help diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index aba71bfa33b1..b3030718c228 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -394,9 +394,14 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + struct mxs_saif *master_saif; u32 scr, stat; int ret; + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; + /* mclk should already be set */ if (!saif->mclk && saif->mclk_in_use) { dev_err(cpu_dai->dev, "set mclk first\n"); @@ -420,6 +425,25 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream, return ret; } + /* prepare clk in hw_param, enable in trigger */ + clk_prepare(saif->clk); + if (saif != master_saif) { + /* + * Set an initial clock rate for the saif internal logic to work + * properly. This is important when working in EXTMASTER mode + * that uses the other saif's BITCLK&LRCLK but it still needs a + * basic clock which should be fast enough for the internal + * logic. + */ + clk_enable(saif->clk); + ret = clk_set_rate(saif->clk, 24000000); + clk_disable(saif->clk); + if (ret) + return ret; + + clk_prepare(master_saif->clk); + } + scr = __raw_readl(saif->base + SAIF_CTRL); scr &= ~BM_SAIF_CTRL_WORD_LENGTH; diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index 009533ab8d18..df65f98211ec 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -59,7 +59,7 @@ static int am3517evm_hw_params(struct snd_pcm_substream *substream, return ret; } - snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0, SND_SOC_CLOCK_IN); if (ret < 0) { printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_FSR_SRC_FSX\n"); diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 34835e8a9160..d33c48baaf71 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -745,7 +745,7 @@ int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) { const char *signal, *src; - if (mcbsp->pdata->mux_signal) + if (!mcbsp->pdata->mux_signal) return -EINVAL; switch (mux) { diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 1046083e90a0..acdd3ef14e08 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -820,3 +820,4 @@ module_platform_driver(asoc_mcbsp_driver); MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>"); MODULE_DESCRIPTION("OMAP I2S SoC Interface"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:omap-mcbsp"); diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 5a649da9122a..f0feb06615f8 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -441,3 +441,4 @@ module_platform_driver(omap_pcm_driver); MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>"); MODULE_DESCRIPTION("OMAP PCM DMA module"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:omap-pcm-audio"); diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index f3ebc38c10fe..b70964ea448c 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -34,9 +34,7 @@ static const struct snd_pcm_hardware dma_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, + SNDRV_PCM_INFO_MMAP_VALID, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U8 | @@ -248,15 +246,11 @@ static int dma_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: prtd->state |= ST_RUNNING; prtd->params->ops->trigger(prtd->params->ch); break; case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: prtd->state &= ~ST_RUNNING; prtd->params->ops->stop(prtd->params->ch); break; diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index b7b2a1f91425..89b064650f14 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -20,7 +20,7 @@ #include <sound/pcm_params.h> #include <plat/audio.h> -#include <plat/dma.h> +#include <mach/dma.h> #include "dma.h" #include "pcm.h" diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f219b2f7ee68..c501af6d8dbe 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -826,7 +826,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } if (!rtd->cpu_dai) { - dev_dbg(card->dev, "CPU DAI %s not registered\n", + dev_err(card->dev, "CPU DAI %s not registered\n", dai_link->cpu_dai_name); return -EPROBE_DEFER; } @@ -857,14 +857,14 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } if (!rtd->codec_dai) { - dev_dbg(card->dev, "CODEC DAI %s not registered\n", + dev_err(card->dev, "CODEC DAI %s not registered\n", dai_link->codec_dai_name); return -EPROBE_DEFER; } } if (!rtd->codec) { - dev_dbg(card->dev, "CODEC %s not registered\n", + dev_err(card->dev, "CODEC %s not registered\n", dai_link->codec_name); return -EPROBE_DEFER; } @@ -888,7 +888,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) rtd->platform = platform; } if (!rtd->platform) { - dev_dbg(card->dev, "platform %s not registered\n", + dev_err(card->dev, "platform %s not registered\n", dai_link->platform_name); return -EPROBE_DEFER; } @@ -1096,7 +1096,7 @@ 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) + if (!codec->write && dev_get_regmap(codec->dev, NULL)) snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (driver->controls) @@ -1481,6 +1481,8 @@ static int soc_check_aux_dev(struct snd_soc_card *card, int num) return 0; } + dev_err(card->dev, "%s not registered\n", aux_dev->codec_name); + return -EPROBE_DEFER; } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dd7c49fafd75..f90139b5f50d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -291,8 +291,11 @@ 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 + else + dapm->bias_level = level; + } else if (!card || dapm != &card->dapm) { dapm->bias_level = level; + } if (ret != 0) goto out; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 7f8b3b7428bb..0c172938b82a 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -103,7 +103,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) } /* Report before the DAPM sync to help users updating micbias status */ - blocking_notifier_call_chain(&jack->notifier, status, jack); + blocking_notifier_call_chain(&jack->notifier, jack->status, jack); snd_soc_dapm_sync(dapm); diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 97c2cac8e92c..8c7f23729446 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -138,7 +138,7 @@ static void spear_pcm_free(struct snd_pcm *pcm) continue; buf = &substream->dma_buffer; - if (!buf && !buf->area) + if (!buf || !buf->area) continue; dma_free_writecombine(pcm->card->dev, buf->bytes, diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index d684df294c0c..76cb1b363b71 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -89,7 +89,6 @@ static struct snd_soc_jack_gpio tegra_alc5632_hp_jack_gpio = { .name = "Headset detection", .report = SND_JACK_HEADSET, .debounce_time = 150, - .invert = 1, }; static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { @@ -177,7 +176,7 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) } alc5632->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); - if (alc5632->gpio_hp_det == -ENODEV) + if (alc5632->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; ret = snd_soc_of_parse_card_name(card, "nvidia,model"); diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 5658bcec1931..8d6900c1ee47 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -334,11 +334,11 @@ static int tegra_pcm_hw_params(struct snd_pcm_substream *substream, 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; + slave_config.dst_maxburst = 4; } else { slave_config.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; slave_config.src_addr = dmap->addr; - slave_config.dst_maxburst = 0; + slave_config.src_maxburst = 4; } slave_config.slave_id = dmap->req_sel; diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 0c5bb33d258e..d4f14e492341 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -284,27 +284,27 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) } else if (np) { pdata->gpio_spkr_en = of_get_named_gpio(np, "nvidia,spkr-en-gpios", 0); - if (pdata->gpio_spkr_en == -ENODEV) + if (pdata->gpio_spkr_en == -EPROBE_DEFER) return -EPROBE_DEFER; pdata->gpio_hp_mute = of_get_named_gpio(np, "nvidia,hp-mute-gpios", 0); - if (pdata->gpio_hp_mute == -ENODEV) + if (pdata->gpio_hp_mute == -EPROBE_DEFER) return -EPROBE_DEFER; pdata->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); - if (pdata->gpio_hp_det == -ENODEV) + if (pdata->gpio_hp_det == -EPROBE_DEFER) 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) + if (pdata->gpio_int_mic_en == -EPROBE_DEFER) 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) + if (pdata->gpio_ext_mic_en == -EPROBE_DEFER) return -EPROBE_DEFER; } diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 62ac0285bfaf..057e28ef770e 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -21,7 +21,7 @@ #include <linux/mfd/dbx500-prcmu.h> #include <mach/hardware.h> -#include <mach/board-mop500-msp.h> +#include <mach/msp.h> #include <sound/soc.h> #include <sound/soc-dai.h> diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index ee14d2dac2f5..eb85113d472a 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -19,7 +19,7 @@ #include <linux/slab.h> #include <mach/hardware.h> -#include <mach/board-mop500-msp.h> +#include <mach/msp.h> #include <sound/soc.h> @@ -663,7 +663,6 @@ 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; @@ -685,15 +684,14 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, if (res == NULL) { dev_err(&pdev->dev, "%s: ERROR: Unable to get resource!\n", __func__); - ret = -ENOMEM; - goto err_res; + return -ENOMEM; } - msp->registers = ioremap(res->start, (res->end - res->start + 1)); + msp->registers = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); if (msp->registers == NULL) { dev_err(&pdev->dev, "%s: ERROR: ioremap failed!\n", __func__); - ret = -ENOMEM; - goto err_res; + return -ENOMEM; } msp->msp_state = MSP_STATE_IDLE; @@ -705,7 +703,7 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, dev_err(&pdev->dev, "%s: ERROR: Failed to allocate I2S-controller!\n", __func__); - goto err_i2s_cont; + return -ENOMEM; } i2s_cont->dev.parent = &pdev->dev; i2s_cont->data = (void *)msp; @@ -716,14 +714,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, 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, @@ -732,11 +722,6 @@ void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev, 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 index 7f71b4a0d4bc..2d9136da9865 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -17,7 +17,7 @@ #include <linux/platform_device.h> -#include <mach/board-mop500-msp.h> +#include <mach/msp.h> #define MSP_INPUT_FREQ_APB 48000000 diff --git a/sound/sound_firmware.c b/sound/sound_firmware.c index 7e96249536b4..37711a5d0d6b 100644 --- a/sound/sound_firmware.c +++ b/sound/sound_firmware.c @@ -23,14 +23,14 @@ static int do_mod_firmware_load(const char *fn, char **fp) if (l <= 0 || l > 131072) { printk(KERN_INFO "Invalid firmware '%s'\n", fn); - filp_close(filp, current->files); + filp_close(filp, NULL); return 0; } dp = vmalloc(l); if (dp == NULL) { printk(KERN_INFO "Out of memory loading '%s'.\n", fn); - filp_close(filp, current->files); + filp_close(filp, NULL); return 0; } pos = 0; @@ -38,10 +38,10 @@ static int do_mod_firmware_load(const char *fn, char **fp) { printk(KERN_INFO "Failed to read '%s'.\n", fn); vfree(dp); - filp_close(filp, current->files); + filp_close(filp, NULL); return 0; } - filp_close(filp, current->files); + filp_close(filp, NULL); *fp = dp; return (int) l; } diff --git a/sound/usb/card.c b/sound/usb/card.c index d5b5c3388e28..4a469f0cb6d4 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -553,7 +553,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, struct snd_usb_audio *chip) { struct snd_card *card; - struct list_head *p; + struct list_head *p, *n; if (chip == (void *)-1L) return; @@ -570,7 +570,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, snd_usb_stream_disconnect(p); } /* release the endpoint resources */ - list_for_each(p, &chip->ep_list) { + list_for_each_safe(p, n, &chip->ep_list) { snd_usb_endpoint_free(p); } /* release the midi resources */ diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 379baad3d5ad..5e634a2eb282 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -111,7 +111,8 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id) return 0; /* If a clock source can't tell us whether it's valid, we assume it is */ - if (!uac2_control_is_readable(cs_desc->bmControls, UAC2_CS_CONTROL_CLOCK_VALID)) + if (!uac2_control_is_readable(cs_desc->bmControls, + UAC2_CS_CONTROL_CLOCK_VALID - 1)) return 1; err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 0f647d22cb4a..060dccb9ec75 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -141,7 +141,7 @@ int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep) * * For implicit feedback, next_packet_size() is unused. */ -static int next_packet_size(struct snd_usb_endpoint *ep) +int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) { unsigned long flags; int ret; @@ -177,15 +177,6 @@ static void retire_inbound_urb(struct snd_usb_endpoint *ep, 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. */ @@ -206,7 +197,13 @@ static void prepare_outbound_urb(struct snd_usb_endpoint *ep, /* no data provider, so send silence */ unsigned int offs = 0; for (i = 0; i < ctx->packets; ++i) { - int counts = ctx->packet_size[i]; + int counts; + + if (ctx->packet_size[i]) + counts = ctx->packet_size[i]; + else + counts = snd_usb_endpoint_next_packet_size(ep); + urb->iso_frame_desc[i].offset = offs * ep->stride; urb->iso_frame_desc[i].length = counts * ep->stride; offs += counts; @@ -370,7 +367,6 @@ static void snd_complete_urb(struct urb *urb) goto exit_clear; } - prepare_outbound_urb_sizes(ep, ctx); prepare_outbound_urb(ep, ctx); } else { retire_inbound_urb(ep, ctx); @@ -799,7 +795,9 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, /** * snd_usb_endpoint_start: start an snd_usb_endpoint * - * @ep: the endpoint to start + * @ep: the endpoint to start + * @can_sleep: flag indicating whether the operation is executed in + * non-atomic context * * 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 @@ -809,7 +807,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, * * Returns an error if the URB submission failed, 0 in all other cases. */ -int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep) { int err; unsigned int i; @@ -822,8 +820,9 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) return 0; /* just to be sure */ - deactivate_urbs(ep, 0, 1); - wait_clear_urbs(ep); + deactivate_urbs(ep, 0, can_sleep); + if (can_sleep) + wait_clear_urbs(ep); ep->active_mask = 0; ep->unlink_mask = 0; @@ -854,7 +853,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) 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); diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index ee2723fb174f..cbbbdf226d66 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -13,7 +13,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, struct audioformat *fmt, struct snd_usb_endpoint *sync_ep); -int snd_usb_endpoint_start(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep); 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); @@ -21,6 +21,7 @@ int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct list_head *head); int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep); void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, struct snd_usb_endpoint *sender, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a1298f379428..f782ce19bf5a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -212,7 +212,7 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, } } -static int start_endpoints(struct snd_usb_substream *subs) +static int start_endpoints(struct snd_usb_substream *subs, int can_sleep) { int err; @@ -225,7 +225,7 @@ static int start_endpoints(struct snd_usb_substream *subs) snd_printdd(KERN_DEBUG "Starting data EP @%p\n", ep); ep->data_subs = subs; - err = snd_usb_endpoint_start(ep); + err = snd_usb_endpoint_start(ep, can_sleep); if (err < 0) { clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags); return err; @@ -236,10 +236,25 @@ static int start_endpoints(struct snd_usb_substream *subs) !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) { struct snd_usb_endpoint *ep = subs->sync_endpoint; + if (subs->data_endpoint->iface != subs->sync_endpoint->iface || + subs->data_endpoint->alt_idx != subs->sync_endpoint->alt_idx) { + err = usb_set_interface(subs->dev, + subs->sync_endpoint->iface, + subs->sync_endpoint->alt_idx); + if (err < 0) { + snd_printk(KERN_ERR + "%d:%d:%d: cannot set interface (%d)\n", + subs->dev->devnum, + subs->sync_endpoint->iface, + subs->sync_endpoint->alt_idx, err); + return -EIO; + } + } + snd_printdd(KERN_DEBUG "Starting sync EP @%p\n", ep); ep->sync_slave = subs->data_endpoint; - err = snd_usb_endpoint_start(ep); + err = snd_usb_endpoint_start(ep, can_sleep); if (err < 0) { clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); return err; @@ -547,7 +562,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) /* 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 start_endpoints(subs, 1); return 0; } @@ -1029,6 +1044,7 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, struct urb *urb) { struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; + struct snd_usb_endpoint *ep = subs->data_endpoint; struct snd_urb_ctx *ctx = urb->context; unsigned int counts, frames, bytes; int i, stride, period_elapsed = 0; @@ -1040,7 +1056,11 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, urb->number_of_packets = 0; spin_lock_irqsave(&subs->lock, flags); for (i = 0; i < ctx->packets; i++) { - counts = ctx->packet_size[i]; + if (ctx->packet_size[i]) + counts = ctx->packet_size[i]; + else + counts = snd_usb_endpoint_next_packet_size(ep); + /* set up descriptor */ urb->iso_frame_desc[i].offset = frames * stride; urb->iso_frame_desc[i].length = counts * stride; @@ -1091,7 +1111,16 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, 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) @@ -1109,12 +1138,32 @@ static void retire_playback_urb(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; int stride = runtime->frame_bits >> 3; int processed = urb->transfer_buffer_length / stride; + int est_delay; + + /* ignore the delay accounting when procssed=0 is given, i.e. + * silent payloads are procssed before handling the actual data + */ + if (!processed) + return; spin_lock_irqsave(&subs->lock, flags); - if (processed > runtime->delay) - runtime->delay = 0; + 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 - runtime->delay -= processed; + 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); } @@ -1172,7 +1221,7 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream switch (cmd) { case SNDRV_PCM_TRIGGER_START: - err = start_endpoints(subs); + err = start_endpoints(subs, 0); if (err < 0) return err; |