From 5163c1eede8e9073e5b6bf1a988ed07d35820343 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 9 Feb 2015 13:26:01 +0200 Subject: ASoC: omap: Kconfig: Support for omap5-uevm analog audio The analog audio is supported via omap-abe-twl6040 machine driver on omap5-uevm. Update the Kconfig file to reflect this and select the Palmas clock driver which is providing the 32K clock for audio on omap5-uevm. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index a2cd3486ac55..e7c78b0406b5 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -100,17 +100,19 @@ config SND_OMAP_SOC_OMAP_TWL4030 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || COMPILE_TEST) + depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || SOC_OMAP5 || COMPILE_TEST) select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 select SND_SOC_DMIC + select COMMON_CLK_PALMAS if SOC_OMAP5 help Say Y if you want to add support for SoC audio on OMAP boards using ABE and twl6040 codec. This driver currently supports: - SDP4430/Blaze boards - PandaBoard (4430) - PandaBoardES (4460) + - omap5-uevm (5432) config SND_OMAP_SOC_OMAP3_PANDORA tristate "SoC Audio support for OMAP3 Pandora" -- cgit v1.2.3-59-g8ed1b From b6d6c6e95ff0e78f9b8393e6b9f25d5a4341ae1a Mon Sep 17 00:00:00 2001 From: Peter Rosin Date: Mon, 9 Feb 2015 16:08:25 +0100 Subject: ASoC: atmel_ssc_dai: Allow more rates When the SSC acts as BCK master, use a ratnum rule to limit the rate instead of only doing the standard rates. When the SSC acts as BCK slave, allow any BCK frequency up to the SSC master clock, divided by either of 2, 3 or 6. Put a cap at 384kHz. Who's /ever/ going to need more than that? The divider of 2, 3 or 6 is selected based on the Serial Clock Ratio Considerations section from the SSC documentation: The Transmitter and the Receiver can be programmed to operate with the clock signals provided on either the TK or RK pins. This allows the SSC to support many slave-mode data transfers. In this case, the maximum clock speed allowed on the RK pin is: - Peripheral clock divided by 2 if Receiver Frame Synchro is input - Peripheral clock divided by 3 if Receiver Frame Synchro is output In addition, the maximum clock speed allowed on the TK pin is: - Peripheral clock divided by 6 if Transmit Frame Synchro is input - Peripheral clock divided by 2 if Transmit Frame Synchro is output Signed-off-by: Peter Rosin Acked-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_ssc_dai.c | 111 ++++++++++++++++++++++++++++++++++++++-- sound/soc/atmel/atmel_ssc_dai.h | 1 + 2 files changed, 108 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 379ac2a6ab16..6b8e64899205 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -187,6 +187,94 @@ static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } +/* + * When the bit clock is input, limit the maximum rate according to the + * Serial Clock Ratio Considerations section from the SSC documentation: + * + * The Transmitter and the Receiver can be programmed to operate + * with the clock signals provided on either the TK or RK pins. + * This allows the SSC to support many slave-mode data transfers. + * In this case, the maximum clock speed allowed on the RK pin is: + * - Peripheral clock divided by 2 if Receiver Frame Synchro is input + * - Peripheral clock divided by 3 if Receiver Frame Synchro is output + * In addition, the maximum clock speed allowed on the TK pin is: + * - Peripheral clock divided by 6 if Transmit Frame Synchro is input + * - Peripheral clock divided by 2 if Transmit Frame Synchro is output + * + * When the bit clock is output, limit the rate according to the + * SSC divider restrictions. + */ +static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct atmel_ssc_info *ssc_p = rule->private; + struct ssc_device *ssc = ssc_p->ssc; + struct snd_interval *i = hw_param_interval(params, rule->var); + struct snd_interval t; + struct snd_ratnum r = { + .den_min = 1, + .den_max = 4095, + .den_step = 1, + }; + unsigned int num = 0, den = 0; + int frame_size; + int mck_div = 2; + int ret; + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) + return frame_size; + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFS: + if ((ssc_p->dir_mask & SSC_DIR_MASK_CAPTURE) + && ssc->clk_from_rk_pin) + /* Receiver Frame Synchro (i.e. capture) + * is output (format is _CFS) and the RK pin + * is used for input (format is _CBM_). + */ + mck_div = 3; + break; + + case SND_SOC_DAIFMT_CBM_CFM: + if ((ssc_p->dir_mask & SSC_DIR_MASK_PLAYBACK) + && !ssc->clk_from_rk_pin) + /* Transmit Frame Synchro (i.e. playback) + * is input (format is _CFM) and the TK pin + * is used for input (format _CBM_ but not + * using the RK pin). + */ + mck_div = 6; + break; + } + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + r.num = ssc_p->mck_rate / mck_div / frame_size; + + ret = snd_interval_ratnum(i, 1, &r, &num, &den); + if (ret >= 0 && den && rule->var == SNDRV_PCM_HW_PARAM_RATE) { + params->rate_num = num; + params->rate_den = den; + } + break; + + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBM_CFM: + t.min = 8000; + t.max = ssc_p->mck_rate / mck_div / frame_size; + t.openmin = t.openmax = 0; + t.integer = 0; + ret = snd_interval_refine(i, &t); + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} /*-------------------------------------------------------------------------*\ * DAI functions @@ -200,6 +288,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; struct atmel_pcm_dma_params *dma_params; int dir, dir_mask; + int ret; pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", ssc_readl(ssc_p->ssc->regs, SR)); @@ -207,6 +296,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, /* Enable PMC peripheral clock for this SSC */ pr_debug("atmel_ssc_dai: Starting clock\n"); clk_enable(ssc_p->ssc->clk); + ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk); /* Reset the SSC to keep it at a clean status */ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); @@ -219,6 +309,17 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, dir_mask = SSC_DIR_MASK_CAPTURE; } + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + atmel_ssc_hw_rule_rate, + ssc_p, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret < 0) { + dev_err(dai->dev, "Failed to specify rate rule: %d\n", ret); + return ret; + } + dma_params = &ssc_dma_params[dai->id][dir]; dma_params->ssc = ssc_p->ssc; dma_params->substream = substream; @@ -783,8 +884,6 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) # define atmel_ssc_resume NULL #endif /* CONFIG_PM */ -#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) - #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -804,12 +903,16 @@ static struct snd_soc_dai_driver atmel_ssc_dai = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .capture = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .ops = &atmel_ssc_dai_ops, }; diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index b1f08d511495..80b153857a88 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -115,6 +115,7 @@ struct atmel_ssc_info { unsigned short rcmr_period; struct atmel_pcm_dma_params *dma_params[2]; struct atmel_ssc_state ssc_state; + unsigned long mck_rate; }; int atmel_ssc_set_audio(int ssc_id); -- cgit v1.2.3-59-g8ed1b From 226e2f1b0bb4a5f724dd119c1eeb8b8e89e87fab Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 12 Feb 2015 16:41:26 +0200 Subject: ASoC: davinci-mcasp: Add support for CBS_CFM mode Support for setups where codec is bitclock slave and frame master. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index de3b155a5011..031c1fb44ae7 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -441,6 +441,18 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); mcasp->bclk_master = 1; break; + case SND_SOC_DAIFMT_CBS_CFM: + /* codec is clock slave and frame master */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); + mcasp->bclk_master = 1; + break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); -- cgit v1.2.3-59-g8ed1b From 7371bd1f4aeb4e1c44b8c1ba8e36ebba4250b59c Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 17 Feb 2015 13:59:26 +0800 Subject: ASoC: rt5670: Add disabled item in dmic pin enum Currently, we will configure dmic related pin definition if pdata.dmic_en is true. However, there is no disable option in the enum. So, any dmic is used, all 3 dmic related pins will be configured. It may cause unexpected pin definition. This patch adds a disable item for each dmic enum and take it as default. So the driver will not set the pin configuration if we don't set dmicx_data_pin in platform data. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index d11b9c207e26..84857bdaa78b 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1967,17 +1967,20 @@ enum { }; enum { + RT5670_DMIC1_DISABLED, RT5670_DMIC_DATA_GPIO6, RT5670_DMIC_DATA_IN2P, RT5670_DMIC_DATA_GPIO7, }; enum { + RT5670_DMIC2_DISABLED, RT5670_DMIC_DATA_GPIO8, RT5670_DMIC_DATA_IN3N, }; enum { + RT5670_DMIC3_DISABLED, RT5670_DMIC_DATA_GPIO9, RT5670_DMIC_DATA_GPIO10, RT5670_DMIC_DATA_GPIO5, -- cgit v1.2.3-59-g8ed1b From 65d17a9ce9f24a3aaf7d614251fdcc1b2121765f Mon Sep 17 00:00:00 2001 From: Nikesh Oswal Date: Mon, 16 Feb 2015 15:25:48 +0000 Subject: ASoC: wm_adsp: Ensure DSP controls are always persistent Currently DSP controls are persistent (across DSP On/Off) only if they were set whilst the DSP is off. This change makes the controls persistent irrespective of when they are set. Signed-off-by: Nikesh Oswal Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 720d6e852986..14414ea23b55 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -420,10 +420,9 @@ static int wm_coeff_put(struct snd_kcontrol *kcontrol, memcpy(ctl->cache, p, ctl->len); - if (!ctl->enabled) { - ctl->set = 1; + ctl->set = 1; + if (!ctl->enabled) return 0; - } return wm_coeff_write_control(kcontrol, p, ctl->len); } -- cgit v1.2.3-59-g8ed1b From be951017453cba2f3eb789413f697b8f14393eec Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 16 Feb 2015 15:25:49 +0000 Subject: ASoC: wm_adsp: Improve round to next 4-byte boundary Whilst the existing code does correctly round to the next 4-byte boundary it does so rather inefficiently. This patch changes the rounding to be simpler and more efficient. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 14414ea23b55..e625cedb0fa9 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1184,7 +1184,6 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) int ret, pos, blocks, type, offset, reg; char *file; struct wm_adsp_buf *buf; - int tmp; file = kzalloc(PAGE_SIZE, GFP_KERNEL); if (file == NULL) @@ -1334,12 +1333,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } } - tmp = le32_to_cpu(blk->len) % 4; - if (tmp) - pos += le32_to_cpu(blk->len) + (4 - tmp) + sizeof(*blk); - else - pos += le32_to_cpu(blk->len) + sizeof(*blk); - + pos += (le32_to_cpu(blk->len) + sizeof(*blk) + 3) & ~0x03; blocks++; } -- cgit v1.2.3-59-g8ed1b From 7ff5eabce4231d199dadc14c23f14a6619f926c0 Mon Sep 17 00:00:00 2001 From: Kenneth Westfield Date: Tue, 17 Feb 2015 00:53:12 -0800 Subject: ASoC: max98357a: Remove use of DRV_NAME Remove use of DRV_NAME define. Signed-off-by: Kenneth Westfield Signed-off-by: Mark Brown --- sound/soc/codecs/max98357a.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index e9e6efbc21dd..4ee23fbc4e12 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -26,8 +26,6 @@ #include #include -#define DRV_NAME "max98357a" - static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -87,9 +85,9 @@ static struct snd_soc_dai_ops max98357a_dai_ops = { }; static struct snd_soc_dai_driver max98357a_dai_driver = { - .name = DRV_NAME, + .name = "max98357a", .playback = { - .stream_name = DRV_NAME "-playback", + .stream_name = "max98357a-playback", .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32, @@ -127,7 +125,7 @@ static int max98357a_platform_remove(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id max98357a_device_id[] = { - { .compatible = "maxim," DRV_NAME, }, + { .compatible = "maxim,max98357a" }, {} }; MODULE_DEVICE_TABLE(of, max98357a_device_id); @@ -135,7 +133,7 @@ MODULE_DEVICE_TABLE(of, max98357a_device_id); static struct platform_driver max98357a_platform_driver = { .driver = { - .name = DRV_NAME, + .name = "max98357a", .of_match_table = of_match_ptr(max98357a_device_id), }, .probe = max98357a_platform_probe, @@ -145,4 +143,3 @@ module_platform_driver(max98357a_platform_driver); MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:" DRV_NAME); -- cgit v1.2.3-59-g8ed1b From 369a9f5f397fe3258ab937ec7a9c2229d0b1a015 Mon Sep 17 00:00:00 2001 From: "Fang, Yang A" Date: Mon, 9 Feb 2015 00:18:12 -0800 Subject: ASoC: Intel: fix machine driver warnings this patch will fix below sparse warnings warning: incorrect type in argument 2 (different base types) expected unsigned int [unsigned] val got restricted snd_pcm_format_t [usertype] sound/soc/intel/haswell.c:61:37 sound/soc/intel/broadwell.c:115:37: sound/soc/intel/bytcr_dpcm_rt5640.c:118:37: sound/soc/intel/cht_bsw_rt5672.c:183:37: sound/soc/intel/cht_bsw_rt5645.c:208:37: Signed-off-by: Fang, Yang A Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/intel/broadwell.c | 4 +--- sound/soc/intel/bytcr_dpcm_rt5640.c | 4 +--- sound/soc/intel/cht_bsw_rt5645.c | 4 +--- sound/soc/intel/cht_bsw_rt5672.c | 4 +--- sound/soc/intel/haswell.c | 4 +--- 5 files changed, 5 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 9cf7d01479ad..fba2ef5dac42 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -110,9 +110,7 @@ static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP0 to 16 bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S16_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/bytcr_dpcm_rt5640.c b/sound/soc/intel/bytcr_dpcm_rt5640.c index 59308629043e..3b262d01c1b3 100644 --- a/sound/soc/intel/bytcr_dpcm_rt5640.c +++ b/sound/soc/intel/bytcr_dpcm_rt5640.c @@ -113,9 +113,7 @@ static int byt_codec_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP2 to 24-bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S24_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/cht_bsw_rt5645.c index bd29617a9ab9..dd935255a020 100644 --- a/sound/soc/intel/cht_bsw_rt5645.c +++ b/sound/soc/intel/cht_bsw_rt5645.c @@ -203,9 +203,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP2 to 24-bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S24_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index ff016621583a..c56f9dfe2129 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -178,9 +178,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP2 to 24-bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S24_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/haswell.c b/sound/soc/intel/haswell.c index 35edf51a52aa..00fddd3f5dfb 100644 --- a/sound/soc/intel/haswell.c +++ b/sound/soc/intel/haswell.c @@ -56,9 +56,7 @@ static int haswell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP0 to 16 bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S16_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); return 0; } -- cgit v1.2.3-59-g8ed1b From 48c7699fb2c799d084ce490bceea14fe04ad12a1 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:53 +0530 Subject: ASoC: core: allow pcms to be registered as nonatomic ALSA core with commit 257f8cce5d40 - "ALSA: pcm: Allow nonatomic trigger operations" allows trigger ops to implemented as nonatomic. For ASoC, we can specify this in dailinks and is updated while snd_pcm is created Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Cc: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +++ sound/soc/soc-pcm.c | 1 + 2 files changed, 4 insertions(+) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade195628..76bc944dcb5c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -954,6 +954,9 @@ struct snd_soc_dai_link { unsigned int symmetric_channels:1; unsigned int symmetric_samplebits:1; + /* Mark this pcm with non atomic ops */ + bool nonatomic; + /* Do not create a PCM for this DAI link (Backend link) */ unsigned int no_pcm:1; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6b0136e7cb88..6e3781e88f9a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2511,6 +2511,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) /* DAPM dai link stream work */ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + pcm->nonatomic = rtd->dai_link->nonatomic; rtd->pcm = pcm; pcm->private_data = rtd; -- cgit v1.2.3-59-g8ed1b From 76ca1c2cd8fc0b8764c6360263e2fbca43495ab2 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:54 +0530 Subject: ASoC: Intel: mark cht machine driver with nonatomic trigger The DSP messages are sent with nonatomic context, which include trigger messages, so mark the driver as nonatomic Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/cht_bsw_rt5672.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index c56f9dfe2129..a5098d6f988b 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -216,6 +216,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .codec_name = "snd-soc-dummy", .platform_name = "sst-mfld-platform", .ignore_suspend = 1, + .nonatomic = true, .dynamic = 1, .dpcm_playback = 1, .dpcm_capture = 1, @@ -238,6 +239,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, + .nonatomic = true, .codec_dai_name = "rt5670-aif1", .codec_name = "i2c-10EC5670:00", .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF -- cgit v1.2.3-59-g8ed1b From 7b9ca9d7e561ebdc93b43277eb69d20a0dc8f5cd Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:55 +0530 Subject: ASoC: Intel: update MMX ID to 3 The updated firmware expects the MMX ID to be used as 3, so update the driver as well Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-atom-controls.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h index dfebfdd5eb2a..daecc58f28af 100644 --- a/sound/soc/intel/sst-atom-controls.h +++ b/sound/soc/intel/sst-atom-controls.h @@ -150,7 +150,7 @@ enum sst_cmd_type { enum sst_task { SST_TASK_SBA = 1, - SST_TASK_MMX, + SST_TASK_MMX = 3, }; enum sst_type { -- cgit v1.2.3-59-g8ed1b From e0b87d476bc13fc55e7000a84cd1d87c8fdc1f2f Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:56 +0530 Subject: ASoC: Intel: add support for pause and resume in sst This adds missing pcm pause and resume ops in the driver Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_drv_interface.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c index 5f75ef3cdd22..5d56fcdd58d8 100644 --- a/sound/soc/intel/sst/sst_drv_interface.c +++ b/sound/soc/intel/sst/sst_drv_interface.c @@ -572,6 +572,35 @@ static int sst_stream_drop(struct device *dev, int str_id) return sst_drop_stream(ctx, str_id); } +static int sst_stream_pause(struct device *dev, int str_id) +{ + struct stream_info *str_info; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + if (ctx->sst_state != SST_FW_RUNNING) + return 0; + + str_info = get_stream_info(ctx, str_id); + if (!str_info) + return -EINVAL; + + return sst_pause_stream(ctx, str_id); +} + +static int sst_stream_resume(struct device *dev, int str_id) +{ + struct stream_info *str_info; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + if (ctx->sst_state != SST_FW_RUNNING) + return 0; + + str_info = get_stream_info(ctx, str_id); + if (!str_info) + return -EINVAL; + return sst_resume_stream(ctx, str_id); +} + static int sst_stream_init(struct device *dev, struct pcm_stream_info *str_info) { int str_id = 0; @@ -633,6 +662,8 @@ static struct sst_ops pcm_ops = { .stream_init = sst_stream_init, .stream_start = sst_stream_start, .stream_drop = sst_stream_drop, + .stream_pause = sst_stream_pause, + .stream_pause_release = sst_stream_resume, .stream_read_tstamp = sst_read_timestamp, .send_byte_stream = sst_send_byte_stream, .close = sst_close_pcm_stream, -- cgit v1.2.3-59-g8ed1b From fc9406ab9b4a9aac0ab9ad213993824cbe9c65ac Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:57 +0530 Subject: ASoC: Intel: add support for pcm stream suspend/resume The driver didn't implement support for pcm stream suspend and resume, so add it Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 7523cbef8780..ea0fa4b90bb0 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -594,11 +594,13 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, ret_val = stream->ops->stream_drop(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: dev_dbg(rtd->dev, "sst: in pause\n"); status = SST_PLATFORM_PAUSED; ret_val = stream->ops->stream_pause(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: dev_dbg(rtd->dev, "sst: in pause release\n"); status = SST_PLATFORM_RUNNING; ret_val = stream->ops->stream_pause_release(sst->dev, str_id); -- cgit v1.2.3-59-g8ed1b From 54e6beccc67129c474aad7578951112c6cf28e01 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:58 +0530 Subject: ASoC: Intel: add support for platform suspend This adds support for platform suspend and resume. We ensure all pcms are suspended by invoking snd_soc_suspend() and then stop the DSP Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 58 +++++++++++++++++++++++++++++++++ sound/soc/intel/sst-mfld-platform.h | 1 + 2 files changed, 59 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index ea0fa4b90bb0..2fbaf2c75d17 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -667,6 +667,9 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) static int sst_soc_probe(struct snd_soc_platform *platform) { + struct sst_data *drv = dev_get_drvdata(platform->dev); + + drv->soc_card = platform->component.card; return sst_dsp_init_v2_dpcm(platform); } @@ -729,9 +732,64 @@ static int sst_platform_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP + +static int sst_soc_prepare(struct device *dev) +{ + struct sst_data *drv = dev_get_drvdata(dev); + int i; + + /* suspend all pcms first */ + snd_soc_suspend(drv->soc_card->dev); + snd_soc_poweroff(drv->soc_card->dev); + + /* set the SSPs to idle */ + for (i = 0; i < drv->soc_card->num_rtd; i++) { + struct snd_soc_dai *dai = drv->soc_card->rtd[i].cpu_dai; + + if (dai->active) { + send_ssp_cmd(dai, dai->name, 0); + sst_handle_vb_timer(dai, false); + } + } + + return 0; +} + +static void sst_soc_complete(struct device *dev) +{ + struct sst_data *drv = dev_get_drvdata(dev); + int i; + + /* restart SSPs */ + for (i = 0; i < drv->soc_card->num_rtd; i++) { + struct snd_soc_dai *dai = drv->soc_card->rtd[i].cpu_dai; + + if (dai->active) { + sst_handle_vb_timer(dai, true); + send_ssp_cmd(dai, dai->name, 1); + } + } + snd_soc_resume(drv->soc_card->dev); +} + +#else + +#define sst_soc_prepare NULL +#define sst_soc_complete NULL + +#endif + + +static const struct dev_pm_ops sst_platform_pm = { + .prepare = sst_soc_prepare, + .complete = sst_soc_complete, +}; + static struct platform_driver sst_platform_driver = { .driver = { .name = "sst-mfld-platform", + .pm = &sst_platform_pm, }, .probe = sst_platform_probe, .remove = sst_platform_remove, diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 79c8d1246a8f..9094314be2b0 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -174,6 +174,7 @@ struct sst_data { struct sst_platform_data *pdata; struct snd_sst_bytes_v2 *byte_stream; struct mutex lock; + struct snd_soc_card *soc_card; }; int sst_register_dsp(struct sst_device *sst); int sst_unregister_dsp(struct sst_device *sst); -- cgit v1.2.3-59-g8ed1b From 5c88b4e91d3b6a3d701d7b134fa945e6309e7068 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 12 Feb 2015 10:00:01 +0530 Subject: ASoC: Intel: Add memcpy32_fromio as well Export 32-bit version of memcpy for use in suspend/resume. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.h | 3 +++ sound/soc/intel/sst/sst_loader.c | 10 +++++++++- 2 files changed, 12 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index 562bc483d6b7..f793780a50a2 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -544,4 +544,7 @@ int sst_alloc_drv_context(struct intel_sst_drv **ctx, int sst_context_init(struct intel_sst_drv *ctx); void sst_context_cleanup(struct intel_sst_drv *ctx); void sst_configure_runtime_pm(struct intel_sst_drv *ctx); +void memcpy32_toio(void __iomem *dst, const void *src, int count); +void memcpy32_fromio(void *dst, const void __iomem *src, int count); + #endif diff --git a/sound/soc/intel/sst/sst_loader.c b/sound/soc/intel/sst/sst_loader.c index 7888cd707853..e88907ae8b15 100644 --- a/sound/soc/intel/sst/sst_loader.c +++ b/sound/soc/intel/sst/sst_loader.c @@ -39,7 +39,15 @@ #include "sst.h" #include "../sst-dsp.h" -static inline void memcpy32_toio(void __iomem *dst, const void *src, int count) +void memcpy32_toio(void __iomem *dst, const void *src, int count) +{ + /* __iowrite32_copy uses 32-bit count values so divide by 4 for + * right count in words + */ + __iowrite32_copy(dst, src, count/4); +} + +void memcpy32_fromio(void *dst, const void __iomem *src, int count) { /* __iowrite32_copy uses 32-bit count values so divide by 4 for * right count in words -- cgit v1.2.3-59-g8ed1b From 4a8448d4289d7210053a43f9f21e42929beb159b Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 24 Feb 2015 11:39:44 +0530 Subject: ASoC: Intel: add pm support in sst ipc driver This adds support for system pm support. We need to save the dsp memory which gets lost on suspend and restore that on resume Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 128 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst/sst.h | 9 ++++ 2 files changed, 137 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 8a8d56a146e7..8f938112a01f 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -415,6 +415,83 @@ static int intel_sst_runtime_suspend(struct device *dev) return ret; } +static int intel_sst_suspend(struct device *dev) +{ + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + struct sst_fw_save *fw_save; + int i, ret = 0; + + /* check first if we are already in SW reset */ + if (ctx->sst_state == SST_RESET) + return 0; + + /* + * check if any stream is active and running + * they should already by suspend by soc_suspend + */ + for (i = 1; i <= ctx->info.max_streams; i++) { + struct stream_info *stream = &ctx->streams[i]; + + if (stream->status == STREAM_RUNNING) { + dev_err(dev, "stream %d is running, cant susupend, abort\n", i); + return -EBUSY; + } + } + synchronize_irq(ctx->irq_num); + flush_workqueue(ctx->post_msg_wq); + + /* Move the SST state to Reset */ + sst_set_fw_state_locked(ctx, SST_RESET); + + /* tell DSP we are suspending */ + if (ctx->ops->save_dsp_context(ctx)) + return -EBUSY; + + /* save the memories */ + fw_save = kzalloc(sizeof(*fw_save), GFP_KERNEL); + if (!fw_save) + return -ENOMEM; + fw_save->iram = kzalloc(ctx->iram_end - ctx->iram_base, GFP_KERNEL); + if (!fw_save->iram) { + ret = -ENOMEM; + goto iram; + } + fw_save->dram = kzalloc(ctx->dram_end - ctx->dram_base, GFP_KERNEL); + if (!fw_save->dram) { + ret = -ENOMEM; + goto dram; + } + fw_save->sram = kzalloc(SST_MAILBOX_SIZE, GFP_KERNEL); + if (!fw_save->sram) { + ret = -ENOMEM; + goto sram; + } + + fw_save->ddr = kzalloc(ctx->ddr_end - ctx->ddr_base, GFP_KERNEL); + if (!fw_save->ddr) { + ret = -ENOMEM; + goto ddr; + } + + memcpy32_fromio(fw_save->iram, ctx->iram, ctx->iram_end - ctx->iram_base); + memcpy32_fromio(fw_save->dram, ctx->dram, ctx->dram_end - ctx->dram_base); + memcpy32_fromio(fw_save->sram, ctx->mailbox, SST_MAILBOX_SIZE); + memcpy32_fromio(fw_save->ddr, ctx->ddr, ctx->ddr_end - ctx->ddr_base); + + ctx->fw_save = fw_save; + ctx->ops->reset(ctx); + return 0; +ddr: + kfree(fw_save->sram); +sram: + kfree(fw_save->dram); +dram: + kfree(fw_save->iram); +iram: + kfree(fw_save); + return ret; +} + static int intel_sst_runtime_resume(struct device *dev) { int ret = 0; @@ -430,7 +507,58 @@ static int intel_sst_runtime_resume(struct device *dev) return ret; } +static int intel_sst_resume(struct device *dev) +{ + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + struct sst_fw_save *fw_save = ctx->fw_save; + int ret = 0; + struct sst_block *block; + + if (!fw_save) + return intel_sst_runtime_resume(dev); + + sst_set_fw_state_locked(ctx, SST_FW_LOADING); + + /* we have to restore the memory saved */ + ctx->ops->reset(ctx); + + ctx->fw_save = NULL; + + memcpy32_toio(ctx->iram, fw_save->iram, ctx->iram_end - ctx->iram_base); + memcpy32_toio(ctx->dram, fw_save->dram, ctx->dram_end - ctx->dram_base); + memcpy32_toio(ctx->mailbox, fw_save->sram, SST_MAILBOX_SIZE); + memcpy32_toio(ctx->ddr, fw_save->ddr, ctx->ddr_end - ctx->ddr_base); + + kfree(fw_save->sram); + kfree(fw_save->dram); + kfree(fw_save->iram); + kfree(fw_save->ddr); + kfree(fw_save); + + block = sst_create_block(ctx, 0, FW_DWNL_ID); + if (block == NULL) + return -ENOMEM; + + + /* start and wait for ack */ + ctx->ops->start(ctx); + ret = sst_wait_timeout(ctx, block); + if (ret) { + dev_err(ctx->dev, "fw download failed %d\n", ret); + /* FW download failed due to timeout */ + ret = -EBUSY; + + } else { + sst_set_fw_state_locked(ctx, SST_FW_RUNNING); + } + + sst_free_block(ctx, block); + return ret; +} + const struct dev_pm_ops intel_sst_pm = { + .suspend = intel_sst_suspend, + .resume = intel_sst_resume, .runtime_suspend = intel_sst_runtime_suspend, .runtime_resume = intel_sst_runtime_resume, }; diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index f793780a50a2..3f493862e98d 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -337,6 +337,13 @@ struct sst_shim_regs64 { u64 csr2; }; +struct sst_fw_save { + void *iram; + void *dram; + void *sram; + void *ddr; +}; + /** * struct intel_sst_drv - driver ops * @@ -428,6 +435,8 @@ struct intel_sst_drv { * persistent till worker thread gets called */ char firmware_name[FW_NAME_SIZE]; + + struct sst_fw_save *fw_save; }; /* misc definitions */ -- cgit v1.2.3-59-g8ed1b From 9308d1456e30e374d93957e3376a09370be9dc52 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 24 Feb 2015 11:39:45 +0530 Subject: ASoC: Intel: Move the fw download to power_control Thus removing the runtime_resume handler. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 18 +----------------- sound/soc/intel/sst/sst_drv_interface.c | 27 +++++++++++++++++++++++---- 2 files changed, 24 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 8f938112a01f..4d8f73ac5dd9 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -492,21 +492,6 @@ iram: return ret; } -static int intel_sst_runtime_resume(struct device *dev) -{ - int ret = 0; - struct intel_sst_drv *ctx = dev_get_drvdata(dev); - - if (ctx->sst_state == SST_RESET) { - ret = sst_load_fw(ctx); - if (ret) { - dev_err(dev, "FW download fail %d\n", ret); - sst_set_fw_state_locked(ctx, SST_RESET); - } - } - return ret; -} - static int intel_sst_resume(struct device *dev) { struct intel_sst_drv *ctx = dev_get_drvdata(dev); @@ -515,7 +500,7 @@ static int intel_sst_resume(struct device *dev) struct sst_block *block; if (!fw_save) - return intel_sst_runtime_resume(dev); + return 0; sst_set_fw_state_locked(ctx, SST_FW_LOADING); @@ -560,6 +545,5 @@ const struct dev_pm_ops intel_sst_pm = { .suspend = intel_sst_suspend, .resume = intel_sst_resume, .runtime_suspend = intel_sst_runtime_suspend, - .runtime_resume = intel_sst_runtime_resume, }; EXPORT_SYMBOL_GPL(intel_sst_pm); diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c index 5d56fcdd58d8..549af7d7f6d0 100644 --- a/sound/soc/intel/sst/sst_drv_interface.c +++ b/sound/soc/intel/sst/sst_drv_interface.c @@ -138,12 +138,31 @@ int sst_get_stream(struct intel_sst_drv *ctx, static int sst_power_control(struct device *dev, bool state) { struct intel_sst_drv *ctx = dev_get_drvdata(dev); + int ret = 0; - dev_dbg(ctx->dev, "state:%d", state); - if (state == true) - return pm_runtime_get_sync(dev); - else + if (state == true) { + ret = pm_runtime_get_sync(dev); + dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", + atomic_read(&dev->power.usage_count)); + if (ret < 0) { + dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret); + return ret; + } + if ((ctx->sst_state == SST_RESET) && + (atomic_read(&dev->power.usage_count) == 1)) { + ret = sst_load_fw(ctx); + if (ret) { + dev_err(dev, "FW download fail %d\n", ret); + sst_set_fw_state_locked(ctx, SST_RESET); + ret = sst_pm_runtime_put(ctx); + } + } + } else { + dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", + atomic_read(&dev->power.usage_count)); return sst_pm_runtime_put(ctx); + } + return ret; } /* -- cgit v1.2.3-59-g8ed1b From 583e58a0f0e996008780fe4df0f7640890a9b69a Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 24 Feb 2015 11:39:46 +0530 Subject: ASoC: Intel: Remove ignore suspend support In our platform we want platform and codec driver routines to get invoked and don't need the machine routines so remove here Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/cht_bsw_rt5672.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index a5098d6f988b..67db5101bc89 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -215,7 +215,6 @@ static struct snd_soc_dai_link cht_dailink[] = { .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", .platform_name = "sst-mfld-platform", - .ignore_suspend = 1, .nonatomic = true, .dynamic = 1, .dpcm_playback = 1, @@ -246,7 +245,6 @@ static struct snd_soc_dai_link cht_dailink[] = { | SND_SOC_DAIFMT_CBS_CFS, .init = cht_codec_init, .be_hw_params_fixup = cht_codec_fixup, - .ignore_suspend = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ops = &cht_be_ssp2_ops, -- cgit v1.2.3-59-g8ed1b From 3f2dcbeaeb2badb951a68e7d525ff4e55d0b0678 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 24 Feb 2015 11:39:47 +0530 Subject: ASoC: Intel: Remove soc pm handling to allow platform driver handle it Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/cht_bsw_rt5672.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index 67db5101bc89..bc8dcacd5e6a 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -283,7 +283,6 @@ static int snd_cht_mc_probe(struct platform_device *pdev) static struct platform_driver snd_cht_mc_driver = { .driver = { .name = "cht-bsw-rt5672", - .pm = &snd_soc_pm_ops, }, .probe = snd_cht_mc_probe, }; -- cgit v1.2.3-59-g8ed1b From 34d7c3905adb9a9d8f8155857c76314125510817 Mon Sep 17 00:00:00 2001 From: Uwe Kleine-König Date: Sat, 21 Feb 2015 16:33:24 +0100 Subject: ASoC: improve usage of gpiod API MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Since 39b2bbe3d715 (gpio: add flags argument to gpiod_get*() functions) which appeared in v3.17-rc1, the gpiod_get* functions take an additional parameter that allows to specify direction and initial value for output. Simplify drivers accordingly. Also there is an *_optional variant that serves well here. The sematics is slightly changed here by using it as error checking is more strict now: If GPIOLIB is not enabled an error is returned instead of just ignoring the gpio. On one hand this is bad for devices that don't "have" the respective gpio as the driver is failing now. On the other hand there is no means to assert that this gpio is really not needed or if only the driver to control it is not available. The latter is a real reason to fail and so it's defensive to fail here, too. Signed-off-by: Uwe Kleine-König Signed-off-by: Mark Brown --- sound/soc/codecs/adau1977.c | 17 +++++------------ sound/soc/codecs/cs35l32.c | 19 ++++++------------- sound/soc/codecs/cs4265.c | 19 ++++++------------- sound/soc/codecs/sta350.c | 30 +++++++++--------------------- sound/soc/codecs/tas2552.c | 13 +++---------- 5 files changed, 29 insertions(+), 69 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c index 70ab35744aba..7ad8e156e2df 100644 --- a/sound/soc/codecs/adau1977.c +++ b/sound/soc/codecs/adau1977.c @@ -938,22 +938,15 @@ int adau1977_probe(struct device *dev, struct regmap *regmap, adau1977->dvdd_reg = NULL; } - adau1977->reset_gpio = devm_gpiod_get(dev, "reset"); - if (IS_ERR(adau1977->reset_gpio)) { - ret = PTR_ERR(adau1977->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return PTR_ERR(adau1977->reset_gpio); - adau1977->reset_gpio = NULL; - } + adau1977->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(adau1977->reset_gpio)) + return PTR_ERR(adau1977->reset_gpio); dev_set_drvdata(dev, adau1977); - if (adau1977->reset_gpio) { - ret = gpiod_direction_output(adau1977->reset_gpio, 0); - if (ret) - return ret; + if (adau1977->reset_gpio) ndelay(100); - } ret = adau1977_power_enable(adau1977); if (ret) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index f2b8aad21274..60598b230341 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -437,20 +437,13 @@ static int cs35l32_i2c_probe(struct i2c_client *i2c_client, } /* Reset the Device */ - cs35l32->reset_gpio = devm_gpiod_get(&i2c_client->dev, - "reset-gpios"); - if (IS_ERR(cs35l32->reset_gpio)) { - ret = PTR_ERR(cs35l32->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - cs35l32->reset_gpio = NULL; - } else { - ret = gpiod_direction_output(cs35l32->reset_gpio, 0); - if (ret) - return ret; + cs35l32->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs35l32->reset_gpio)) + return PTR_ERR(cs35l32->reset_gpio); + + if (cs35l32->reset_gpio) gpiod_set_value_cansleep(cs35l32->reset_gpio, 1); - } /* initialize codec */ ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_AB, ®); diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index ce6086835ebd..cac48ddf3ba6 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -605,21 +605,14 @@ static int cs4265_i2c_probe(struct i2c_client *i2c_client, return ret; } - cs4265->reset_gpio = devm_gpiod_get(&i2c_client->dev, - "reset-gpios"); - if (IS_ERR(cs4265->reset_gpio)) { - ret = PTR_ERR(cs4265->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - cs4265->reset_gpio = NULL; - } else { - ret = gpiod_direction_output(cs4265->reset_gpio, 0); - if (ret) - return ret; + cs4265->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs4265->reset_gpio)) + return PTR_ERR(cs4265->reset_gpio); + + if (cs4265->reset_gpio) { mdelay(1); gpiod_set_value_cansleep(cs4265->reset_gpio, 1); - } i2c_set_clientdata(i2c_client, cs4265); diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c index bda2ee18769e..669e3228241e 100644 --- a/sound/soc/codecs/sta350.c +++ b/sound/soc/codecs/sta350.c @@ -1213,27 +1213,15 @@ static int sta350_i2c_probe(struct i2c_client *i2c, #endif /* GPIOs */ - sta350->gpiod_nreset = devm_gpiod_get(dev, "reset"); - if (IS_ERR(sta350->gpiod_nreset)) { - ret = PTR_ERR(sta350->gpiod_nreset); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta350->gpiod_nreset = NULL; - } else { - gpiod_direction_output(sta350->gpiod_nreset, 0); - } - - sta350->gpiod_power_down = devm_gpiod_get(dev, "power-down"); - if (IS_ERR(sta350->gpiod_power_down)) { - ret = PTR_ERR(sta350->gpiod_power_down); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta350->gpiod_power_down = NULL; - } else { - gpiod_direction_output(sta350->gpiod_power_down, 0); - } + sta350->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(sta350->gpiod_nreset)) + return PTR_ERR(sta350->gpiod_nreset); + + sta350->gpiod_power_down = devm_gpiod_get(dev, "power-down", + GPIOD_OUT_LOW); + if (IS_ERR(sta350->gpiod_power_down)) + return PTR_ERR(sta350->gpiod_power_down); /* regulators */ for (i = 0; i < ARRAY_SIZE(sta350->supplies); i++) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index ae23acdd2708..dfb4ff5cc9ea 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -485,16 +485,9 @@ static int tas2552_probe(struct i2c_client *client, if (data == NULL) return -ENOMEM; - data->enable_gpio = devm_gpiod_get(dev, "enable"); - if (IS_ERR(data->enable_gpio)) { - ret = PTR_ERR(data->enable_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - data->enable_gpio = NULL; - } else { - gpiod_direction_output(data->enable_gpio, 0); - } + data->enable_gpio = devm_gpiod_get(dev, "enable", GPIOD_OUT_LOW); + if (IS_ERR(data->enable_gpio)) + return PTR_ERR(data->enable_gpio); data->tas2552_client = client; data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); -- cgit v1.2.3-59-g8ed1b From 5890bd5256bc026c425361fa087dc05c7a24d853 Mon Sep 17 00:00:00 2001 From: Peter Rosin Date: Mon, 16 Feb 2015 22:02:47 +0100 Subject: ASoC: pcm512x: Rearrange to not repeat dacsrc_rate / dac_div Signed-off-by: Peter Rosin Signed-off-by: Mark Brown --- sound/soc/codecs/pcm512x.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 9974f201a08f..f11c76f1acfe 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -863,28 +863,29 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, dacsrc_rate = sck_rate; } + osr_div = DIV_ROUND_CLOSEST(dac_rate, osr_rate); + if (osr_div > 128) { + dev_err(dev, "Failed to find OSR divider\n"); + return -EINVAL; + } + dac_div = DIV_ROUND_CLOSEST(dacsrc_rate, dac_rate); if (dac_div > 128) { dev_err(dev, "Failed to find DAC divider\n"); return -EINVAL; } + dac_rate = dacsrc_rate / dac_div; - ncp_div = DIV_ROUND_CLOSEST(dacsrc_rate / dac_div, 1536000); - if (ncp_div > 128 || dacsrc_rate / dac_div / ncp_div > 2048000) { + ncp_div = DIV_ROUND_CLOSEST(dac_rate, 1536000); + if (ncp_div > 128 || dac_rate / ncp_div > 2048000) { /* run NCP no faster than 2048000 Hz, but why? */ - ncp_div = DIV_ROUND_UP(dacsrc_rate / dac_div, 2048000); + ncp_div = DIV_ROUND_UP(dac_rate, 2048000); if (ncp_div > 128) { dev_err(dev, "Failed to find NCP divider\n"); return -EINVAL; } } - osr_div = DIV_ROUND_CLOSEST(dac_rate, osr_rate); - if (osr_div > 128) { - dev_err(dev, "Failed to find OSR divider\n"); - return -EINVAL; - } - idac = mck_rate / (dsp_div * sample_rate); ret = regmap_write(pcm512x->regmap, PCM512x_DSP_CLKDIV, dsp_div - 1); -- cgit v1.2.3-59-g8ed1b From f29933c9ae4b8f30c713186d3babb630c7cfb4f2 Mon Sep 17 00:00:00 2001 From: Peter Rosin Date: Mon, 23 Feb 2015 21:03:33 +0100 Subject: ASoC: pcm512x: Allow independently overclocking PLL, DAC and DSP When using non-standard rates, a relatively small amount of overclocking can make a big difference to a number of cases. - Not all rates are possible to achieve with the PLL, due to divider restrictions. - The higher oversampling rates that can be used by the DAC, the simpler the analog output filters get (mirror frequencies move up, away from the desired spectrum). - The more work the DSP can perform per sample, the better. For standard rates, there is little to gain as everything is designed just right, and the needed overclocking to make a real difference would be significant. Signed-off-by: Peter Rosin Signed-off-by: Mark Brown --- sound/soc/codecs/pcm512x.c | 161 +++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 150 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index f11c76f1acfe..4b5f1fe9be97 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -54,6 +54,9 @@ struct pcm512x_priv { int pll_d; int pll_p; unsigned long real_pll; + unsigned long overclock_pll; + unsigned long overclock_dac; + unsigned long overclock_dsp; }; /* @@ -224,6 +227,90 @@ static bool pcm512x_volatile(struct device *dev, unsigned int reg) } } +static int pcm512x_overclock_pll_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = pcm512x->overclock_pll; + return 0; +} + +static int pcm512x_overclock_pll_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_STANDBY: + break; + default: + return -EBUSY; + } + + pcm512x->overclock_pll = ucontrol->value.integer.value[0]; + return 0; +} + +static int pcm512x_overclock_dsp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = pcm512x->overclock_dsp; + return 0; +} + +static int pcm512x_overclock_dsp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_STANDBY: + break; + default: + return -EBUSY; + } + + pcm512x->overclock_dsp = ucontrol->value.integer.value[0]; + return 0; +} + +static int pcm512x_overclock_dac_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = pcm512x->overclock_dac; + return 0; +} + +static int pcm512x_overclock_dac_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_STANDBY: + break; + default: + return -EBUSY; + } + + pcm512x->overclock_dac = ucontrol->value.integer.value[0]; + return 0; +} + static const DECLARE_TLV_DB_SCALE(digital_tlv, -10350, 50, 1); static const DECLARE_TLV_DB_SCALE(analog_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 80, 0); @@ -328,6 +415,13 @@ SOC_ENUM("Volume Ramp Up Rate", pcm512x_vnuf), SOC_ENUM("Volume Ramp Up Step", pcm512x_vnus), SOC_ENUM("Volume Ramp Down Emergency Rate", pcm512x_vedf), SOC_ENUM("Volume Ramp Down Emergency Step", pcm512x_veds), + +SOC_SINGLE_EXT("Max Overclock PLL", SND_SOC_NOPM, 0, 20, 0, + pcm512x_overclock_pll_get, pcm512x_overclock_pll_put), +SOC_SINGLE_EXT("Max Overclock DSP", SND_SOC_NOPM, 0, 40, 0, + pcm512x_overclock_dsp_get, pcm512x_overclock_dsp_put), +SOC_SINGLE_EXT("Max Overclock DAC", SND_SOC_NOPM, 0, 40, 0, + pcm512x_overclock_dac_get, pcm512x_overclock_dac_put), }; static const struct snd_soc_dapm_widget pcm512x_dapm_widgets[] = { @@ -346,6 +440,45 @@ static const struct snd_soc_dapm_route pcm512x_dapm_routes[] = { { "OUTR", NULL, "DACR" }, }; +static unsigned long pcm512x_pll_max(struct pcm512x_priv *pcm512x) +{ + return 25000000 + 25000000 * pcm512x->overclock_pll / 100; +} + +static unsigned long pcm512x_dsp_max(struct pcm512x_priv *pcm512x) +{ + return 50000000 + 50000000 * pcm512x->overclock_dsp / 100; +} + +static unsigned long pcm512x_dac_max(struct pcm512x_priv *pcm512x, + unsigned long rate) +{ + return rate + rate * pcm512x->overclock_dac / 100; +} + +static unsigned long pcm512x_sck_max(struct pcm512x_priv *pcm512x) +{ + if (!pcm512x->pll_out) + return 25000000; + return pcm512x_pll_max(pcm512x); +} + +static unsigned long pcm512x_ncp_target(struct pcm512x_priv *pcm512x, + unsigned long dac_rate) +{ + /* + * If the DAC is not actually overclocked, use the good old + * NCP target rate... + */ + if (dac_rate <= 6144000) + return 1536000; + /* + * ...but if the DAC is in fact overclocked, bump the NCP target + * rate to get the recommended dividers even when overclocking. + */ + return pcm512x_dac_max(pcm512x, 1536000); +} + static const u32 pcm512x_dai_rates[] = { 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, 384000, @@ -359,6 +492,7 @@ static const struct snd_pcm_hw_constraint_list constraints_slave = { static int pcm512x_hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { + struct pcm512x_priv *pcm512x = rule->private; struct snd_interval ranges[2]; int frame_size; @@ -377,7 +511,7 @@ static int pcm512x_hw_rule_rate(struct snd_pcm_hw_params *params, */ memset(ranges, 0, sizeof(ranges)); ranges[0].min = 8000; - ranges[0].max = 25000000 / frame_size / 2; + ranges[0].max = pcm512x_sck_max(pcm512x) / frame_size / 2; ranges[1].min = DIV_ROUND_UP(16000000, frame_size); ranges[1].max = 384000; break; @@ -408,7 +542,7 @@ static int pcm512x_dai_startup_master(struct snd_pcm_substream *substream, return snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, pcm512x_hw_rule_rate, - NULL, + pcm512x, SNDRV_PCM_HW_PARAM_FRAME_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1); @@ -517,6 +651,8 @@ static unsigned long pcm512x_find_sck(struct snd_soc_dai *dai, unsigned long bclk_rate) { struct device *dev = dai->dev; + struct snd_soc_codec *codec = dai->codec; + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); unsigned long sck_rate; int pow2; @@ -527,9 +663,10 @@ static unsigned long pcm512x_find_sck(struct snd_soc_dai *dai, * as many factors of 2 as possible, as that makes it easier * to find a fast DAC rate */ - pow2 = 1 << fls((25000000 - 16000000) / bclk_rate); + pow2 = 1 << fls((pcm512x_pll_max(pcm512x) - 16000000) / bclk_rate); for (; pow2; pow2 >>= 1) { - sck_rate = rounddown(25000000, bclk_rate * pow2); + sck_rate = rounddown(pcm512x_pll_max(pcm512x), + bclk_rate * pow2); if (sck_rate >= 16000000) break; } @@ -678,7 +815,7 @@ static unsigned long pcm512x_pllin_dac_rate(struct snd_soc_dai *dai, return 0; /* futile, quit early */ /* run DAC no faster than 6144000 Hz */ - for (dac_rate = rounddown(6144000, osr_rate); + for (dac_rate = rounddown(pcm512x_dac_max(pcm512x, 6144000), osr_rate); dac_rate; dac_rate -= osr_rate) { @@ -805,7 +942,7 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, osr_rate = 16 * sample_rate; /* run DSP no faster than 50 MHz */ - dsp_div = mck_rate > 50000000 ? 2 : 1; + dsp_div = mck_rate > pcm512x_dsp_max(pcm512x) ? 2 : 1; dac_rate = pcm512x_pllin_dac_rate(dai, osr_rate, pllin_rate); if (dac_rate) { @@ -836,7 +973,8 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, dacsrc_rate = pllin_rate; } else { /* run DAC no faster than 6144000 Hz */ - unsigned long dac_mul = 6144000 / osr_rate; + unsigned long dac_mul = pcm512x_dac_max(pcm512x, 6144000) + / osr_rate; unsigned long sck_mul = sck_rate / osr_rate; for (; dac_mul; dac_mul--) { @@ -876,7 +1014,8 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, } dac_rate = dacsrc_rate / dac_div; - ncp_div = DIV_ROUND_CLOSEST(dac_rate, 1536000); + ncp_div = DIV_ROUND_CLOSEST(dac_rate, + pcm512x_ncp_target(pcm512x, dac_rate)); if (ncp_div > 128 || dac_rate / ncp_div > 2048000) { /* run NCP no faster than 2048000 Hz, but why? */ ncp_div = DIV_ROUND_UP(dac_rate, 2048000); @@ -938,11 +1077,11 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, return ret; } - if (sample_rate <= 48000) + if (sample_rate <= pcm512x_dac_max(pcm512x, 48000)) fssp = PCM512x_FSSP_48KHZ; - else if (sample_rate <= 96000) + else if (sample_rate <= pcm512x_dac_max(pcm512x, 96000)) fssp = PCM512x_FSSP_96KHZ; - else if (sample_rate <= 192000) + else if (sample_rate <= pcm512x_dac_max(pcm512x, 192000)) fssp = PCM512x_FSSP_192KHZ; else fssp = PCM512x_FSSP_384KHZ; -- cgit v1.2.3-59-g8ed1b From f23e860edbb3f2208c0ab3448e756689bb4a3760 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Sat, 14 Feb 2015 17:22:49 -0800 Subject: ASoC: core: Add extra dapm properties for Device Tree The current helper functions, snd_soc_of_parse_audio_simple_widgets() and snd_soc_of_parse_audio_routing(), set dapm_widgets and dapm_routes without caring if they are already set by using build-in widgets and routes in the card driver. So there could be one of them, build-in one or Device Tree one, overrided by the other depending on which one was assigned later. This patch adds an extra pair of dapm_widgets and dapm_routes for DT use only so as to prevent unexpected overriding. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 16 ++++++++++++---- 2 files changed, 17 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade195628..f66a1ef98a40 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1071,11 +1071,16 @@ struct snd_soc_card { /* * Card-specific routes and widgets. + * Note: of_dapm_xxx for Device Tree; Otherwise for driver build-in. */ const struct snd_soc_dapm_widget *dapm_widgets; int num_dapm_widgets; const struct snd_soc_dapm_route *dapm_routes; int num_dapm_routes; + const struct snd_soc_dapm_widget *of_dapm_widgets; + int num_of_dapm_widgets; + const struct snd_soc_dapm_route *of_dapm_routes; + int num_of_dapm_routes; bool fully_routed; struct work_struct deferred_resume_work; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 30579ca5bacb..5c0658d49609 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1561,6 +1561,10 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets, card->num_dapm_widgets); + if (card->of_dapm_widgets) + snd_soc_dapm_new_controls(&card->dapm, card->of_dapm_widgets, + card->num_of_dapm_widgets); + /* initialise the sound card only once */ if (card->probe) { ret = card->probe(card); @@ -1616,6 +1620,10 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); + if (card->of_dapm_routes) + snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes, + card->num_of_dapm_routes); + for (i = 0; i < card->num_links; i++) { if (card->dai_link[i].dai_fmt) snd_soc_runtime_set_dai_fmt(&card->rtd[i], @@ -3223,8 +3231,8 @@ int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, widgets[i].name = wname; } - card->dapm_widgets = widgets; - card->num_dapm_widgets = num_widgets; + card->of_dapm_widgets = widgets; + card->num_of_dapm_widgets = num_widgets; return 0; } @@ -3308,8 +3316,8 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } } - card->num_dapm_routes = num_routes; - card->dapm_routes = routes; + card->num_of_dapm_routes = num_routes; + card->of_dapm_routes = routes; return 0; } -- cgit v1.2.3-59-g8ed1b From 3185878a70e721644b0e32ebbc0a039616551949 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Sat, 14 Feb 2015 17:22:50 -0800 Subject: ASoC: fsl-asoc-card: Add snd_soc_of_parse_audio_routing() This patch adds snd_soc_of_parse_audio_routing() to get dapm routes configurations via Device Tree. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 3f6959c8e2f7..de438871040b 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -512,6 +512,12 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) memcpy(priv->dai_link, fsl_asoc_card_dai, sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); + ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing"); + if (ret) { + dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret); + goto asrc_fail; + } + /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; priv->dai_link[0].codec_of_node = codec_np; -- cgit v1.2.3-59-g8ed1b From e2cef68d5903cc2052e9f6e46b323b7ead695e73 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Tue, 10 Feb 2015 17:01:56 +0800 Subject: ASoC: rt286: add jack detection disable with NULL jack passed Some platforms, e.g. WSB, don't need jack detection when system is in Suspend, for power save reason. Here add headphone/mic jack detection disable feature with NULL jack passed in, when disabled, it will disable interrupt, and disable LDO1, which is used for jack detection when headphone is plugged in. Signed-off-by: Jie Yang Reviewed-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index f374840a5a7c..16723b167fbf 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -395,9 +395,20 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) rt286->jack = jack; - /* Send an initial empty report */ - snd_soc_jack_report(rt286->jack, 0, - SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + if (jack) { + /* enable IRQ */ + if (rt286->jack->status | SND_JACK_HEADPHONE) + snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO1"); + regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x2); + /* Send an initial empty report */ + snd_soc_jack_report(rt286->jack, rt286->jack->status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + } else { + /* disable IRQ */ + regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x0); + snd_soc_dapm_disable_pin(&codec->dapm, "LDO1"); + } + snd_soc_dapm_sync(&codec->dapm); return 0; } -- cgit v1.2.3-59-g8ed1b From 5bb400ce4a9b100a2dd3f5db17c4c76877cecade Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 25 Feb 2015 21:37:52 +0530 Subject: ASoC: Intel: wrap runtime_pm usage count under CONFIG_PM The struct dev_pm_ops defines usage_count only when CONFIG_PM is defined. So we should use this variable only in cases where this falg is true. So we define a local variable and read the value under this flag. In non PM cases, we set this to 1. Reported-by: kbuild test robot Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_drv_interface.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c index 549af7d7f6d0..f0e4b99b3aeb 100644 --- a/sound/soc/intel/sst/sst_drv_interface.c +++ b/sound/soc/intel/sst/sst_drv_interface.c @@ -139,17 +139,23 @@ static int sst_power_control(struct device *dev, bool state) { struct intel_sst_drv *ctx = dev_get_drvdata(dev); int ret = 0; + int usage_count = 0; + +#ifdef CONFIG_PM + usage_count = atomic_read(&dev->power.usage_count); +#else + usage_count = 1; +#endif if (state == true) { ret = pm_runtime_get_sync(dev); - dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", - atomic_read(&dev->power.usage_count)); + + dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count); if (ret < 0) { dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret); return ret; } - if ((ctx->sst_state == SST_RESET) && - (atomic_read(&dev->power.usage_count) == 1)) { + if ((ctx->sst_state == SST_RESET) && (usage_count == 1)) { ret = sst_load_fw(ctx); if (ret) { dev_err(dev, "FW download fail %d\n", ret); @@ -158,8 +164,7 @@ static int sst_power_control(struct device *dev, bool state) } } } else { - dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", - atomic_read(&dev->power.usage_count)); + dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", usage_count); return sst_pm_runtime_put(ctx); } return ret; -- cgit v1.2.3-59-g8ed1b From 92b2ad2c9e18ca2bfa8727af7edcd372d9acaac4 Mon Sep 17 00:00:00 2001 From: Kenneth Westfield Date: Tue, 24 Feb 2015 22:39:04 -0800 Subject: ASoC: max98357a: Use standard DAI names Use the standard naming convention for the codec DAI. Signed-off-by: Kenneth Westfield Signed-off-by: Mark Brown --- sound/soc/codecs/max98357a.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 4ee23fbc4e12..bf3e933ee895 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -85,9 +85,9 @@ static struct snd_soc_dai_ops max98357a_dai_ops = { }; static struct snd_soc_dai_driver max98357a_dai_driver = { - .name = "max98357a", + .name = "HiFi", .playback = { - .stream_name = "max98357a-playback", + .stream_name = "HiFi Playback", .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32, -- cgit v1.2.3-59-g8ed1b From 223c055aa0eb7e606eb7132e339ce66bb8d7be0d Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 18 Dec 2014 11:32:52 +0800 Subject: ASoC: rt5670: set platform data by dmi This patch set specific data according to dmi data. Signed-off-by: Jin, Yao Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 8a0833de1665..cd47ef1f5561 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -2549,6 +2550,17 @@ static struct acpi_device_id rt5670_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5670_acpi_match); #endif +static const struct dmi_system_id dmi_platform_intel_braswell[] = { + { + .ident = "Intel Braswell", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Braswell CRB"), + }, + }, + {} +}; + static int rt5670_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2568,6 +2580,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, if (pdata) rt5670->pdata = *pdata; + if (dmi_check_system(dmi_platform_intel_braswell)) { + rt5670->pdata.dmic_en = true; + rt5670->pdata.dmic1_data_pin = RT5670_DMIC_DATA_IN2P; + rt5670->pdata.jd_mode = 1; + } + rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap); if (IS_ERR(rt5670->regmap)) { ret = PTR_ERR(rt5670->regmap); -- cgit v1.2.3-59-g8ed1b From 64e89e5f55484d289c8b326521e5a12291e2283e Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 15 Dec 2014 15:42:33 +0800 Subject: ASoC: rt5670: Add runtime PM support This patch adds runtime PM support on rt5670 codec. Signed-off-by: Lin Mengdong Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index cd47ef1f5561..78d85de8af6f 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include #include @@ -2734,18 +2735,26 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, } + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5670, rt5670_dai, ARRAY_SIZE(rt5670_dai)); if (ret < 0) goto err; + pm_runtime_put(&i2c->dev); + return 0; err: + pm_runtime_disable(&i2c->dev); + return ret; } static int rt5670_i2c_remove(struct i2c_client *i2c) { + pm_runtime_disable(&i2c->dev); snd_soc_unregister_codec(&i2c->dev); return 0; -- cgit v1.2.3-59-g8ed1b From 77e3ea2801c8ca4700e6b17053b625b8a981ac77 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 15 Dec 2014 15:42:34 +0800 Subject: ASoC: rt5670: Keep sysclk on if JD func is used System clock is necessary for rt5670 JD function. We assume system clock source will be set in machine driver. So there are two things left we should do in codec driver. 1. Set sysclk to codec internal clock in probe since machine driver may not do that before JD function is registered. 2. Power up PLL once sysclk source is switched to PLL. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 78d85de8af6f..0a027bc94399 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2190,6 +2190,13 @@ static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai, if (freq == rt5670->sysclk && clk_id == rt5670->sysclk_src) return 0; + if (rt5670->pdata.jd_mode) { + if (clk_id == RT5670_SCLK_S_PLL1) + snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1"); + else + snd_soc_dapm_disable_pin(&codec->dapm, "PLL1"); + snd_soc_dapm_sync(&codec->dapm); + } switch (clk_id) { case RT5670_SCLK_S_MCLK: reg_val |= RT5670_SCLK_SRC_MCLK; @@ -2628,6 +2635,10 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, } if (rt5670->pdata.jd_mode) { + regmap_update_bits(rt5670->regmap, RT5670_GLB_CLK, + RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_RCCLK); + rt5670->sysclk = 0; + rt5670->sysclk_src = RT5670_SCLK_S_RCCLK; regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, RT5670_PWR_MB, RT5670_PWR_MB); regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG2, -- cgit v1.2.3-59-g8ed1b From 3aebec3a701e70d6fe2816891e5abea066492779 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 7 Jan 2015 10:19:06 +0800 Subject: ASoC: rt5670: redefine ASRC control registers 0x84 and 0x85 The previous definition of registers 0x84 and 0x85 doesn't match the datasheet. So this patch removes the wrong definition and writes a new one for the two registers. Signed-off-by: Bard Liao Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.h | 65 +++++++++++++++++------------------------------ 1 file changed, 24 insertions(+), 41 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 84857bdaa78b..82553b1726cd 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1023,50 +1023,33 @@ #define RT5670_DMIC_2_M_NOR (0x0 << 8) #define RT5670_DMIC_2_M_ASYN (0x1 << 8) +/* ASRC clock source selection (0x84, 0x85) */ +#define RT5670_CLK_SEL_SYS (0x0) +#define RT5670_CLK_SEL_I2S1_ASRC (0x1) +#define RT5670_CLK_SEL_I2S2_ASRC (0x2) +#define RT5670_CLK_SEL_I2S3_ASRC (0x3) +#define RT5670_CLK_SEL_SYS2 (0x5) +#define RT5670_CLK_SEL_SYS3 (0x6) + /* ASRC Control 2 (0x84) */ -#define RT5670_MDA_L_M_MASK (0x1 << 15) -#define RT5670_MDA_L_M_SFT 15 -#define RT5670_MDA_L_M_NOR (0x0 << 15) -#define RT5670_MDA_L_M_ASYN (0x1 << 15) -#define RT5670_MDA_R_M_MASK (0x1 << 14) -#define RT5670_MDA_R_M_SFT 14 -#define RT5670_MDA_R_M_NOR (0x0 << 14) -#define RT5670_MDA_R_M_ASYN (0x1 << 14) -#define RT5670_MAD_L_M_MASK (0x1 << 13) -#define RT5670_MAD_L_M_SFT 13 -#define RT5670_MAD_L_M_NOR (0x0 << 13) -#define RT5670_MAD_L_M_ASYN (0x1 << 13) -#define RT5670_MAD_R_M_MASK (0x1 << 12) -#define RT5670_MAD_R_M_SFT 12 -#define RT5670_MAD_R_M_NOR (0x0 << 12) -#define RT5670_MAD_R_M_ASYN (0x1 << 12) -#define RT5670_ADC_M_MASK (0x1 << 11) -#define RT5670_ADC_M_SFT 11 -#define RT5670_ADC_M_NOR (0x0 << 11) -#define RT5670_ADC_M_ASYN (0x1 << 11) -#define RT5670_STO_DAC_M_MASK (0x1 << 5) -#define RT5670_STO_DAC_M_SFT 5 -#define RT5670_STO_DAC_M_NOR (0x0 << 5) -#define RT5670_STO_DAC_M_ASYN (0x1 << 5) -#define RT5670_I2S1_R_D_MASK (0x1 << 4) -#define RT5670_I2S1_R_D_SFT 4 -#define RT5670_I2S1_R_D_DIS (0x0 << 4) -#define RT5670_I2S1_R_D_EN (0x1 << 4) -#define RT5670_I2S2_R_D_MASK (0x1 << 3) -#define RT5670_I2S2_R_D_SFT 3 -#define RT5670_I2S2_R_D_DIS (0x0 << 3) -#define RT5670_I2S2_R_D_EN (0x1 << 3) -#define RT5670_PRE_SCLK_MASK (0x3) -#define RT5670_PRE_SCLK_SFT 0 -#define RT5670_PRE_SCLK_512 (0x0) -#define RT5670_PRE_SCLK_1024 (0x1) -#define RT5670_PRE_SCLK_2048 (0x2) +#define RT5670_DA_STO_CLK_SEL_MASK (0xf << 12) +#define RT5670_DA_STO_CLK_SEL_SFT 12 +#define RT5670_DA_MONOL_CLK_SEL_MASK (0xf << 8) +#define RT5670_DA_MONOL_CLK_SEL_SFT 8 +#define RT5670_DA_MONOR_CLK_SEL_MASK (0xf << 4) +#define RT5670_DA_MONOR_CLK_SEL_SFT 4 +#define RT5670_AD_STO1_CLK_SEL_MASK (0xf << 0) +#define RT5670_AD_STO1_CLK_SEL_SFT 0 /* ASRC Control 3 (0x85) */ -#define RT5670_I2S1_RATE_MASK (0xf << 12) -#define RT5670_I2S1_RATE_SFT 12 -#define RT5670_I2S2_RATE_MASK (0xf << 8) -#define RT5670_I2S2_RATE_SFT 8 +#define RT5670_UP_CLK_SEL_MASK (0xf << 12) +#define RT5670_UP_CLK_SEL_SFT 12 +#define RT5670_DOWN_CLK_SEL_MASK (0xf << 8) +#define RT5670_DOWN_CLK_SEL_SFT 8 +#define RT5670_AD_MONOL_CLK_SEL_MASK (0xf << 4) +#define RT5670_AD_MONOL_CLK_SEL_SFT 4 +#define RT5670_AD_MONOR_CLK_SEL_MASK (0xf << 0) +#define RT5670_AD_MONOR_CLK_SEL_SFT 0 /* ASRC Control 4 (0x89) */ #define RT5670_I2S1_PD_MASK (0x7 << 12) -- cgit v1.2.3-59-g8ed1b From ea232b3f7233f9242e5a1287377fedb6972dfea4 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Wed, 7 Jan 2015 10:19:12 +0800 Subject: ASoC: rt5670: add API to select ASRC clock source When codec is in slave mode, ASRC can suppress noise for asynchronous MCLK and LRCLK or special I2S format. This patch defines an API to select the clock source for specified filters. And the codec driver will turn on ASRC for these filters if ASRC is selected as their clock source. Signed-off-by: Bard Liao Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 83 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5670.h | 15 +++++++++ 2 files changed, 98 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0a027bc94399..0632b7458a53 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -590,6 +590,89 @@ static int can_use_asrc(struct snd_soc_dapm_widget *source, return 0; } + +/** + * rt5670_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @codec: SoC audio codec device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5670 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the codec driver will turn on ASRC + * for these filters if ASRC is selected as their clock source. + */ +int rt5670_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src) +{ + unsigned int asrc2_mask = 0, asrc2_value = 0; + unsigned int asrc3_mask = 0, asrc3_value = 0; + + if (clk_src > RT5670_CLK_SEL_SYS3) + return -EINVAL; + + if (filter_mask & RT5670_DA_STEREO_FILTER) { + asrc2_mask |= RT5670_DA_STO_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_DA_STO_CLK_SEL_MASK) + | (clk_src << RT5670_DA_STO_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_DA_MONO_L_FILTER) { + asrc2_mask |= RT5670_DA_MONOL_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_DA_MONOL_CLK_SEL_MASK) + | (clk_src << RT5670_DA_MONOL_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_DA_MONO_R_FILTER) { + asrc2_mask |= RT5670_DA_MONOR_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_DA_MONOR_CLK_SEL_MASK) + | (clk_src << RT5670_DA_MONOR_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_AD_STEREO_FILTER) { + asrc2_mask |= RT5670_AD_STO1_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_AD_STO1_CLK_SEL_MASK) + | (clk_src << RT5670_AD_STO1_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_AD_MONO_L_FILTER) { + asrc3_mask |= RT5670_AD_MONOL_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_AD_MONOL_CLK_SEL_MASK) + | (clk_src << RT5670_AD_MONOL_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_AD_MONO_R_FILTER) { + asrc3_mask |= RT5670_AD_MONOR_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_AD_MONOR_CLK_SEL_MASK) + | (clk_src << RT5670_AD_MONOR_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_UP_RATE_FILTER) { + asrc3_mask |= RT5670_UP_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_UP_CLK_SEL_MASK) + | (clk_src << RT5670_UP_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_DOWN_RATE_FILTER) { + asrc3_mask |= RT5670_DOWN_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_DOWN_CLK_SEL_MASK) + | (clk_src << RT5670_DOWN_CLK_SEL_SFT); + } + + if (asrc2_mask) + snd_soc_update_bits(codec, RT5670_ASRC_2, + asrc2_mask, asrc2_value); + + if (asrc3_mask) + snd_soc_update_bits(codec, RT5670_ASRC_3, + asrc3_mask, asrc3_value); + return 0; +} +EXPORT_SYMBOL_GPL(rt5670_sel_asrc_clk_src); + /* Digital Mixer */ static const struct snd_kcontrol_new rt5670_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 82553b1726cd..0a67adbcfbc3 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1969,6 +1969,21 @@ enum { RT5670_DMIC_DATA_GPIO5, }; +/* filter mask */ +enum { + RT5670_DA_STEREO_FILTER = 0x1, + RT5670_DA_MONO_L_FILTER = (0x1 << 1), + RT5670_DA_MONO_R_FILTER = (0x1 << 2), + RT5670_AD_STEREO_FILTER = (0x1 << 3), + RT5670_AD_MONO_L_FILTER = (0x1 << 4), + RT5670_AD_MONO_R_FILTER = (0x1 << 5), + RT5670_UP_RATE_FILTER = (0x1 << 6), + RT5670_DOWN_RATE_FILTER = (0x1 << 7), +}; + +int rt5670_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src); + struct rt5670_priv { struct snd_soc_codec *codec; struct rt5670_platform_data pdata; -- cgit v1.2.3-59-g8ed1b From ab1f70952f61504f60805f13660c8740adcbe14f Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 11 Feb 2015 19:18:51 +0800 Subject: ASoC: rt5677: Add the chip type to distinguish the setting of the clock source There is only one clock source in the rt5676, so the chip type is added to distinguish the setting of the clock source in the VAD function. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 28 ++++++++++++++++++++++------ sound/soc/codecs/rt5677.h | 6 ++++++ 2 files changed, 28 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5d0bb8748dd1..ab62777dbd33 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -718,11 +718,24 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) RT5677_LDO1_SEL_MASK, 0x0); regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, RT5677_PWR_LDO1, RT5677_PWR_LDO1); - regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, - RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); - regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, - RT5677_PLL2_PR_SRC_MASK | RT5677_DSP_CLK_SRC_MASK, - RT5677_PLL2_PR_SRC_MCLK2 | RT5677_DSP_CLK_SRC_BYPASS); + switch (rt5677->type) { + case RT5677: + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, + RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_PLL2_PR_SRC_MASK | + RT5677_DSP_CLK_SRC_MASK, + RT5677_PLL2_PR_SRC_MCLK2 | + RT5677_DSP_CLK_SRC_BYPASS); + break; + case RT5676: + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_DSP_CLK_SRC_MASK, + RT5677_DSP_CLK_SRC_BYPASS); + break; + default: + break; + } regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x07ff); regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07fd); rt5677_set_dsp_mode(codec, true); @@ -4733,7 +4746,8 @@ static const struct regmap_config rt5677_regmap = { }; static const struct i2c_device_id rt5677_i2c_id[] = { - { "rt5677", 0 }, + { "rt5677", RT5677 }, + { "rt5676", RT5676 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); @@ -4850,6 +4864,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5677); + rt5677->type = id->driver_data; + if (pdata) rt5677->pdata = *pdata; diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index c0a625f290cc..07df96b43f59 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1665,6 +1665,11 @@ enum { RT5677_IRQ_JD3, }; +enum rt5677_type { + RT5677, + RT5676, +}; + struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; @@ -1681,6 +1686,7 @@ struct rt5677_priv { int pll_in; int pll_out; int pow_ldo2; /* POW_LDO2 pin */ + enum rt5677_type type; #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; #endif -- cgit v1.2.3-59-g8ed1b From cbca4076d156c93cedadabb0e463ba0db16bb166 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 25 Feb 2015 17:36:14 +0800 Subject: ASoC: rt5677: Keep the LDO2 powered while used in the suspend mode The patch keeps the ldo2 power while the DSP function of "Voice Wake Up" used in the suspend mode. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index ab62777dbd33..5ff7ffaec5cc 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4513,10 +4513,10 @@ static int rt5677_suspend(struct snd_soc_codec *codec) if (!rt5677->dsp_vad_en) { regcache_cache_only(rt5677->regmap, true); regcache_mark_dirty(rt5677->regmap); - } - if (gpio_is_valid(rt5677->pow_ldo2)) - gpio_set_value_cansleep(rt5677->pow_ldo2, 0); + if (gpio_is_valid(rt5677->pow_ldo2)) + gpio_set_value_cansleep(rt5677->pow_ldo2, 0); + } return 0; } @@ -4525,12 +4525,12 @@ static int rt5677_resume(struct snd_soc_codec *codec) { struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); - if (gpio_is_valid(rt5677->pow_ldo2)) { - gpio_set_value_cansleep(rt5677->pow_ldo2, 1); - msleep(10); - } - if (!rt5677->dsp_vad_en) { + if (gpio_is_valid(rt5677->pow_ldo2)) { + gpio_set_value_cansleep(rt5677->pow_ldo2, 1); + msleep(10); + } + regcache_cache_only(rt5677->regmap, false); regcache_sync(rt5677->regmap); } -- cgit v1.2.3-59-g8ed1b From a0cf43e2f0f391dad7882febbf04423e73e3ff99 Mon Sep 17 00:00:00 2001 From: Tomeu Vizoso Date: Thu, 12 Feb 2015 09:41:55 +0100 Subject: ASoC: tegra: Expose Headphones pin to userspace So userspace can enable or disable it based on the current policy. Signed-off-by: Tomeu Vizoso Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_max98090.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index af3fb997b752..8df71a436f11 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -136,6 +136,7 @@ static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = { }; static const struct snd_kcontrol_new tegra_max98090_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), SOC_DAPM_PIN_SWITCH("Speakers"), }; -- cgit v1.2.3-59-g8ed1b From 3a4562f756617b4b210fc487bfe23853a450d3c1 Mon Sep 17 00:00:00 2001 From: Tomeu Vizoso Date: Thu, 12 Feb 2015 09:41:56 +0100 Subject: ASoC: tegra: Add sink for the internal mic to tegra_max98090 Also adds a control for the pin of the internal mic, so userspace can apply policy when the state of the external mic jack changes. Signed-off-by: Tomeu Vizoso Acked-by: Stephen Warren Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt | 1 + sound/soc/tegra/tegra_max98090.c | 2 ++ 2 files changed, 3 insertions(+) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt index c949abc2992f..c3495beba358 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt @@ -18,6 +18,7 @@ Required properties: * Headphones * Speakers * Mic Jack + * Int Mic - nvidia,i2s-controller : The phandle of the Tegra I2S controller that's connected to the CODEC. diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index 8df71a436f11..29ea87cd852e 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -133,11 +133,13 @@ static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), }; static const struct snd_kcontrol_new tegra_max98090_controls[] = { SOC_DAPM_PIN_SWITCH("Headphones"), SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Int Mic"), }; static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) -- cgit v1.2.3-59-g8ed1b From dd3001490834e10615d9eb229b3e9bbcc0070541 Mon Sep 17 00:00:00 2001 From: Tomeu Vizoso Date: Thu, 12 Feb 2015 09:41:57 +0100 Subject: ASoC: tegra: Add control for the Mic Jack pin So userspace can enable and disable the external microphone. Signed-off-by: Tomeu Vizoso Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_max98090.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index 29ea87cd852e..1f20c2c40a5a 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -139,6 +139,7 @@ static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = { static const struct snd_kcontrol_new tegra_max98090_controls[] = { SOC_DAPM_PIN_SWITCH("Headphones"), SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), SOC_DAPM_PIN_SWITCH("Int Mic"), }; -- cgit v1.2.3-59-g8ed1b From 0004defd4e44d81966b0c4164c2ee01f20ab357b Mon Sep 17 00:00:00 2001 From: Vishal Thanki Date: Tue, 3 Mar 2015 18:59:00 +0530 Subject: ASoC: simple-card: Add a NULL pointer check in asoc_simple_card_dai_link_of Make sure devm_kzalloc() succeeds. Signed-off-by: Vishal Thanki Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index f7c6734bd5da..fb550b5869d2 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -372,6 +372,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, strlen(dai_link->cpu_dai_name) + strlen(dai_link->codec_dai_name) + 2, GFP_KERNEL); + if (!name) { + ret = -ENOMEM; + goto dai_link_of_err; + } + sprintf(name, "%s-%s", dai_link->cpu_dai_name, dai_link->codec_dai_name); dai_link->name = dai_link->stream_name = name; -- cgit v1.2.3-59-g8ed1b From c472b93990e02c31f02322ddf0fdd9d571169310 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:16 +0100 Subject: ASoC: sn95031: Pass CODEC to sn95031_jack_detection() The sn95031 driver currently gets the CODEC implicitly from the jack that is passed to sn95031_jack_detection(). But the codec field is going to be removed from the snd_soc_jack struct, so refactor things to pass the CODEC explicitly. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 14 ++++++++------ sound/soc/codecs/sn95031.h | 3 ++- sound/soc/intel/mfld_machine.c | 13 +++++++------ 3 files changed, 17 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 47b257e41809..1e5d2643c286 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -783,19 +783,21 @@ static inline void sn95031_enable_jack_btn(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_BTNCTRL2, 0x01); } -static int sn95031_get_headset_state(struct snd_soc_jack *mfld_jack) +static int sn95031_get_headset_state(struct snd_soc_codec *codec, + struct snd_soc_jack *mfld_jack) { - int micbias = sn95031_get_mic_bias(mfld_jack->codec); + int micbias = sn95031_get_mic_bias(codec); int jack_type = snd_soc_jack_get_type(mfld_jack, micbias); pr_debug("jack type detected = %d\n", jack_type); if (jack_type == SND_JACK_HEADSET) - sn95031_enable_jack_btn(mfld_jack->codec); + sn95031_enable_jack_btn(codec); return jack_type; } -void sn95031_jack_detection(struct mfld_jack_data *jack_data) +void sn95031_jack_detection(struct snd_soc_codec *codec, + struct mfld_jack_data *jack_data) { unsigned int status; unsigned int mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_HEADSET; @@ -809,11 +811,11 @@ void sn95031_jack_detection(struct mfld_jack_data *jack_data) status = SND_JACK_HEADSET | SND_JACK_BTN_1; } else if (jack_data->intr_id & 0x4) { pr_debug("headset or headphones inserted\n"); - status = sn95031_get_headset_state(jack_data->mfld_jack); + status = sn95031_get_headset_state(codec, jack_data->mfld_jack); } else if (jack_data->intr_id & 0x8) { pr_debug("headset or headphones removed\n"); status = 0; - sn95031_disable_jack_btn(jack_data->mfld_jack->codec); + sn95031_disable_jack_btn(codec); } else { pr_err("unidentified interrupt\n"); return; diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h index 20376d234fb8..7651fe4e6a45 100644 --- a/sound/soc/codecs/sn95031.h +++ b/sound/soc/codecs/sn95031.h @@ -127,6 +127,7 @@ struct mfld_jack_data { struct snd_soc_jack *mfld_jack; }; -extern void sn95031_jack_detection(struct mfld_jack_data *jack_data); +extern void sn95031_jack_detection(struct snd_soc_codec *codec, + struct mfld_jack_data *jack_data); #endif diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c index 90b7a57713a0..d22b44db824e 100644 --- a/sound/soc/intel/mfld_machine.c +++ b/sound/soc/intel/mfld_machine.c @@ -228,10 +228,13 @@ static void mfld_jack_check(unsigned int intr_status) { struct mfld_jack_data jack_data; + if (!mfld_codec) + return; + jack_data.mfld_jack = &mfld_jack; jack_data.intr_id = intr_status; - sn95031_jack_detection(&jack_data); + sn95031_jack_detection(mfld_codec, &jack_data); /* TODO: add american headset detection post gpiolib support */ } @@ -240,8 +243,6 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) struct snd_soc_dapm_context *dapm = &runtime->card->dapm; int ret_val; - mfld_codec = runtime->codec; - /* default is earpiece pin, userspace sets it explcitly */ snd_soc_dapm_disable_pin(dapm, "Headphones"); /* default is lineout NC, userspace sets it explcitly */ @@ -254,7 +255,7 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_disable_pin(dapm, "LINEINR"); /* Headset and button jack detection */ - ret_val = snd_soc_jack_new(mfld_codec, "Intel(R) MID Audio Jack", + ret_val = snd_soc_jack_new(runtime->codec, "Intel(R) MID Audio Jack", SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack); if (ret_val) { @@ -275,6 +276,8 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) return ret_val; } + mfld_codec = runtime->codec; + /* we want to check if anything is inserted at boot, * so send a fake event to codec and it will read adc * to find if anything is there or not */ @@ -359,8 +362,6 @@ static irqreturn_t snd_mfld_jack_detection(int irq, void *data) { struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data; - if (mfld_jack.codec == NULL) - return IRQ_HANDLED; mfld_jack_check(mc_drv_ctx->interrupt_status); return IRQ_HANDLED; -- cgit v1.2.3-59-g8ed1b From 970939964c26db4643f58d4e84487821962e0b65 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:17 +0100 Subject: ASoC: Allow to register jacks at the card level Jacks are typically card level elements, but are currently registered with a CODEC. When it was originally introduced snd_soc_jack_new() took a snd_soc_card as its parameter, but at that time DAPM was only implemented at the CODEC level and there was only one CODEC per card. This made it clear which CODEC to use for the jack DAPM operations. But the multi-component patchset added support for having multiple CODECs per card and with it the API was updated to register jacks with a specific CODEC instance instead. Subsequently DAPM support at the card level has been introduced, but the snd_soc_jack_new() API has so remained unchanged. This leaves us with the issue that the DAPM pins that are managed by the jack detection logic usually are part of the card DAPM context but are accessed through a CODEC DAPM context. Currently this works fine, but might break in the future if we take a more hierarchical approach to DAPM contexts. Furthermore with componentization progressing systems that do not register a snd_soc_codec might appear, while these system may still want to able to register a jack. This patch addresses these issues by adding a new function called snd_soc_card_jack_new() that can be used to register jacks with the card rather than a CODEC. This new function is mostly identical to snd_soc_jack_new() except that it additionally allows to directly specify the DAPM pins associated with the jack. This was done since most users of snd_soc_jack_new() typically call snd_soc_jack_add_pins() right after it, which is not necessary with the new API and allows to reduce the amount of boiler plate code. The old snd_soc_jack_new() is re-implemented as a wrapper around snd_soc_card_jack_new(). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 28 +++++++++++++++++++++++++--- sound/soc/soc-jack.c | 42 ++++++++++++++++++++++++++---------------- 2 files changed, 51 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade195628..99d9ce272209 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -450,8 +450,10 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); /* Jack reporting */ -int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, - struct snd_soc_jack *jack); +int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type, + struct snd_soc_jack *jack, struct snd_soc_jack_pin *pins, + unsigned int num_pins); + void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask); int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, struct snd_soc_jack_pin *pins); @@ -659,7 +661,7 @@ struct snd_soc_jack_gpio { struct snd_soc_jack { struct mutex mutex; struct snd_jack *jack; - struct snd_soc_codec *codec; + struct snd_soc_card *card; struct list_head pins; int status; struct blocking_notifier_head notifier; @@ -1482,6 +1484,26 @@ static inline struct snd_soc_platform *snd_soc_kcontrol_platform( return snd_soc_component_to_platform(snd_soc_kcontrol_component(kcontrol)); } +/** + * snd_soc_jack_new - Create a new jack + * @codec: ASoC CODEC + * @id: an identifying string for this jack + * @type: a bitmask of enum snd_jack_type values that can be detected by + * this jack + * @jack: structure to use for the jack + * + * Creates a new jack object. + * + * Returns zero if successful, or a negative error code on failure. + * On success jack will be initialised. + */ +static inline int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, + int type, struct snd_soc_jack *jack) +{ + return snd_soc_card_jack_new(codec->component.card, id, type, jack, + NULL, 0); +} + int snd_soc_util_init(void); void snd_soc_util_exit(void); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 4380dcc064a5..9f60c25c4568 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -22,30 +22,42 @@ #include /** - * snd_soc_jack_new - Create a new jack - * @codec: ASoC codec + * snd_soc_card_jack_new - Create a new jack + * @card: ASoC card * @id: an identifying string for this jack * @type: a bitmask of enum snd_jack_type values that can be detected by * this jack * @jack: structure to use for the jack + * @pins: Array of jack pins to be added to the jack or NULL + * @num_pins: Number of elements in the @pins array * * Creates a new jack object. * * Returns zero if successful, or a negative error code on failure. * On success jack will be initialised. */ -int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, - struct snd_soc_jack *jack) +int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type, + struct snd_soc_jack *jack, struct snd_soc_jack_pin *pins, + unsigned int num_pins) { + int ret; + mutex_init(&jack->mutex); - jack->codec = codec; + jack->card = card; INIT_LIST_HEAD(&jack->pins); INIT_LIST_HEAD(&jack->jack_zones); BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier); - return snd_jack_new(codec->component.card->snd_card, id, type, &jack->jack); + ret = snd_jack_new(card->snd_card, id, type, &jack->jack); + if (ret) + return ret; + + if (num_pins) + return snd_soc_jack_add_pins(jack, num_pins, pins); + + return 0; } -EXPORT_SYMBOL_GPL(snd_soc_jack_new); +EXPORT_SYMBOL_GPL(snd_soc_card_jack_new); /** * snd_soc_jack_report - Report the current status for a jack @@ -63,7 +75,6 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_new); */ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) { - struct snd_soc_codec *codec; struct snd_soc_dapm_context *dapm; struct snd_soc_jack_pin *pin; unsigned int sync = 0; @@ -74,8 +85,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) if (!jack) return; - codec = jack->codec; - dapm = &codec->dapm; + dapm = &jack->card->dapm; mutex_lock(&jack->mutex); @@ -175,12 +185,12 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, for (i = 0; i < count; i++) { if (!pins[i].pin) { - dev_err(jack->codec->dev, "ASoC: No name for pin %d\n", + dev_err(jack->card->dev, "ASoC: No name for pin %d\n", i); return -EINVAL; } if (!pins[i].mask) { - dev_err(jack->codec->dev, "ASoC: No mask for pin %d" + dev_err(jack->card->dev, "ASoC: No mask for pin %d" " (%s)\n", i, pins[i].pin); return -EINVAL; } @@ -260,7 +270,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) static irqreturn_t gpio_handler(int irq, void *data) { struct snd_soc_jack_gpio *gpio = data; - struct device *dev = gpio->jack->codec->component.card->dev; + struct device *dev = gpio->jack->card->dev; trace_snd_soc_jack_irq(gpio->name); @@ -299,7 +309,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, for (i = 0; i < count; i++) { if (!gpios[i].name) { - dev_err(jack->codec->dev, + dev_err(jack->card->dev, "ASoC: No name for gpio at index %d\n", i); ret = -EINVAL; goto undo; @@ -320,7 +330,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, } else { /* legacy GPIO number */ if (!gpio_is_valid(gpios[i].gpio)) { - dev_err(jack->codec->dev, + dev_err(jack->card->dev, "ASoC: Invalid gpio %d\n", gpios[i].gpio); ret = -EINVAL; @@ -350,7 +360,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, if (gpios[i].wake) { ret = irq_set_irq_wake(gpiod_to_irq(gpios[i].desc), 1); if (ret != 0) - dev_err(jack->codec->dev, + dev_err(jack->card->dev, "ASoC: Failed to mark GPIO at index %d as wake source: %d\n", i, ret); } -- cgit v1.2.3-59-g8ed1b From 386669fcec85a16cb81cd19236abe76abe0f1fc1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:18 +0100 Subject: ASoC: simple-card: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index f7c6734bd5da..b8ee47b7ba9c 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -176,11 +176,11 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) return ret; if (gpio_is_valid(priv->gpio_hp_det)) { - snd_soc_jack_new(codec->codec, "Headphones", SND_JACK_HEADPHONE, - &simple_card_hp_jack); - snd_soc_jack_add_pins(&simple_card_hp_jack, - ARRAY_SIZE(simple_card_hp_jack_pins), - simple_card_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphones", + SND_JACK_HEADPHONE, + &simple_card_hp_jack, + simple_card_hp_jack_pins, + ARRAY_SIZE(simple_card_hp_jack_pins)); simple_card_hp_jack_gpio.gpio = priv->gpio_hp_det; simple_card_hp_jack_gpio.invert = priv->gpio_hp_det_invert; @@ -189,11 +189,11 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) } if (gpio_is_valid(priv->gpio_mic_det)) { - snd_soc_jack_new(codec->codec, "Mic Jack", SND_JACK_MICROPHONE, - &simple_card_mic_jack); - snd_soc_jack_add_pins(&simple_card_mic_jack, - ARRAY_SIZE(simple_card_mic_jack_pins), - simple_card_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", + SND_JACK_MICROPHONE, + &simple_card_mic_jack, + simple_card_mic_jack_pins, + ARRAY_SIZE(simple_card_mic_jack_pins)); simple_card_mic_jack_gpio.gpio = priv->gpio_mic_det; simple_card_mic_jack_gpio.invert = priv->gpio_mic_det_invert; snd_soc_jack_add_gpios(&simple_card_mic_jack, 1, -- cgit v1.2.3-59-g8ed1b From 27cb64b474516421001932d966ca3184795d4e29 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:19 +0100 Subject: ASoC: imx-es8328: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/fsl/imx-es8328.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index f8cf10e16ce9..20e7400e2611 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.c @@ -53,9 +53,9 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) /* Headphone jack detection */ if (gpio_is_valid(data->jack_gpio)) { - ret = snd_soc_jack_new(rtd->codec, "Headphone", - SND_JACK_HEADPHONE | SND_JACK_BTN_0, - &headset_jack); + ret = snd_soc_card_jack_new(rtd->card, "Headphone", + SND_JACK_HEADPHONE | SND_JACK_BTN_0, + &headset_jack, NULL, 0); if (ret) return ret; -- cgit v1.2.3-59-g8ed1b From 47ec96d4ca7e4a7b9b8b115a10d59e89f794ef95 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:20 +0100 Subject: ASoC: wm1133-ev: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/fsl/wm1133-ev1.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index a958937ab405..0653aa83c927 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -205,16 +205,14 @@ static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; /* Headphone jack detection */ - snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack); - snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), - hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE, + &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins)); wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE); /* Microphone jack detection */ - snd_soc_jack_new(codec, "Microphone", - SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack); - snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), - mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Microphone", + SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack, + mic_jack_pins, ARRAY_SIZE(mic_jack_pins)); wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE, SND_JACK_BTN_0); -- cgit v1.2.3-59-g8ed1b From 85c85e5d6d579a5ff8b5471c4e753946eedbf788 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:21 +0100 Subject: ASoC: broadwell: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/intel/broadwell.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 9cf7d01479ad..9effa3da982f 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -80,15 +80,9 @@ static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; int ret = 0; - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset); - - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&broadwell_headset, - ARRAY_SIZE(broadwell_headset_pins), - broadwell_headset_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset, + broadwell_headset_pins, ARRAY_SIZE(broadwell_headset_pins)); if (ret) return ret; -- cgit v1.2.3-59-g8ed1b From e0f7dd9d88f4c151aeca45d290e171d907249888 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:22 +0100 Subject: ASoC: byt-max98090: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/intel/byt-max98090.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index 9832afe7d22c..d8b1f038da1c 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -84,7 +84,6 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = { static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) { int ret; - struct snd_soc_codec *codec = runtime->codec; struct snd_soc_card *card = runtime->card; struct byt_max98090_private *drv = snd_soc_card_get_drvdata(card); struct snd_soc_jack *jack = &drv->jack; @@ -100,13 +99,9 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) } /* Enable jack detection */ - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_LINEOUT | SND_JACK_HEADSET, jack); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + ret = snd_soc_card_jack_new(runtime->card, "Headset", + SND_JACK_LINEOUT | SND_JACK_HEADSET, jack, + hs_jack_pins, ARRAY_SIZE(hs_jack_pins)); if (ret) return ret; -- cgit v1.2.3-59-g8ed1b From fb1edb4b68a829619bcd50a0c23c557000d0d638 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:23 +0100 Subject: ASoC: cht_bsw_rt5645: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/intel/cht_bsw_rt5645.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/cht_bsw_rt5645.c index bd29617a9ab9..0bfca2192ca0 100644 --- a/sound/soc/intel/cht_bsw_rt5645.c +++ b/sound/soc/intel/cht_bsw_rt5645.c @@ -169,17 +169,17 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) return ret; } - ret = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, - &ctx->hp_jack); + ret = snd_soc_card_jack_new(runtime->card, "Headphone Jack", + SND_JACK_HEADPHONE, &ctx->hp_jack, + NULL, 0); if (ret) { dev_err(runtime->dev, "HP jack creation failed %d\n", ret); return ret; } - ret = snd_soc_jack_new(codec, "Mic Jack", - SND_JACK_MICROPHONE, - &ctx->mic_jack); + ret = snd_soc_card_jack_new(runtime->card, "Mic Jack", + SND_JACK_MICROPHONE, &ctx->mic_jack, + NULL, 0); if (ret) { dev_err(runtime->dev, "Mic jack creation failed %d\n", ret); return ret; -- cgit v1.2.3-59-g8ed1b From af13cbc1a288d3921f1af739da84371e6c53aea3 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:24 +0100 Subject: ASoC: mfld_machine: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/mfld_machine.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c index d22b44db824e..49c09a0add79 100644 --- a/sound/soc/intel/mfld_machine.c +++ b/sound/soc/intel/mfld_machine.c @@ -255,20 +255,15 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_disable_pin(dapm, "LINEINR"); /* Headset and button jack detection */ - ret_val = snd_soc_jack_new(runtime->codec, "Intel(R) MID Audio Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1, &mfld_jack); + ret_val = snd_soc_card_jack_new(runtime->card, + "Intel(R) MID Audio Jack", SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack, + mfld_jack_pins, ARRAY_SIZE(mfld_jack_pins)); if (ret_val) { pr_err("jack creation failed\n"); return ret_val; } - ret_val = snd_soc_jack_add_pins(&mfld_jack, - ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins); - if (ret_val) { - pr_err("adding jack pins failed\n"); - return ret_val; - } ret_val = snd_soc_jack_add_zones(&mfld_jack, ARRAY_SIZE(mfld_zones), mfld_zones); if (ret_val) { -- cgit v1.2.3-59-g8ed1b From df8c66189dd42f719c75800a526bdc901f300f41 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:25 +0100 Subject: ASoC: ams-deltea: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/omap/ams-delta.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 706613077c15..16cc95fa4573 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -479,8 +479,8 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) /* Add hook switch - can be used to control the codec from userspace * even if line discipline fails */ - ret = snd_soc_jack_new(rtd->codec, "hook_switch", - SND_JACK_HEADSET, &ams_delta_hook_switch); + ret = snd_soc_card_jack_new(card, "hook_switch", SND_JACK_HEADSET, + &ams_delta_hook_switch, NULL, 0); if (ret) dev_warn(card->dev, "Failed to allocate resources for hook switch, " -- cgit v1.2.3-59-g8ed1b From 25649592cfa6c210c9f86670472b864782c8d677 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:26 +0100 Subject: ASoC: omap-abe-twl6040: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-abe-twl6040.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index b9c65f1ad5a8..0843a68f277c 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -182,17 +182,17 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) /* Headset jack detection only if it is supported */ if (priv->jack_detection) { - ret = snd_soc_jack_new(codec, "Headset Jack", - SND_JACK_HEADSET, &hs_jack); + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET, &hs_jack, + hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (ret) return ret; - ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); } - return ret; + return 0; } static const struct snd_soc_dapm_route dmic_audio_map[] = { -- cgit v1.2.3-59-g8ed1b From da21cf6d65283680247da74c3d03f7e5cdfb40d1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:27 +0100 Subject: ASoC: omap-twl4030: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-twl4030.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index fb1f6bb87cd4..3673ada43bfb 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -170,14 +170,10 @@ static int omap_twl4030_init(struct snd_soc_pcm_runtime *rtd) if (priv->jack_detect > 0) { hs_jack_gpios[0].gpio = priv->jack_detect; - ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, - &priv->hs_jack); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&priv->hs_jack, - ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET, &priv->hs_jack, + hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (ret) return ret; -- cgit v1.2.3-59-g8ed1b From 753d45e6b886c93a2a8a88eddaca345643a87f4e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:28 +0100 Subject: ASoC: rx51: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/omap/rx51.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 7f299357c2d2..c2ddf0fbfa28 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -311,9 +311,9 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) } /* AV jack detection */ - err = snd_soc_jack_new(codec, "AV Jack", - SND_JACK_HEADSET | SND_JACK_VIDEOOUT, - &rx51_av_jack); + err = snd_soc_card_jack_new(rtd->card, "AV Jack", + SND_JACK_HEADSET | SND_JACK_VIDEOOUT, + &rx51_av_jack, NULL, 0); if (err) { dev_err(card->dev, "Failed to add AV Jack\n"); return err; -- cgit v1.2.3-59-g8ed1b From f7a4433b498384f0e300c51b654910f3e03b5ca6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:29 +0100 Subject: ASoC: hx4700: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/hx4700.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 73eb5ddf9753..9f8be7cd567e 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -126,17 +126,12 @@ static const struct snd_soc_dapm_route hx4700_audio_map[] = { */ static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; int err; /* Jack detection API stuff */ - err = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, &hs_jack); - if (err) - return err; - - err = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pin), - hs_jack_pin); + err = snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &hs_jack, hs_jack_pin, + ARRAY_SIZE(hs_jack_pin)); if (err) return err; -- cgit v1.2.3-59-g8ed1b From bc1e2e06a07ad4c0c021165b34fa8259bdf4d8c6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:30 +0100 Subject: ASoC: palm27x: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/palm27x.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 910336c5ebeb..c20bbc042425 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -75,17 +75,12 @@ static struct snd_soc_card palm27x_asoc; static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; int err; /* Jack detection API stuff */ - err = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, &hs_jack); - if (err) - return err; - - err = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + err = snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &hs_jack, hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (err) return err; -- cgit v1.2.3-59-g8ed1b From 3b14125bc553a0fe091a5d43a22be41cdc43b156 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:31 +0100 Subject: ASoC: ttc-dkb: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/ttc-dkb.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c index 5001dbb9b257..1753c7d9e760 100644 --- a/sound/soc/pxa/ttc-dkb.c +++ b/sound/soc/pxa/ttc-dkb.c @@ -78,15 +78,12 @@ static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; /* Headset jack detection */ - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE - | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, - &hs_jack); - snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); - snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE, - &mic_jack); - snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), - mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, + &hs_jack, hs_jack_pins, ARRAY_SIZE(hs_jack_pins)); + snd_soc_card_jack_new(rtd->card, "Microphone Jack", SND_JACK_MICROPHONE, + &mic_jack, mic_jack_pins, + ARRAY_SIZE(mic_jack_pins)); /* headphone, microphone detection & headset short detection */ pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE, -- cgit v1.2.3-59-g8ed1b From d30d141f9cb7eb9fb3f03af11146dc0d2b393ff2 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:32 +0100 Subject: ASoC: z2: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/z2.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 76ccb172d0a7..bcbfbe8303f7 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -143,13 +143,9 @@ static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "MONO1"); /* Jack detection API stuff */ - ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, - &hs_jack); - if (ret) - goto err; - - ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET, + &hs_jack, hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (ret) goto err; -- cgit v1.2.3-59-g8ed1b From dfe11f282c61808f7140d9dd741f7e54cf97cda6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:33 +0100 Subject: ASoC: h1980_uda1380: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/h1940_uda1380.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 59b044255b78..c72e9fb26658 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -162,13 +162,8 @@ static struct platform_device *s3c24xx_snd_device; static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; - - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &hp_jack); - - snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), - hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, + &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins)); snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), hp_jack_gpios); -- cgit v1.2.3-59-g8ed1b From 39ec5109d6089e1acd04b51b9df5349f5b8a7f5c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:34 +0100 Subject: ASoC: littlemill: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/littlemill.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 141519c21e21..31a820eb0ac3 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -260,12 +260,12 @@ static int littlemill_late_probe(struct snd_soc_card *card) if (ret < 0) return ret; - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_MECHANICAL | - SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3 | - SND_JACK_BTN_4 | SND_JACK_BTN_5, - &littlemill_headset); + ret = snd_soc_card_jack_new(card, "Headset", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | + SND_JACK_BTN_4 | SND_JACK_BTN_5, + &littlemill_headset, NULL, 0); if (ret) return ret; -- cgit v1.2.3-59-g8ed1b From f97e0eacf2b5d9c1a470e53df60519d555ac5a75 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:35 +0100 Subject: ASoC: lowland: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/lowland.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index 243dea7ba38f..5f156093101e 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -56,16 +56,10 @@ static int lowland_wm5100_init(struct snd_soc_pcm_runtime *rtd) return ret; } - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_LINEOUT | SND_JACK_HEADSET | - SND_JACK_BTN_0, - &lowland_headset); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&lowland_headset, - ARRAY_SIZE(lowland_headset_pins), - lowland_headset_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset", SND_JACK_LINEOUT | + SND_JACK_HEADSET | SND_JACK_BTN_0, + &lowland_headset, lowland_headset_pins, + ARRAY_SIZE(lowland_headset_pins)); if (ret) return ret; -- cgit v1.2.3-59-g8ed1b From e9c9a723eea5102fa6adedf454e02fff6201a3c3 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:36 +0100 Subject: ASoC: rx1950_uda1380: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/rx1950_uda1380.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 873f2cb4bebe..35e37c457f1f 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -211,13 +211,8 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream, static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; - - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &hp_jack); - - snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), - hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, + &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins)); snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), hp_jack_gpios); -- cgit v1.2.3-59-g8ed1b From 55b2ed2d9dd8c611837f34ca29df881eb0a1de8d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:37 +0100 Subject: ASoC: smartq: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/smartq_wm8987.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 8291d2a5f152..dfbe2db1c407 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -151,13 +151,10 @@ static int smartq_wm8987_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); /* Headphone jack detection */ - err = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, &smartq_jack); - if (err) - return err; - - err = snd_soc_jack_add_pins(&smartq_jack, ARRAY_SIZE(smartq_jack_pins), - smartq_jack_pins); + err = snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &smartq_jack, + smartq_jack_pins, + ARRAY_SIZE(smartq_jack_pins)); if (err) return err; -- cgit v1.2.3-59-g8ed1b From 663976ad478b50664353fdf19a5a3dcad3cb4e22 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:38 +0100 Subject: ASoC: speyside: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/speyside.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 5ec7c52282f2..2dcb988bdff2 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -153,16 +153,10 @@ static int speyside_wm8996_init(struct snd_soc_pcm_runtime *rtd) pr_err("Failed to request HP_SEL GPIO: %d\n", ret); gpio_direction_output(WM8996_HPSEL_GPIO, speyside_jack_polarity); - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_LINEOUT | SND_JACK_HEADSET | - SND_JACK_BTN_0, - &speyside_headset); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&speyside_headset, - ARRAY_SIZE(speyside_headset_pins), - speyside_headset_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset", SND_JACK_LINEOUT | + SND_JACK_HEADSET | SND_JACK_BTN_0, + &speyside_headset, speyside_headset_pins, + ARRAY_SIZE(speyside_headset_pins)); if (ret) return ret; -- cgit v1.2.3-59-g8ed1b From 3fd94f37da000a2b562a3f4e6c553b7ab1ad9e19 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:39 +0100 Subject: ASoC: tobermory: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/tobermory.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index 9c80506527c4..85ccfb7188cb 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -179,15 +179,10 @@ static int tobermory_late_probe(struct snd_soc_card *card) if (ret < 0) return ret; - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, - &tobermory_headset); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&tobermory_headset, - ARRAY_SIZE(tobermory_headset_pins), - tobermory_headset_pins); + ret = snd_soc_card_jack_new(card, "Headset", SND_JACK_HEADSET | + SND_JACK_BTN_0, &tobermory_headset, + tobermory_headset_pins, + ARRAY_SIZE(tobermory_headset_pins)); if (ret) return ret; -- cgit v1.2.3-59-g8ed1b From 12cc6d1dca4d3a9e929090cb0cf9ef452f414518 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:40 +0100 Subject: ASoC: tegra_alc5632: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_alc5632.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 769aca2fc5f5..6dcd06a966c7 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -106,11 +106,10 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(rtd->card); - snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, - &tegra_alc5632_hs_jack); - snd_soc_jack_add_pins(&tegra_alc5632_hs_jack, - ARRAY_SIZE(tegra_alc5632_hs_jack_pins), - tegra_alc5632_hs_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET, + &tegra_alc5632_hs_jack, + tegra_alc5632_hs_jack_pins, + ARRAY_SIZE(tegra_alc5632_hs_jack_pins)); if (gpio_is_valid(machine->gpio_hp_det)) { tegra_alc5632_hp_jack_gpio.gpio = machine->gpio_hp_det; -- cgit v1.2.3-59-g8ed1b From d020e77c61b8a9d563d205cfcec7e71090d1377d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:41 +0100 Subject: ASoC: tegra_max98090: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_max98090.c | 22 ++++++++++------------ 1 file changed, 10 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index af3fb997b752..6760f0ebc133 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -141,16 +141,14 @@ static const struct snd_kcontrol_new tegra_max98090_controls[] = { static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; struct tegra_max98090 *machine = snd_soc_card_get_drvdata(rtd->card); if (gpio_is_valid(machine->gpio_hp_det)) { - snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, - &tegra_max98090_hp_jack); - snd_soc_jack_add_pins(&tegra_max98090_hp_jack, - ARRAY_SIZE(tegra_max98090_hp_jack_pins), - tegra_max98090_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphones", + SND_JACK_HEADPHONE, + &tegra_max98090_hp_jack, + tegra_max98090_hp_jack_pins, + ARRAY_SIZE(tegra_max98090_hp_jack_pins)); tegra_max98090_hp_jack_gpio.gpio = machine->gpio_hp_det; snd_soc_jack_add_gpios(&tegra_max98090_hp_jack, @@ -159,11 +157,11 @@ static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) } if (gpio_is_valid(machine->gpio_mic_det)) { - snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, - &tegra_max98090_mic_jack); - snd_soc_jack_add_pins(&tegra_max98090_mic_jack, - ARRAY_SIZE(tegra_max98090_mic_jack_pins), - tegra_max98090_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", + SND_JACK_MICROPHONE, + &tegra_max98090_mic_jack, + tegra_max98090_mic_jack_pins, + ARRAY_SIZE(tegra_max98090_mic_jack_pins)); tegra_max98090_mic_jack_gpio.gpio = machine->gpio_mic_det; snd_soc_jack_add_gpios(&tegra_max98090_mic_jack, -- cgit v1.2.3-59-g8ed1b From 00eafe3b1b191c9b2611b74c03e1b573ae257b1e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:42 +0100 Subject: ASoC: tegra_rt5640: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_rt5640.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index ed759a3076b8..773daecaa5e8 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -108,15 +108,11 @@ static const struct snd_kcontrol_new tegra_rt5640_controls[] = { static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(rtd->card); - snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, - &tegra_rt5640_hp_jack); - snd_soc_jack_add_pins(&tegra_rt5640_hp_jack, - ARRAY_SIZE(tegra_rt5640_hp_jack_pins), - tegra_rt5640_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphones", SND_JACK_HEADPHONE, + &tegra_rt5640_hp_jack, tegra_rt5640_hp_jack_pins, + ARRAY_SIZE(tegra_rt5640_hp_jack_pins)); if (gpio_is_valid(machine->gpio_hp_det)) { tegra_rt5640_hp_jack_gpio.gpio = machine->gpio_hp_det; -- cgit v1.2.3-59-g8ed1b From 783b1e7948010ded40eba784b558d86d72ae2ef4 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:43 +0100 Subject: ASoC: tegra_rt5677: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_rt5677.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index e4cf978a6e3a..68d8b67e79c1 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -146,10 +146,9 @@ static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(rtd->card); - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &tegra_rt5677_hp_jack); - snd_soc_jack_add_pins(&tegra_rt5677_hp_jack, 1, - &tegra_rt5677_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, + &tegra_rt5677_hp_jack, + &tegra_rt5677_hp_jack_pins, 1); if (gpio_is_valid(machine->gpio_hp_det)) { tegra_rt5677_hp_jack_gpio.gpio = machine->gpio_hp_det; @@ -158,10 +157,9 @@ static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd) } - snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, - &tegra_rt5677_mic_jack); - snd_soc_jack_add_pins(&tegra_rt5677_mic_jack, 1, - &tegra_rt5677_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_rt5677_mic_jack, + &tegra_rt5677_mic_jack_pins, 1); if (gpio_is_valid(machine->gpio_mic_present)) { tegra_rt5677_mic_jack_gpio.gpio = machine->gpio_mic_present; -- cgit v1.2.3-59-g8ed1b From 7ba8cbb2f0fd9ff232fa19159e2646bf64135126 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:44 +0100 Subject: ASoC: tegra_wm8903: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_wm8903.c | 18 ++++++++---------- 1 file changed, 8 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index e52420dae2b4..4a95b70f0cf0 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -177,21 +177,19 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) if (gpio_is_valid(machine->gpio_hp_det)) { tegra_wm8903_hp_jack_gpio.gpio = machine->gpio_hp_det; - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &tegra_wm8903_hp_jack); - snd_soc_jack_add_pins(&tegra_wm8903_hp_jack, - ARRAY_SIZE(tegra_wm8903_hp_jack_pins), - tegra_wm8903_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &tegra_wm8903_hp_jack, + tegra_wm8903_hp_jack_pins, + ARRAY_SIZE(tegra_wm8903_hp_jack_pins)); snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack, 1, &tegra_wm8903_hp_jack_gpio); } - snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, - &tegra_wm8903_mic_jack); - snd_soc_jack_add_pins(&tegra_wm8903_mic_jack, - ARRAY_SIZE(tegra_wm8903_mic_jack_pins), - tegra_wm8903_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_wm8903_mic_jack, + tegra_wm8903_mic_jack_pins, + ARRAY_SIZE(tegra_wm8903_mic_jack_pins)); wm8903_mic_detect(codec, &tegra_wm8903_mic_jack, SND_JACK_MICROPHONE, 0); -- cgit v1.2.3-59-g8ed1b From 4c03a5ebc7f75e98b32591d1d2c6758c811dcbef Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 Mar 2015 16:45:17 +0200 Subject: ASoC: davinci: Select SND_EDMA_SOC when SND_DAVINCI_SOC is enabled edma-pcm going to replace davinci-pcm as platform driver for daVinci platform. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 2b81ca418d2a..eae4e229f341 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -1,10 +1,11 @@ config SND_DAVINCI_SOC tristate "SoC Audio for TI DAVINCI" depends on ARCH_DAVINCI + select SND_EDMA_SOC config SND_EDMA_SOC tristate "SoC Audio for Texas Instruments chips using eDMA (AM33XX/43XX)" - depends on SOC_AM33XX || SOC_AM43XX + depends on SOC_AM33XX || SOC_AM43XX || ARCH_DAVINCI select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M here if you want audio support for TI SoC which uses eDMA. -- cgit v1.2.3-59-g8ed1b From 257ade78b6019cf1570c1239894a7a6a549768e1 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 Mar 2015 16:45:18 +0200 Subject: ASoC: davinci-i2s: Convert to use edma-pcm The edma-pcm can replace the old davinci-pcm as platform driver. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-i2s.c | 67 ++++++++++++++++------------------------- 1 file changed, 26 insertions(+), 41 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 15fb28fc8e1b..56cb4d95637d 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -23,8 +23,9 @@ #include #include #include +#include -#include "davinci-pcm.h" +#include "edma-pcm.h" #include "davinci-i2s.h" @@ -122,7 +123,8 @@ static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { struct davinci_mcbsp_dev { struct device *dev; - struct davinci_pcm_dma_params dma_params[2]; + struct snd_dmaengine_dai_dma_data dma_data[2]; + int dma_request[2]; void __iomem *base; #define MOD_DSP_A 0 #define MOD_DSP_B 1 @@ -419,8 +421,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); - struct davinci_pcm_dma_params *dma_params = - &dev->dma_params[substream->stream]; struct snd_interval *i = NULL; int mcbsp_word_length, master; unsigned int rcr, xcr, srgr, clk_div, freq, framesize; @@ -532,8 +532,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } } - dma_params->acnt = dma_params->data_type = data_type[fmt]; - dma_params->fifo_level = 0; mcbsp_word_length = asp_word_length[fmt]; switch (master) { @@ -600,15 +598,6 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int davinci_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); - - snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); - return 0; -} - static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -620,7 +609,6 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 static const struct snd_soc_dai_ops davinci_i2s_dai_ops = { - .startup = davinci_i2s_startup, .shutdown = davinci_i2s_shutdown, .prepare = davinci_i2s_prepare, .trigger = davinci_i2s_trigger, @@ -630,7 +618,18 @@ static const struct snd_soc_dai_ops davinci_i2s_dai_ops = { }; +static int davinci_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); + + dai->playback_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dai->capture_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + + return 0; +} + static struct snd_soc_dai_driver davinci_i2s_dai = { + .probe = davinci_i2s_dai_probe, .playback = { .channels_min = 2, .channels_max = 2, @@ -651,11 +650,9 @@ static const struct snd_soc_component_driver davinci_i2s_component = { static int davinci_i2s_probe(struct platform_device *pdev) { - struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea, *res; - enum dma_event_q asp_chan_q = EVENTQ_0; - enum dma_event_q ram_chan_q = EVENTQ_1; + int *dma; int ret; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -676,22 +673,6 @@ static int davinci_i2s_probe(struct platform_device *pdev) GFP_KERNEL); if (!dev) return -ENOMEM; - if (pdata) { - dev->enable_channel_combine = pdata->enable_channel_combine; - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].sram_size = - pdata->sram_size_playback; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = - pdata->sram_size_capture; - dev->clk_input_pin = pdata->clk_input_pin; - dev->i2s_accurate_sck = pdata->i2s_accurate_sck; - asp_chan_q = pdata->asp_chan_q; - ram_chan_q = pdata->ram_chan_q; - } - - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].asp_chan_q = asp_chan_q; - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].ram_chan_q = ram_chan_q; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].asp_chan_q = asp_chan_q; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].ram_chan_q = ram_chan_q; dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) @@ -705,10 +686,10 @@ static int davinci_i2s_probe(struct platform_device *pdev) goto err_release_clk; } - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG); - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG); /* first TX, then RX */ @@ -718,7 +699,9 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENXIO; goto err_release_clk; } - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start; + dma = &dev->dma_request[SNDRV_PCM_STREAM_PLAYBACK]; + *dma = res->start; + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = dma; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!res) { @@ -726,9 +709,11 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENXIO; goto err_release_clk; } - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; - dev->dev = &pdev->dev; + dma = &dev->dma_request[SNDRV_PCM_STREAM_CAPTURE]; + *dma = res->start; + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = dma; + dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); ret = snd_soc_register_component(&pdev->dev, &davinci_i2s_component, @@ -736,7 +721,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) if (ret != 0) goto err_release_clk; - ret = davinci_soc_platform_register(&pdev->dev); + ret = edma_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); goto err_unregister_component; -- cgit v1.2.3-59-g8ed1b From 62731d33c41d95914a0a796f319924e22e7ea411 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 Mar 2015 16:45:19 +0200 Subject: ASoC: davinci-vcif: Convert to use edma-pcm The edma-pcm can replace the old davinci-pcm as platform driver. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-vcif.c | 55 ++++++++++++++++++---------------------- 1 file changed, 24 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 5bee04279ebe..fabd05f24aeb 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -33,8 +33,9 @@ #include #include #include +#include -#include "davinci-pcm.h" +#include "edma-pcm.h" #include "davinci-i2s.h" #define MOD_REG_BIT(val, mask, set) do { \ @@ -47,7 +48,8 @@ struct davinci_vcif_dev { struct davinci_vc *davinci_vc; - struct davinci_pcm_dma_params dma_params[2]; + struct snd_dmaengine_dai_dma_data dma_data[2]; + int dma_request[2]; }; static void davinci_vcif_start(struct snd_pcm_substream *substream) @@ -93,8 +95,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, { struct davinci_vcif_dev *davinci_vcif_dev = snd_soc_dai_get_drvdata(dai); struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; - struct davinci_pcm_dma_params *dma_params = - &davinci_vcif_dev->dma_params[substream->stream]; u32 w; /* Restart the codec before setup */ @@ -113,16 +113,12 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, /* Determine xfer data type */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: - dma_params->data_type = 0; - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | DAVINCI_VC_CTRL_RD_UNSIGNED | DAVINCI_VC_CTRL_WD_BITS_8 | DAVINCI_VC_CTRL_WD_UNSIGNED, 1); break; case SNDRV_PCM_FORMAT_S8: - dma_params->data_type = 1; - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | DAVINCI_VC_CTRL_WD_BITS_8, 1); @@ -130,8 +126,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, DAVINCI_VC_CTRL_WD_UNSIGNED, 0); break; case SNDRV_PCM_FORMAT_S16_LE: - dma_params->data_type = 2; - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | DAVINCI_VC_CTRL_RD_UNSIGNED | DAVINCI_VC_CTRL_WD_BITS_8 | @@ -142,8 +136,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - dma_params->acnt = dma_params->data_type; - writel(w, davinci_vc->base + DAVINCI_VC_CTRL); return 0; @@ -172,24 +164,25 @@ static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int davinci_vcif_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai); - - snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); - return 0; -} - #define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000 static const struct snd_soc_dai_ops davinci_vcif_dai_ops = { - .startup = davinci_vcif_startup, .trigger = davinci_vcif_trigger, .hw_params = davinci_vcif_hw_params, }; +static int davinci_vcif_dai_probe(struct snd_soc_dai *dai) +{ + struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai); + + dai->playback_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dai->capture_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + + return 0; +} + static struct snd_soc_dai_driver davinci_vcif_dai = { + .probe = davinci_vcif_dai_probe, .playback = { .channels_min = 1, .channels_max = 2, @@ -225,16 +218,16 @@ static int davinci_vcif_probe(struct platform_device *pdev) /* DMA tx params */ davinci_vcif_dev->davinci_vc = davinci_vc; - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = - davinci_vc->davinci_vcif.dma_tx_channel; - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = - davinci_vc->davinci_vcif.dma_tx_addr; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = + &davinci_vc->davinci_vcif.dma_tx_channel; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = + davinci_vc->davinci_vcif.dma_tx_addr; /* DMA rx params */ - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = - davinci_vc->davinci_vcif.dma_rx_channel; - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = - davinci_vc->davinci_vcif.dma_rx_addr; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = + &davinci_vc->davinci_vcif.dma_rx_channel; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = + davinci_vc->davinci_vcif.dma_rx_addr; dev_set_drvdata(&pdev->dev, davinci_vcif_dev); @@ -245,7 +238,7 @@ static int davinci_vcif_probe(struct platform_device *pdev) return ret; } - ret = davinci_soc_platform_register(&pdev->dev); + ret = edma_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); snd_soc_unregister_component(&pdev->dev); -- cgit v1.2.3-59-g8ed1b From 9759e7ef53138c5ab46ea516ad08977eb5770393 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 Mar 2015 16:45:20 +0200 Subject: ASoC: davinci-mcasp: Deprecate the use of davinci-pcm in favor of edma-pcm The edma-pcm performs as good as the old davinci-pcm and it's use does not require the 'ping-pong' mode of davinci-pcm, which was introduced to overcome under/over flow issues when using davinci-pcm. Keep the SND_DAVINCI_SOC config option to select the SND_EDMA_SOC to avoid regression in audio support. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 15 +++---- sound/soc/davinci/davinci-mcasp.c | 87 +++++++++------------------------------ 2 files changed, 27 insertions(+), 75 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index eae4e229f341..3736d9aabc56 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -1,15 +1,16 @@ config SND_DAVINCI_SOC - tristate "SoC Audio for TI DAVINCI" + tristate depends on ARCH_DAVINCI select SND_EDMA_SOC config SND_EDMA_SOC - tristate "SoC Audio for Texas Instruments chips using eDMA (AM33XX/43XX)" + tristate "SoC Audio for Texas Instruments chips using eDMA" depends on SOC_AM33XX || SOC_AM43XX || ARCH_DAVINCI select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M here if you want audio support for TI SoC which uses eDMA. The following line of SoCs are supported by this platform driver: + - daVinci devices - AM335x - AM437x/AM438x @@ -18,7 +19,7 @@ config SND_DAVINCI_SOC_I2S config SND_DAVINCI_SOC_MCASP tristate "Multichannel Audio Serial Port (McASP) support" - depends on SND_DAVINCI_SOC || SND_OMAP_SOC || SND_EDMA_SOC + depends on SND_OMAP_SOC || SND_EDMA_SOC help Say Y or M here if you want to have support for McASP IP found in various Texas Instruments SoCs like: @@ -46,7 +47,7 @@ config SND_AM33XX_SOC_EVM config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" - depends on SND_DAVINCI_SOC && I2C + depends on SND_EDMA_SOC && I2C depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_GENERIC_EVM help @@ -74,7 +75,7 @@ endchoice config SND_DM6467_SOC_EVM tristate "SoC Audio support for DaVinci DM6467 EVM" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM && I2C + depends on SND_EDMA_SOC && MACH_DAVINCI_DM6467_EVM && I2C select SND_DAVINCI_SOC_GENERIC_EVM select SND_SOC_SPDIF @@ -83,7 +84,7 @@ config SND_DM6467_SOC_EVM config SND_DA830_SOC_EVM tristate "SoC Audio support for DA830/OMAP-L137 EVM" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM && I2C + depends on SND_EDMA_SOC && MACH_DAVINCI_DA830_EVM && I2C select SND_DAVINCI_SOC_GENERIC_EVM help @@ -92,7 +93,7 @@ config SND_DA830_SOC_EVM config SND_DA850_SOC_EVM tristate "SoC Audio support for DA850/OMAP-L138 EVM" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA850_EVM && I2C + depends on SND_EDMA_SOC && MACH_DAVINCI_DA850_EVM && I2C select SND_DAVINCI_SOC_GENERIC_EVM help Say Y if you want to add support for SoC audio on TI diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 031c1fb44ae7..0c882995a357 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include @@ -36,7 +37,6 @@ #include #include -#include "davinci-pcm.h" #include "edma-pcm.h" #include "davinci-mcasp.h" @@ -65,7 +65,6 @@ struct davinci_mcasp_context { }; struct davinci_mcasp { - struct davinci_pcm_dma_params dma_params[2]; struct snd_dmaengine_dai_dma_data dma_data[2]; void __iomem *base; u32 fifo_base; @@ -82,6 +81,7 @@ struct davinci_mcasp { u16 bclk_lrclk_ratio; int streams; u32 irq_request[2]; + int dma_request[2]; int sysclk_freq; bool bclk_master; @@ -643,7 +643,6 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, int period_words, int channels) { - struct davinci_pcm_dma_params *dma_params = &mcasp->dma_params[stream]; struct snd_dmaengine_dai_dma_data *dma_data = &mcasp->dma_data[stream]; int i; u8 tx_ser = 0; @@ -711,10 +710,8 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, * For example if three serializers are enabled the DMA * need to transfer three words per DMA request. */ - dma_params->fifo_level = active_serializers; dma_data->maxburst = active_serializers; } else { - dma_params->fifo_level = 0; dma_data->maxburst = 0; } return 0; @@ -746,7 +743,6 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, /* Configure the burst size for platform drivers */ if (numevt == 1) numevt = 0; - dma_params->fifo_level = numevt; dma_data->maxburst = numevt; return 0; @@ -872,8 +868,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); - struct davinci_pcm_dma_params *dma_params = - &mcasp->dma_params[substream->stream]; int word_length; int channels = params_channels(params); int period_size = params_period_size(params); @@ -914,31 +908,26 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: case SNDRV_PCM_FORMAT_S8: - dma_params->data_type = 1; word_length = 8; break; case SNDRV_PCM_FORMAT_U16_LE: case SNDRV_PCM_FORMAT_S16_LE: - dma_params->data_type = 2; word_length = 16; break; case SNDRV_PCM_FORMAT_U24_3LE: case SNDRV_PCM_FORMAT_S24_3LE: - dma_params->data_type = 3; word_length = 24; break; case SNDRV_PCM_FORMAT_U24_LE: case SNDRV_PCM_FORMAT_S24_LE: - dma_params->data_type = 4; word_length = 24; break; case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: - dma_params->data_type = 4; word_length = 32; break; @@ -947,11 +936,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (mcasp->version == MCASP_VERSION_2 && !dma_params->fifo_level) - dma_params->acnt = 4; - else - dma_params->acnt = dma_params->data_type; - davinci_config_channel_size(mcasp, word_length); if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) @@ -1055,17 +1039,8 @@ static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - if (mcasp->version >= MCASP_VERSION_3) { - /* Using dmaengine PCM */ - dai->playback_dma_data = - &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; - dai->capture_dma_data = - &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; - } else { - /* Using davinci-pcm */ - dai->playback_dma_data = mcasp->dma_params; - dai->capture_dma_data = mcasp->dma_params; - } + dai->playback_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dai->capture_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; return 0; } @@ -1184,28 +1159,24 @@ static const struct snd_soc_component_driver davinci_mcasp_component = { static struct davinci_mcasp_pdata dm646x_mcasp_pdata = { .tx_dma_offset = 0x400, .rx_dma_offset = 0x400, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_1, }; static struct davinci_mcasp_pdata da830_mcasp_pdata = { .tx_dma_offset = 0x2000, .rx_dma_offset = 0x2000, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_2, }; static struct davinci_mcasp_pdata am33xx_mcasp_pdata = { .tx_dma_offset = 0, .rx_dma_offset = 0, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_3, }; static struct davinci_mcasp_pdata dra7_mcasp_pdata = { .tx_dma_offset = 0x200, .rx_dma_offset = 0x284, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_4, }; @@ -1382,12 +1353,12 @@ nodata: static int davinci_mcasp_probe(struct platform_device *pdev) { - struct davinci_pcm_dma_params *dma_params; struct snd_dmaengine_dai_dma_data *dma_data; struct resource *mem, *ioarea, *res, *dat; struct davinci_mcasp_pdata *pdata; struct davinci_mcasp *mcasp; char *irq_name; + int *dma; int irq; int ret; @@ -1521,59 +1492,45 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (dat) mcasp->dat_port = true; - dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; - dma_params->asp_chan_q = pdata->asp_chan_q; - dma_params->ram_chan_q = pdata->ram_chan_q; - dma_params->sram_pool = pdata->sram_pool; - dma_params->sram_size = pdata->sram_size_playback; if (dat) - dma_params->dma_addr = dat->start; + dma_data->addr = dat->start; else - dma_params->dma_addr = mem->start + pdata->tx_dma_offset; - - /* Unconditional dmaengine stuff */ - dma_data->addr = dma_params->dma_addr; + dma_data->addr = mem->start + pdata->tx_dma_offset; + dma = &mcasp->dma_request[SNDRV_PCM_STREAM_PLAYBACK]; res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (res) - dma_params->channel = res->start; + *dma = res->start; else - dma_params->channel = pdata->tx_dma_channel; + *dma = pdata->tx_dma_channel; /* dmaengine filter data for DT and non-DT boot */ if (pdev->dev.of_node) dma_data->filter_data = "tx"; else - dma_data->filter_data = &dma_params->channel; + dma_data->filter_data = dma; /* RX is not valid in DIT mode */ if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) { - dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE]; dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; - dma_params->asp_chan_q = pdata->asp_chan_q; - dma_params->ram_chan_q = pdata->ram_chan_q; - dma_params->sram_pool = pdata->sram_pool; - dma_params->sram_size = pdata->sram_size_capture; if (dat) - dma_params->dma_addr = dat->start; + dma_data->addr = dat->start; else - dma_params->dma_addr = mem->start + pdata->rx_dma_offset; - - /* Unconditional dmaengine stuff */ - dma_data->addr = dma_params->dma_addr; + dma_data->addr = mem->start + pdata->rx_dma_offset; + dma = &mcasp->dma_request[SNDRV_PCM_STREAM_CAPTURE]; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (res) - dma_params->channel = res->start; + *dma = res->start; else - dma_params->channel = pdata->rx_dma_channel; + *dma = pdata->rx_dma_channel; /* dmaengine filter data for DT and non-DT boot */ if (pdev->dev.of_node) dma_data->filter_data = "rx"; else - dma_data->filter_data = &dma_params->channel; + dma_data->filter_data = dma; } if (mcasp->version < MCASP_VERSION_3) { @@ -1596,17 +1553,11 @@ static int davinci_mcasp_probe(struct platform_device *pdev) goto err; switch (mcasp->version) { -#if IS_BUILTIN(CONFIG_SND_DAVINCI_SOC) || \ - (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ - IS_MODULE(CONFIG_SND_DAVINCI_SOC)) - case MCASP_VERSION_1: - case MCASP_VERSION_2: - ret = davinci_soc_platform_register(&pdev->dev); - break; -#endif #if IS_BUILTIN(CONFIG_SND_EDMA_SOC) || \ (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ IS_MODULE(CONFIG_SND_EDMA_SOC)) + case MCASP_VERSION_1: + case MCASP_VERSION_2: case MCASP_VERSION_3: ret = edma_pcm_platform_register(&pdev->dev); break; -- cgit v1.2.3-59-g8ed1b From 4da4608c91308d0d15dd022074724446c15710dc Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 Mar 2015 16:45:21 +0200 Subject: ASoC: davinci: Remove unused davinci-pcm platform driver All DAI drivers has been converted to use edma-pcm instead of davinci-pcm and the driver can be removed from the tree. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/Makefile | 2 - sound/soc/davinci/davinci-pcm.c | 861 ---------------------------------------- sound/soc/davinci/davinci-pcm.h | 41 -- 3 files changed, 904 deletions(-) delete mode 100644 sound/soc/davinci/davinci-pcm.c delete mode 100644 sound/soc/davinci/davinci-pcm.h (limited to 'sound') diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index 09bf2ba92d38..f883933c1a19 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,11 +1,9 @@ # DAVINCI Platform Support -snd-soc-davinci-objs := davinci-pcm.o snd-soc-edma-objs := edma-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o snd-soc-davinci-vcif-objs:= davinci-vcif.o -obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_EDMA_SOC) += snd-soc-edma.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c deleted file mode 100644 index 7809e9d935fc..000000000000 --- a/sound/soc/davinci/davinci-pcm.c +++ /dev/null @@ -1,861 +0,0 @@ -/* - * ALSA PCM interface for the TI DAVINCI processor - * - * Author: Vladimir Barinov, - * Copyright: (C) 2007 MontaVista Software, Inc., - * added SRAM ping/pong (C) 2008 Troy Kisky - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include -#include -#include -#include -#include -#include -#include -#include - -#include -#include -#include -#include - -#include - -#include "davinci-pcm.h" - -#ifdef DEBUG -static void print_buf_info(int slot, char *name) -{ - struct edmacc_param p; - if (slot < 0) - return; - edma_read_slot(slot, &p); - printk(KERN_DEBUG "%s: 0x%x, opt=%x, src=%x, a_b_cnt=%x dst=%x\n", - name, slot, p.opt, p.src, p.a_b_cnt, p.dst); - printk(KERN_DEBUG " src_dst_bidx=%x link_bcntrld=%x src_dst_cidx=%x ccnt=%x\n", - p.src_dst_bidx, p.link_bcntrld, p.src_dst_cidx, p.ccnt); -} -#else -static void print_buf_info(int slot, char *name) -{ -} -#endif - -static struct snd_pcm_hardware pcm_hardware_playback = { - .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_BATCH), - .buffer_bytes_max = 128 * 1024, - .period_bytes_min = 32, - .period_bytes_max = 8 * 1024, - .periods_min = 16, - .periods_max = 255, - .fifo_size = 0, -}; - -static struct snd_pcm_hardware pcm_hardware_capture = { - .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_BATCH), - .buffer_bytes_max = 128 * 1024, - .period_bytes_min = 32, - .period_bytes_max = 8 * 1024, - .periods_min = 16, - .periods_max = 255, - .fifo_size = 0, -}; - -/* - * How ping/pong works.... - * - * Playback: - * ram_params - copys 2*ping_size from start of SDRAM to iram, - * links to ram_link2 - * ram_link2 - copys rest of SDRAM to iram in ping_size units, - * links to ram_link - * ram_link - copys entire SDRAM to iram in ping_size uints, - * links to self - * - * asp_params - same as asp_link[0] - * asp_link[0] - copys from lower half of iram to asp port - * links to asp_link[1], triggers iram copy event on completion - * asp_link[1] - copys from upper half of iram to asp port - * links to asp_link[0], triggers iram copy event on completion - * triggers interrupt only needed to let upper SOC levels update position - * in stream on completion - * - * When playback is started: - * ram_params started - * asp_params started - * - * Capture: - * ram_params - same as ram_link, - * links to ram_link - * ram_link - same as playback - * links to self - * - * asp_params - same as playback - * asp_link[0] - same as playback - * asp_link[1] - same as playback - * - * When capture is started: - * asp_params started - */ -struct davinci_runtime_data { - spinlock_t lock; - int period; /* current DMA period */ - int asp_channel; /* Master DMA channel */ - int asp_link[2]; /* asp parameter link channel, ping/pong */ - struct davinci_pcm_dma_params *params; /* DMA params */ - int ram_channel; - int ram_link; - int ram_link2; - struct edmacc_param asp_params; - struct edmacc_param ram_params; -}; - -static void davinci_pcm_period_elapsed(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - - prtd->period++; - if (unlikely(prtd->period >= runtime->periods)) - prtd->period = 0; -} - -static void davinci_pcm_period_reset(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - - prtd->period = 0; -} -/* - * Not used with ping/pong - */ -static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int period_size; - unsigned int dma_offset; - dma_addr_t dma_pos; - dma_addr_t src, dst; - unsigned short src_bidx, dst_bidx; - unsigned short src_cidx, dst_cidx; - unsigned int data_type; - unsigned short acnt; - unsigned int count; - unsigned int fifo_level; - - period_size = snd_pcm_lib_period_bytes(substream); - dma_offset = prtd->period * period_size; - dma_pos = runtime->dma_addr + dma_offset; - fifo_level = prtd->params->fifo_level; - - pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " - "dma_ptr = %x period_size=%x\n", prtd->asp_link[0], dma_pos, - period_size); - - data_type = prtd->params->data_type; - count = period_size / data_type; - if (fifo_level) - count /= fifo_level; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - src = dma_pos; - dst = prtd->params->dma_addr; - src_bidx = data_type; - dst_bidx = 4; - src_cidx = data_type * fifo_level; - dst_cidx = 0; - } else { - src = prtd->params->dma_addr; - dst = dma_pos; - src_bidx = 0; - dst_bidx = data_type; - src_cidx = 0; - dst_cidx = data_type * fifo_level; - } - - acnt = prtd->params->acnt; - edma_set_src(prtd->asp_link[0], src, INCR, W8BIT); - edma_set_dest(prtd->asp_link[0], dst, INCR, W8BIT); - - edma_set_src_index(prtd->asp_link[0], src_bidx, src_cidx); - edma_set_dest_index(prtd->asp_link[0], dst_bidx, dst_cidx); - - if (!fifo_level) - edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, - ASYNC); - else - edma_set_transfer_params(prtd->asp_link[0], acnt, - fifo_level, - count, fifo_level, - ABSYNC); -} - -static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) -{ - struct snd_pcm_substream *substream = data; - struct davinci_runtime_data *prtd = substream->runtime->private_data; - - print_buf_info(prtd->ram_channel, "i ram_channel"); - pr_debug("davinci_pcm: link=%d, status=0x%x\n", link, ch_status); - - if (unlikely(ch_status != EDMA_DMA_COMPLETE)) - return; - - if (snd_pcm_running(substream)) { - spin_lock(&prtd->lock); - if (prtd->ram_channel < 0) { - /* No ping/pong must fix up link dma data*/ - davinci_pcm_enqueue_dma(substream); - } - davinci_pcm_period_elapsed(substream); - spin_unlock(&prtd->lock); - snd_pcm_period_elapsed(substream); - } -} - -#ifdef CONFIG_GENERIC_ALLOCATOR -static int allocate_sram(struct snd_pcm_substream *substream, - struct gen_pool *sram_pool, unsigned size, - struct snd_pcm_hardware *ppcm) -{ - struct snd_dma_buffer *buf = &substream->dma_buffer; - struct snd_dma_buffer *iram_dma = NULL; - dma_addr_t iram_phys = 0; - void *iram_virt = NULL; - - if (buf->private_data || !size) - return 0; - - ppcm->period_bytes_max = size; - iram_virt = gen_pool_dma_alloc(sram_pool, size, &iram_phys); - if (!iram_virt) - goto exit1; - iram_dma = kzalloc(sizeof(*iram_dma), GFP_KERNEL); - if (!iram_dma) - goto exit2; - iram_dma->area = iram_virt; - iram_dma->addr = iram_phys; - memset(iram_dma->area, 0, size); - iram_dma->bytes = size; - buf->private_data = iram_dma; - return 0; -exit2: - if (iram_virt) - gen_pool_free(sram_pool, (unsigned)iram_virt, size); -exit1: - return -ENOMEM; -} - -static void davinci_free_sram(struct snd_pcm_substream *substream, - struct snd_dma_buffer *iram_dma) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct gen_pool *sram_pool = prtd->params->sram_pool; - - gen_pool_free(sram_pool, (unsigned) iram_dma->area, iram_dma->bytes); -} -#else -static int allocate_sram(struct snd_pcm_substream *substream, - struct gen_pool *sram_pool, unsigned size, - struct snd_pcm_hardware *ppcm) -{ - return 0; -} - -static void davinci_free_sram(struct snd_pcm_substream *substream, - struct snd_dma_buffer *iram_dma) -{ -} -#endif - -/* - * Only used with ping/pong. - * This is called after runtime->dma_addr, period_bytes and data_type are valid - */ -static int ping_pong_dma_setup(struct snd_pcm_substream *substream) -{ - unsigned short ram_src_cidx, ram_dst_cidx; - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd = runtime->private_data; - struct snd_dma_buffer *iram_dma = - (struct snd_dma_buffer *)substream->dma_buffer.private_data; - struct davinci_pcm_dma_params *params = prtd->params; - unsigned int data_type = params->data_type; - unsigned int acnt = params->acnt; - /* divide by 2 for ping/pong */ - unsigned int ping_size = snd_pcm_lib_period_bytes(substream) >> 1; - unsigned int fifo_level = prtd->params->fifo_level; - unsigned int count; - if ((data_type == 0) || (data_type > 4)) { - printk(KERN_ERR "%s: data_type=%i\n", __func__, data_type); - return -EINVAL; - } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - dma_addr_t asp_src_pong = iram_dma->addr + ping_size; - ram_src_cidx = ping_size; - ram_dst_cidx = -ping_size; - edma_set_src(prtd->asp_link[1], asp_src_pong, INCR, W8BIT); - - edma_set_src_index(prtd->asp_link[0], data_type, - data_type * fifo_level); - edma_set_src_index(prtd->asp_link[1], data_type, - data_type * fifo_level); - - edma_set_src(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); - } else { - dma_addr_t asp_dst_pong = iram_dma->addr + ping_size; - ram_src_cidx = -ping_size; - ram_dst_cidx = ping_size; - edma_set_dest(prtd->asp_link[1], asp_dst_pong, INCR, W8BIT); - - edma_set_dest_index(prtd->asp_link[0], data_type, - data_type * fifo_level); - edma_set_dest_index(prtd->asp_link[1], data_type, - data_type * fifo_level); - - edma_set_dest(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); - } - - if (!fifo_level) { - count = ping_size / data_type; - edma_set_transfer_params(prtd->asp_link[0], acnt, count, - 1, 0, ASYNC); - edma_set_transfer_params(prtd->asp_link[1], acnt, count, - 1, 0, ASYNC); - } else { - count = ping_size / (data_type * fifo_level); - edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, - count, fifo_level, ABSYNC); - edma_set_transfer_params(prtd->asp_link[1], acnt, fifo_level, - count, fifo_level, ABSYNC); - } - - edma_set_src_index(prtd->ram_link, ping_size, ram_src_cidx); - edma_set_dest_index(prtd->ram_link, ping_size, ram_dst_cidx); - edma_set_transfer_params(prtd->ram_link, ping_size, 2, - runtime->periods, 2, ASYNC); - - /* init master params */ - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - edma_read_slot(prtd->ram_link, &prtd->ram_params); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - struct edmacc_param p_ram; - /* Copy entire iram buffer before playback started */ - prtd->ram_params.a_b_cnt = (1 << 16) | (ping_size << 1); - /* 0 dst_bidx */ - prtd->ram_params.src_dst_bidx = (ping_size << 1); - /* 0 dst_cidx */ - prtd->ram_params.src_dst_cidx = (ping_size << 1); - prtd->ram_params.ccnt = 1; - - /* Skip 1st period */ - edma_read_slot(prtd->ram_link, &p_ram); - p_ram.src += (ping_size << 1); - p_ram.ccnt -= 1; - edma_write_slot(prtd->ram_link2, &p_ram); - /* - * When 1st started, ram -> iram dma channel will fill the - * entire iram. Then, whenever a ping/pong asp buffer finishes, - * 1/2 iram will be filled. - */ - prtd->ram_params.link_bcntrld = - EDMA_CHAN_SLOT(prtd->ram_link2) << 5; - } - return 0; -} - -/* 1 asp tx or rx channel using 2 parameter channels - * 1 ram to/from iram channel using 1 parameter channel - * - * Playback - * ram copy channel kicks off first, - * 1st ram copy of entire iram buffer completion kicks off asp channel - * asp tcc always kicks off ram copy of 1/2 iram buffer - * - * Record - * asp channel starts, tcc kicks off ram copy - */ -static int request_ping_pong(struct snd_pcm_substream *substream, - struct davinci_runtime_data *prtd, - struct snd_dma_buffer *iram_dma) -{ - dma_addr_t asp_src_ping; - dma_addr_t asp_dst_ping; - int ret; - struct davinci_pcm_dma_params *params = prtd->params; - - /* Request ram master channel */ - ret = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, - davinci_pcm_dma_irq, substream, - prtd->params->ram_chan_q); - if (ret < 0) - goto exit1; - - /* Request ram link channel */ - ret = prtd->ram_link = edma_alloc_slot( - EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit2; - - ret = prtd->asp_link[1] = edma_alloc_slot( - EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit3; - - prtd->ram_link2 = -1; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - ret = prtd->ram_link2 = edma_alloc_slot( - EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit4; - } - /* circle ping-pong buffers */ - edma_link(prtd->asp_link[0], prtd->asp_link[1]); - edma_link(prtd->asp_link[1], prtd->asp_link[0]); - /* circle ram buffers */ - edma_link(prtd->ram_link, prtd->ram_link); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - asp_src_ping = iram_dma->addr; - asp_dst_ping = params->dma_addr; /* fifo */ - } else { - asp_src_ping = params->dma_addr; /* fifo */ - asp_dst_ping = iram_dma->addr; - } - /* ping */ - edma_set_src(prtd->asp_link[0], asp_src_ping, INCR, W16BIT); - edma_set_dest(prtd->asp_link[0], asp_dst_ping, INCR, W16BIT); - edma_set_src_index(prtd->asp_link[0], 0, 0); - edma_set_dest_index(prtd->asp_link[0], 0, 0); - - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN); - prtd->asp_params.opt |= TCCHEN | - EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(prtd->asp_link[0], &prtd->asp_params); - - /* pong */ - edma_set_src(prtd->asp_link[1], asp_src_ping, INCR, W16BIT); - edma_set_dest(prtd->asp_link[1], asp_dst_ping, INCR, W16BIT); - edma_set_src_index(prtd->asp_link[1], 0, 0); - edma_set_dest_index(prtd->asp_link[1], 0, 0); - - edma_read_slot(prtd->asp_link[1], &prtd->asp_params); - prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f)); - /* interrupt after every pong completion */ - prtd->asp_params.opt |= TCINTEN | TCCHEN | - EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(prtd->asp_link[1], &prtd->asp_params); - - /* ram */ - edma_set_src(prtd->ram_link, iram_dma->addr, INCR, W32BIT); - edma_set_dest(prtd->ram_link, iram_dma->addr, INCR, W32BIT); - pr_debug("%s: audio dma channels/slots in use for ram:%u %u %u," - "for asp:%u %u %u\n", __func__, - prtd->ram_channel, prtd->ram_link, prtd->ram_link2, - prtd->asp_channel, prtd->asp_link[0], - prtd->asp_link[1]); - return 0; -exit4: - edma_free_channel(prtd->asp_link[1]); - prtd->asp_link[1] = -1; -exit3: - edma_free_channel(prtd->ram_link); - prtd->ram_link = -1; -exit2: - edma_free_channel(prtd->ram_channel); - prtd->ram_channel = -1; -exit1: - return ret; -} - -static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) -{ - struct snd_dma_buffer *iram_dma; - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct davinci_pcm_dma_params *params = prtd->params; - int ret; - - if (!params) - return -ENODEV; - - /* Request asp master DMA channel */ - ret = prtd->asp_channel = edma_alloc_channel(params->channel, - davinci_pcm_dma_irq, substream, - prtd->params->asp_chan_q); - if (ret < 0) - goto exit1; - - /* Request asp link channels */ - ret = prtd->asp_link[0] = edma_alloc_slot( - EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit2; - - iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data; - if (iram_dma) { - if (request_ping_pong(substream, prtd, iram_dma) == 0) - return 0; - printk(KERN_WARNING "%s: dma channel allocation failed," - "not using sram\n", __func__); - } - - /* Issue transfer completion IRQ when the channel completes a - * transfer, then always reload from the same slot (by a kind - * of loopback link). The completion IRQ handler will update - * the reload slot with a new buffer. - * - * REVISIT save p_ram here after setting up everything except - * the buffer and its length (ccnt) ... use it as a template - * so davinci_pcm_enqueue_dma() takes less time in IRQ. - */ - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - prtd->asp_params.opt |= TCINTEN | - EDMA_TCC(EDMA_CHAN_SLOT(prtd->asp_channel)); - prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(prtd->asp_link[0]) << 5; - edma_write_slot(prtd->asp_link[0], &prtd->asp_params); - return 0; -exit2: - edma_free_channel(prtd->asp_channel); - prtd->asp_channel = -1; -exit1: - return ret; -} - -static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - spin_lock(&prtd->lock); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - edma_start(prtd->asp_channel); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - prtd->ram_channel >= 0) { - /* copy 1st iram buffer */ - edma_start(prtd->ram_channel); - } - break; - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - edma_resume(prtd->asp_channel); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - edma_pause(prtd->asp_channel); - break; - default: - ret = -EINVAL; - break; - } - - spin_unlock(&prtd->lock); - - return ret; -} - -static int davinci_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - - davinci_pcm_period_reset(substream); - if (prtd->ram_channel >= 0) { - int ret = ping_pong_dma_setup(substream); - if (ret < 0) - return ret; - - edma_write_slot(prtd->ram_channel, &prtd->ram_params); - edma_write_slot(prtd->asp_channel, &prtd->asp_params); - - print_buf_info(prtd->ram_channel, "ram_channel"); - print_buf_info(prtd->ram_link, "ram_link"); - print_buf_info(prtd->ram_link2, "ram_link2"); - print_buf_info(prtd->asp_channel, "asp_channel"); - print_buf_info(prtd->asp_link[0], "asp_link[0]"); - print_buf_info(prtd->asp_link[1], "asp_link[1]"); - - /* - * There is a phase offset of 2 periods between the position - * used by dma setup and the position reported in the pointer - * function. - * - * The phase offset, when not using ping-pong buffers, is due to - * the two consecutive calls to davinci_pcm_enqueue_dma() below. - * - * Whereas here, with ping-pong buffers, the phase is due to - * there being an entire buffer transfer complete before the - * first dma completion event triggers davinci_pcm_dma_irq(). - */ - davinci_pcm_period_elapsed(substream); - davinci_pcm_period_elapsed(substream); - - return 0; - } - davinci_pcm_enqueue_dma(substream); - davinci_pcm_period_elapsed(substream); - - /* Copy self-linked parameter RAM entry into master channel */ - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - edma_write_slot(prtd->asp_channel, &prtd->asp_params); - davinci_pcm_enqueue_dma(substream); - davinci_pcm_period_elapsed(substream); - - return 0; -} - -static snd_pcm_uframes_t -davinci_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd = runtime->private_data; - unsigned int offset; - int asp_count; - unsigned int period_size = snd_pcm_lib_period_bytes(substream); - - /* - * There is a phase offset of 2 periods between the position used by dma - * setup and the position reported in the pointer function. Either +2 in - * the dma setup or -2 here in the pointer function (with wrapping, - * both) accounts for this offset -- choose the latter since it makes - * the first-time setup clearer. - */ - spin_lock(&prtd->lock); - asp_count = prtd->period - 2; - spin_unlock(&prtd->lock); - - if (asp_count < 0) - asp_count += runtime->periods; - asp_count *= period_size; - - offset = bytes_to_frames(runtime, asp_count); - if (offset >= runtime->buffer_size) - offset = 0; - - return offset; -} - -static int davinci_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd; - struct snd_pcm_hardware *ppcm; - int ret = 0; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *pa; - struct davinci_pcm_dma_params *params; - - pa = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - if (!pa) - return -ENODEV; - params = &pa[substream->stream]; - - ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - &pcm_hardware_playback : &pcm_hardware_capture; - allocate_sram(substream, params->sram_pool, params->sram_size, ppcm); - snd_soc_set_runtime_hwparams(substream, ppcm); - /* ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - return ret; - - prtd = kzalloc(sizeof(struct davinci_runtime_data), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - spin_lock_init(&prtd->lock); - prtd->params = params; - prtd->asp_channel = -1; - prtd->asp_link[0] = prtd->asp_link[1] = -1; - prtd->ram_channel = -1; - prtd->ram_link = -1; - prtd->ram_link2 = -1; - - runtime->private_data = prtd; - - ret = davinci_pcm_dma_request(substream); - if (ret) { - printk(KERN_ERR "davinci_pcm: Failed to get dma channels\n"); - kfree(prtd); - } - - return ret; -} - -static int davinci_pcm_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd = runtime->private_data; - - if (prtd->ram_channel >= 0) - edma_stop(prtd->ram_channel); - if (prtd->asp_channel >= 0) - edma_stop(prtd->asp_channel); - if (prtd->asp_link[0] >= 0) - edma_unlink(prtd->asp_link[0]); - if (prtd->asp_link[1] >= 0) - edma_unlink(prtd->asp_link[1]); - if (prtd->ram_link >= 0) - edma_unlink(prtd->ram_link); - - if (prtd->asp_link[0] >= 0) - edma_free_slot(prtd->asp_link[0]); - if (prtd->asp_link[1] >= 0) - edma_free_slot(prtd->asp_link[1]); - if (prtd->asp_channel >= 0) - edma_free_channel(prtd->asp_channel); - if (prtd->ram_link >= 0) - edma_free_slot(prtd->ram_link); - if (prtd->ram_link2 >= 0) - edma_free_slot(prtd->ram_link2); - if (prtd->ram_channel >= 0) - edma_free_channel(prtd->ram_channel); - - kfree(prtd); - - return 0; -} - -static int davinci_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - return snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(hw_params)); -} - -static int davinci_pcm_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - -static int davinci_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops davinci_pcm_ops = { - .open = davinci_pcm_open, - .close = davinci_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = davinci_pcm_hw_params, - .hw_free = davinci_pcm_hw_free, - .prepare = davinci_pcm_prepare, - .trigger = davinci_pcm_trigger, - .pointer = davinci_pcm_pointer, - .mmap = davinci_pcm_mmap, -}; - -static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream, - size_t size) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - - pr_debug("davinci_pcm: preallocate_dma_buffer: area=%p, addr=%p, " - "size=%d\n", (void *) buf->area, (void *) buf->addr, size); - - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - return 0; -} - -static void davinci_pcm_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - struct snd_dma_buffer *iram_dma; - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - iram_dma = buf->private_data; - if (iram_dma) { - davinci_free_sram(substream, iram_dma); - kfree(iram_dma); - } - } -} - -static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = davinci_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK, - pcm_hardware_playback.buffer_bytes_max); - if (ret) - return ret; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = davinci_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE, - pcm_hardware_capture.buffer_bytes_max); - if (ret) - return ret; - } - - return 0; -} - -static struct snd_soc_platform_driver davinci_soc_platform = { - .ops = &davinci_pcm_ops, - .pcm_new = davinci_pcm_new, - .pcm_free = davinci_pcm_free, -}; - -int davinci_soc_platform_register(struct device *dev) -{ - return devm_snd_soc_register_platform(dev, &davinci_soc_platform); -} -EXPORT_SYMBOL_GPL(davinci_soc_platform_register); - -MODULE_AUTHOR("Vladimir Barinov"); -MODULE_DESCRIPTION("TI DAVINCI PCM DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h deleted file mode 100644 index 0fe2346a9aa2..000000000000 --- a/sound/soc/davinci/davinci-pcm.h +++ /dev/null @@ -1,41 +0,0 @@ -/* - * ALSA PCM interface for the TI DAVINCI processor - * - * Author: Vladimir Barinov, - * Copyright: (C) 2007 MontaVista Software, Inc., - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _DAVINCI_PCM_H -#define _DAVINCI_PCM_H - -#include -#include -#include - -struct davinci_pcm_dma_params { - int channel; /* sync dma channel ID */ - unsigned short acnt; - dma_addr_t dma_addr; /* device physical address for DMA */ - unsigned sram_size; - struct gen_pool *sram_pool; /* SRAM gen_pool for ping pong */ - enum dma_event_q asp_chan_q; /* event queue number for ASP channel */ - enum dma_event_q ram_chan_q; /* event queue number for RAM channel */ - unsigned char data_type; /* xfer data type */ - unsigned char convert_mono_stereo; - unsigned int fifo_level; -}; - -#if IS_ENABLED(CONFIG_SND_DAVINCI_SOC) -int davinci_soc_platform_register(struct device *dev); -#else -static inline int davinci_soc_platform_register(struct device *dev) -{ - return 0; -} -#endif /* CONFIG_SND_DAVINCI_SOC */ - -#endif -- cgit v1.2.3-59-g8ed1b From 6742e15cf92a8dc3065843a627952ed518e08267 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Tue, 3 Mar 2015 13:28:53 +0200 Subject: ASoC: omap-pcm: Allow only formats with 1, 2, and 4 byte physical size sDMA support only transfer elements with 1, 2, and 4 byte physical size. Initialize the pcm driver accordingly. Signed-off-by: Jyri Sarha Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index f4b05bc23e4b..e49ee2383a88 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -39,7 +39,7 @@ #define pcm_omap1510() 0 #endif -static const struct snd_pcm_hardware omap_pcm_hardware = { +static struct snd_pcm_hardware omap_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | @@ -53,6 +53,24 @@ static const struct snd_pcm_hardware omap_pcm_hardware = { .buffer_bytes_max = 128 * 1024, }; +/* sDMA supports only 1, 2, and 4 byte transfer elements. */ +static void omap_pcm_limit_supported_formats(void) +{ + int i; + + for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { + switch (snd_pcm_format_physical_width(i)) { + case 8: + case 16: + case 32: + omap_pcm_hardware.formats |= (1LL << i); + break; + default: + break; + } + } +} + /* this may get called several times by oss emulation */ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -235,6 +253,7 @@ static struct snd_soc_platform_driver omap_soc_platform = { int omap_pcm_platform_register(struct device *dev) { + omap_pcm_limit_supported_formats(); return devm_snd_soc_register_platform(dev, &omap_soc_platform); } EXPORT_SYMBOL_GPL(omap_pcm_platform_register); -- cgit v1.2.3-59-g8ed1b From 2bf9eba14340a53776a742f2c8a0bfbd9c86d259 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 3 Mar 2015 18:31:29 +0800 Subject: ASoC: rt5670: Fix the speaker mono output issue We need to set left/right control for the speaker amp to get stereo output on speaker. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0632b7458a53..592f961b5de5 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2700,6 +2700,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, regmap_write(rt5670->regmap, RT5670_RESET, 0); + regmap_read(rt5670->regmap, RT5670_VENDOR_ID, &val); + if (val >= 4) + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0980); + else + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0d00); + ret = regmap_register_patch(rt5670->regmap, init_list, ARRAY_SIZE(init_list)); if (ret != 0) -- cgit v1.2.3-59-g8ed1b From bbed297d373471c8e4c3183bf67472a768576664 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Sun, 22 Feb 2015 16:43:21 +0000 Subject: ASoC: wm8804: Split out bus drivers Simplify dependencies by using new style split out bus interfaces. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 18 +++++- sound/soc/codecs/Makefile | 4 ++ sound/soc/codecs/wm8804-i2c.c | 64 +++++++++++++++++++ sound/soc/codecs/wm8804-spi.c | 56 +++++++++++++++++ sound/soc/codecs/wm8804.c | 139 +++++------------------------------------- sound/soc/codecs/wm8804.h | 7 +++ 6 files changed, 162 insertions(+), 126 deletions(-) create mode 100644 sound/soc/codecs/wm8804-i2c.c create mode 100644 sound/soc/codecs/wm8804-spi.c (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 064e6c18e109..1d17988df796 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -141,7 +141,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8770 if SPI_MASTER select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8782 - select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8804_I2C if I2C + select SND_SOC_WM8804_SPI if SPI_MASTER select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C select SND_SOC_WM8904 if I2C @@ -744,8 +745,19 @@ config SND_SOC_WM8782 tristate config SND_SOC_WM8804 - tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver" - depends on SND_SOC_I2C_AND_SPI + tristate + +config SND_SOC_WM8804_I2C + tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver I2C" + depends on I2C + select SND_SOC_WM8804 + select REGMAP_I2C + +config SND_SOC_WM8804_SPI + tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver SPI" + depends on SPI_MASTER + select SND_SOC_WM8804 + select REGMAP_SPI config SND_SOC_WM8900 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 69b8666d187a..7acb6c174cb4 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -145,6 +145,8 @@ snd-soc-wm8770-objs := wm8770.o snd-soc-wm8776-objs := wm8776.o snd-soc-wm8782-objs := wm8782.o snd-soc-wm8804-objs := wm8804.o +snd-soc-wm8804-i2c-objs := wm8804-i2c.o +snd-soc-wm8804-spi-objs := wm8804-spi.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8904-objs := wm8904.o @@ -323,6 +325,8 @@ obj-$(CONFIG_SND_SOC_WM8770) += snd-soc-wm8770.o obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o obj-$(CONFIG_SND_SOC_WM8782) += snd-soc-wm8782.o obj-$(CONFIG_SND_SOC_WM8804) += snd-soc-wm8804.o +obj-$(CONFIG_SND_SOC_WM8804_I2C) += snd-soc-wm8804-i2c.o +obj-$(CONFIG_SND_SOC_WM8804_SPI) += snd-soc-wm8804-spi.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8904) += snd-soc-wm8904.o diff --git a/sound/soc/codecs/wm8804-i2c.c b/sound/soc/codecs/wm8804-i2c.c new file mode 100644 index 000000000000..5bd4af2b4059 --- /dev/null +++ b/sound/soc/codecs/wm8804-i2c.c @@ -0,0 +1,64 @@ +/* + * wm8804-i2c.c -- WM8804 S/PDIF transceiver driver - I2C + * + * Copyright 2015 Cirrus Logic Inc + * + * Author: Charles Keepax + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include + +#include "wm8804.h" + +static int wm8804_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_i2c(i2c, &wm8804_regmap_config); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return wm8804_probe(&i2c->dev, regmap); +} + +static int wm8804_i2c_remove(struct i2c_client *i2c) +{ + wm8804_remove(&i2c->dev); + return 0; +} + +static const struct i2c_device_id wm8804_i2c_id[] = { + { "wm8804", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8804_i2c_id); + +static const struct of_device_id wm8804_of_match[] = { + { .compatible = "wlf,wm8804", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8804_of_match); + +static struct i2c_driver wm8804_i2c_driver = { + .driver = { + .name = "wm8804", + .owner = THIS_MODULE, + .of_match_table = wm8804_of_match, + }, + .probe = wm8804_i2c_probe, + .remove = wm8804_i2c_remove, + .id_table = wm8804_i2c_id +}; + +module_i2c_driver(wm8804_i2c_driver); + +MODULE_DESCRIPTION("ASoC WM8804 driver - I2C"); +MODULE_AUTHOR("Charles Keepax "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8804-spi.c b/sound/soc/codecs/wm8804-spi.c new file mode 100644 index 000000000000..287e11e90794 --- /dev/null +++ b/sound/soc/codecs/wm8804-spi.c @@ -0,0 +1,56 @@ +/* + * wm8804-spi.c -- WM8804 S/PDIF transceiver driver - SPI + * + * Copyright 2015 Cirrus Logic Inc + * + * Author: Charles Keepax + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include + +#include "wm8804.h" + +static int wm8804_spi_probe(struct spi_device *spi) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_spi(spi, &wm8804_regmap_config); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return wm8804_probe(&spi->dev, regmap); +} + +static int wm8804_spi_remove(struct spi_device *spi) +{ + wm8804_remove(&spi->dev); + return 0; +} + +static const struct of_device_id wm8804_of_match[] = { + { .compatible = "wlf,wm8804", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8804_of_match); + +static struct spi_driver wm8804_spi_driver = { + .driver = { + .name = "wm8804", + .owner = THIS_MODULE, + .of_match_table = wm8804_of_match, + }, + .probe = wm8804_spi_probe, + .remove = wm8804_spi_remove +}; + +module_spi_driver(wm8804_spi_driver); + +MODULE_DESCRIPTION("ASoC WM8804 driver - SPI"); +MODULE_AUTHOR("Charles Keepax "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index b2b0e68f707e..b5a04fc5060f 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -15,10 +15,7 @@ #include #include #include -#include #include -#include -#include #include #include #include @@ -518,7 +515,7 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm8804_remove(struct snd_soc_codec *codec) +static int wm8804_codec_remove(struct snd_soc_codec *codec) { struct wm8804_priv *wm8804; int i; @@ -531,7 +528,7 @@ static int wm8804_remove(struct snd_soc_codec *codec) return 0; } -static int wm8804_probe(struct snd_soc_codec *codec) +static int wm8804_codec_probe(struct snd_soc_codec *codec) { struct wm8804_priv *wm8804; int i, id1, id2, ret; @@ -649,8 +646,8 @@ static struct snd_soc_dai_driver wm8804_dai = { }; static const struct snd_soc_codec_driver soc_codec_dev_wm8804 = { - .probe = wm8804_probe, - .remove = wm8804_remove, + .probe = wm8804_codec_probe, + .remove = wm8804_codec_remove, .set_bias_level = wm8804_set_bias_level, .idle_bias_off = true, @@ -658,13 +655,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .num_controls = ARRAY_SIZE(wm8804_snd_controls), }; -static const struct of_device_id wm8804_of_match[] = { - { .compatible = "wlf,wm8804", }, - { } -}; -MODULE_DEVICE_TABLE(of, wm8804_of_match); - -static const struct regmap_config wm8804_regmap_config = { +const struct regmap_config wm8804_regmap_config = { .reg_bits = 8, .val_bits = 8, @@ -675,128 +666,30 @@ static const struct regmap_config wm8804_regmap_config = { .reg_defaults = wm8804_reg_defaults, .num_reg_defaults = ARRAY_SIZE(wm8804_reg_defaults), }; +EXPORT_SYMBOL_GPL(wm8804_regmap_config); -#if defined(CONFIG_SPI_MASTER) -static int wm8804_spi_probe(struct spi_device *spi) +int wm8804_probe(struct device *dev, struct regmap *regmap) { struct wm8804_priv *wm8804; - int ret; - wm8804 = devm_kzalloc(&spi->dev, sizeof *wm8804, GFP_KERNEL); + wm8804 = devm_kzalloc(dev, sizeof(*wm8804), GFP_KERNEL); if (!wm8804) return -ENOMEM; - wm8804->regmap = devm_regmap_init_spi(spi, &wm8804_regmap_config); - if (IS_ERR(wm8804->regmap)) { - ret = PTR_ERR(wm8804->regmap); - return ret; - } - - spi_set_drvdata(spi, wm8804); + dev_set_drvdata(dev, wm8804); - ret = snd_soc_register_codec(&spi->dev, - &soc_codec_dev_wm8804, &wm8804_dai, 1); - - return ret; -} - -static int wm8804_spi_remove(struct spi_device *spi) -{ - snd_soc_unregister_codec(&spi->dev); - return 0; -} - -static struct spi_driver wm8804_spi_driver = { - .driver = { - .name = "wm8804", - .owner = THIS_MODULE, - .of_match_table = wm8804_of_match, - }, - .probe = wm8804_spi_probe, - .remove = wm8804_spi_remove -}; -#endif - -#if IS_ENABLED(CONFIG_I2C) -static int wm8804_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) -{ - struct wm8804_priv *wm8804; - int ret; - - wm8804 = devm_kzalloc(&i2c->dev, sizeof *wm8804, GFP_KERNEL); - if (!wm8804) - return -ENOMEM; + wm8804->regmap = regmap; - wm8804->regmap = devm_regmap_init_i2c(i2c, &wm8804_regmap_config); - if (IS_ERR(wm8804->regmap)) { - ret = PTR_ERR(wm8804->regmap); - return ret; - } - - i2c_set_clientdata(i2c, wm8804); - - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_wm8804, &wm8804_dai, 1); - return ret; -} - -static int wm8804_i2c_remove(struct i2c_client *i2c) -{ - snd_soc_unregister_codec(&i2c->dev); - return 0; -} - -static const struct i2c_device_id wm8804_i2c_id[] = { - { "wm8804", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, wm8804_i2c_id); - -static struct i2c_driver wm8804_i2c_driver = { - .driver = { - .name = "wm8804", - .owner = THIS_MODULE, - .of_match_table = wm8804_of_match, - }, - .probe = wm8804_i2c_probe, - .remove = wm8804_i2c_remove, - .id_table = wm8804_i2c_id -}; -#endif - -static int __init wm8804_modinit(void) -{ - int ret = 0; - -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&wm8804_i2c_driver); - if (ret) { - printk(KERN_ERR "Failed to register wm8804 I2C driver: %d\n", - ret); - } -#endif -#if defined(CONFIG_SPI_MASTER) - ret = spi_register_driver(&wm8804_spi_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register wm8804 SPI driver: %d\n", - ret); - } -#endif - return ret; + return snd_soc_register_codec(dev, &soc_codec_dev_wm8804, + &wm8804_dai, 1); } -module_init(wm8804_modinit); +EXPORT_SYMBOL_GPL(wm8804_probe); -static void __exit wm8804_exit(void) +void wm8804_remove(struct device *dev) { -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&wm8804_i2c_driver); -#endif -#if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&wm8804_spi_driver); -#endif + snd_soc_unregister_codec(dev); } -module_exit(wm8804_exit); +EXPORT_SYMBOL_GPL(wm8804_remove); MODULE_DESCRIPTION("ASoC WM8804 driver"); MODULE_AUTHOR("Dimitris Papastamos "); diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index e72d4f4ba6b1..a39a2563dc67 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -13,6 +13,8 @@ #ifndef _WM8804_H #define _WM8804_H +#include + /* * Register values. */ @@ -62,4 +64,9 @@ #define WM8804_MCLKDIV_256FS 0 #define WM8804_MCLKDIV_128FS 1 +extern const struct regmap_config wm8804_regmap_config; + +int wm8804_probe(struct device *dev, struct regmap *regmap); +void wm8804_remove(struct device *dev); + #endif /* _WM8804_H */ -- cgit v1.2.3-59-g8ed1b From 6f2c9348095ae1a489abafe2ab3db7deca406e49 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Sun, 22 Feb 2015 16:43:22 +0000 Subject: ASoC: wm8804: Merge CODEC probe and bus probe All of the things in the CODEC probe, such as getting the regulators and verifying the chip ID, are better done in bus probe. It is better to fail during bus probe if this is the wrong chip and all resource allocation should be done in the bus probe anyway. This patch merges the CODEC probe into bus probe. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804.c | 180 +++++++++++++++++++++------------------------- 1 file changed, 82 insertions(+), 98 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index b5a04fc5060f..1bd4ace29594 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -182,9 +182,9 @@ static bool wm8804_volatile(struct device *dev, unsigned int reg) } } -static int wm8804_reset(struct snd_soc_codec *codec) +static int wm8804_reset(struct wm8804_priv *wm8804) { - return snd_soc_write(codec, WM8804_RST_DEVID1, 0x0); + return regmap_write(wm8804->regmap, WM8804_RST_DEVID1, 0x0); } static int wm8804_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) @@ -515,100 +515,6 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm8804_codec_remove(struct snd_soc_codec *codec) -{ - struct wm8804_priv *wm8804; - int i; - - wm8804 = snd_soc_codec_get_drvdata(codec); - - for (i = 0; i < ARRAY_SIZE(wm8804->supplies); ++i) - regulator_unregister_notifier(wm8804->supplies[i].consumer, - &wm8804->disable_nb[i]); - return 0; -} - -static int wm8804_codec_probe(struct snd_soc_codec *codec) -{ - struct wm8804_priv *wm8804; - int i, id1, id2, ret; - - wm8804 = snd_soc_codec_get_drvdata(codec); - - for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) - wm8804->supplies[i].supply = wm8804_supply_names[i]; - - ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8804->supplies), - wm8804->supplies); - if (ret) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - return ret; - } - - wm8804->disable_nb[0].notifier_call = wm8804_regulator_event_0; - wm8804->disable_nb[1].notifier_call = wm8804_regulator_event_1; - - /* This should really be moved into the regulator core */ - for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) { - ret = regulator_register_notifier(wm8804->supplies[i].consumer, - &wm8804->disable_nb[i]); - if (ret != 0) { - dev_err(codec->dev, - "Failed to register regulator notifier: %d\n", - ret); - } - } - - ret = regulator_bulk_enable(ARRAY_SIZE(wm8804->supplies), - wm8804->supplies); - if (ret) { - dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); - return ret; - } - - id1 = snd_soc_read(codec, WM8804_RST_DEVID1); - if (id1 < 0) { - dev_err(codec->dev, "Failed to read device ID: %d\n", id1); - ret = id1; - goto err_reg_enable; - } - - id2 = snd_soc_read(codec, WM8804_DEVID2); - if (id2 < 0) { - dev_err(codec->dev, "Failed to read device ID: %d\n", id2); - ret = id2; - goto err_reg_enable; - } - - id2 = (id2 << 8) | id1; - - if (id2 != 0x8805) { - dev_err(codec->dev, "Invalid device ID: %#x\n", id2); - ret = -EINVAL; - goto err_reg_enable; - } - - ret = snd_soc_read(codec, WM8804_DEVREV); - if (ret < 0) { - dev_err(codec->dev, "Failed to read device revision: %d\n", - ret); - goto err_reg_enable; - } - dev_info(codec->dev, "revision %c\n", ret + 'A'); - - ret = wm8804_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset: %d\n", ret); - goto err_reg_enable; - } - - return 0; - -err_reg_enable: - regulator_bulk_disable(ARRAY_SIZE(wm8804->supplies), wm8804->supplies); - return ret; -} - static const struct snd_soc_dai_ops wm8804_dai_ops = { .hw_params = wm8804_hw_params, .set_fmt = wm8804_set_fmt, @@ -646,8 +552,6 @@ static struct snd_soc_dai_driver wm8804_dai = { }; static const struct snd_soc_codec_driver soc_codec_dev_wm8804 = { - .probe = wm8804_codec_probe, - .remove = wm8804_codec_remove, .set_bias_level = wm8804_set_bias_level, .idle_bias_off = true, @@ -671,6 +575,8 @@ EXPORT_SYMBOL_GPL(wm8804_regmap_config); int wm8804_probe(struct device *dev, struct regmap *regmap) { struct wm8804_priv *wm8804; + unsigned int id1, id2; + int i, ret; wm8804 = devm_kzalloc(dev, sizeof(*wm8804), GFP_KERNEL); if (!wm8804) @@ -680,13 +586,91 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) wm8804->regmap = regmap; + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) + wm8804->supplies[i].supply = wm8804_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(wm8804->supplies), + wm8804->supplies); + if (ret) { + dev_err(dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + wm8804->disable_nb[0].notifier_call = wm8804_regulator_event_0; + wm8804->disable_nb[1].notifier_call = wm8804_regulator_event_1; + + /* This should really be moved into the regulator core */ + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) { + ret = regulator_register_notifier(wm8804->supplies[i].consumer, + &wm8804->disable_nb[i]); + if (ret != 0) { + dev_err(dev, + "Failed to register regulator notifier: %d\n", + ret); + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8804->supplies), + wm8804->supplies); + if (ret) { + dev_err(dev, "Failed to enable supplies: %d\n", ret); + goto err_reg_enable; + } + + ret = regmap_read(regmap, WM8804_RST_DEVID1, &id1); + if (ret < 0) { + dev_err(dev, "Failed to read device ID: %d\n", ret); + goto err_reg_enable; + } + + ret = regmap_read(regmap, WM8804_DEVID2, &id2); + if (ret < 0) { + dev_err(dev, "Failed to read device ID: %d\n", ret); + goto err_reg_enable; + } + + id2 = (id2 << 8) | id1; + + if (id2 != 0x8805) { + dev_err(dev, "Invalid device ID: %#x\n", id2); + ret = -EINVAL; + goto err_reg_enable; + } + + ret = regmap_read(regmap, WM8804_DEVREV, &id1); + if (ret < 0) { + dev_err(dev, "Failed to read device revision: %d\n", + ret); + goto err_reg_enable; + } + dev_info(dev, "revision %c\n", id1 + 'A'); + + ret = wm8804_reset(wm8804); + if (ret < 0) { + dev_err(dev, "Failed to issue reset: %d\n", ret); + goto err_reg_enable; + } + return snd_soc_register_codec(dev, &soc_codec_dev_wm8804, &wm8804_dai, 1); + +err_reg_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8804->supplies), wm8804->supplies); + return ret; } EXPORT_SYMBOL_GPL(wm8804_probe); void wm8804_remove(struct device *dev) { + struct wm8804_priv *wm8804; + int i; + + wm8804 = dev_get_drvdata(dev); + + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); ++i) + regulator_unregister_notifier(wm8804->supplies[i].consumer, + &wm8804->disable_nb[i]); + snd_soc_unregister_codec(dev); } EXPORT_SYMBOL_GPL(wm8804_remove); -- cgit v1.2.3-59-g8ed1b