March 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/1] ASoC: Samsung: Fix build on multiplatform
by Sachin Kamat 27 Mar '14

27 Mar '14

PCM and S/PDIF drivers referenced mach headers for a trivial data structure. This caused build errors on multiplatform builds as machine headers are not accessible from driver files. Move the data structure definition to the driver header and remove the dependency. While at it rename the structure to avoid multiple definition errors as the same structure is also used by the platform code. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/samsung/ac97.c | 6 +++--- sound/soc/samsung/dma.h | 6 +++++- sound/soc/samsung/i2s.c | 6 +++--- sound/soc/samsung/pcm.c | 5 ++--- sound/soc/samsung/s3c2412-i2s.c | 4 ++-- sound/soc/samsung/s3c24xx-i2s.c | 4 ++-- sound/soc/samsung/spdif.c | 3 +-- 7 files changed, 18 insertions(+), 16 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 4a88e36c82ec..76b072bd4ba2 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -39,15 +39,15 @@ struct s3c_ac97_info { }; static struct s3c_ac97_info s3c_ac97; -static struct s3c2410_dma_client s3c_dma_client_out = { +static struct s3c_dma_client s3c_dma_client_out = { .name = "AC97 PCMOut" }; -static struct s3c2410_dma_client s3c_dma_client_in = { +static struct s3c_dma_client s3c_dma_client_in = { .name = "AC97 PCMIn" }; -static struct s3c2410_dma_client s3c_dma_client_micin = { +static struct s3c_dma_client s3c_dma_client_micin = { .name = "AC97 MicIn" }; diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 225e5378014e..ad7c0f04f00d 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -14,8 +14,12 @@ #include <sound/dmaengine_pcm.h> +struct s3c_dma_client { + char *name; +}; + struct s3c_dma_params { - struct s3c2410_dma_client *client; /* stream identifier */ + struct s3c_dma_client *client; /* stream identifier */ int channel; /* Channel ID */ dma_addr_t dma_addr; int dma_size; /* Size of the DMA transfer */ diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 0a9b44c940ce..048ead967199 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1211,10 +1211,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->dma_playback.dma_addr = regs_base + I2STXD; pri_dai->dma_capture.dma_addr = regs_base + I2SRXD; pri_dai->dma_playback.client = - (struct s3c2410_dma_client *)&pri_dai->dma_playback; + (struct s3c_dma_client *)&pri_dai->dma_playback; pri_dai->dma_playback.ch_name = "tx"; pri_dai->dma_capture.client = - (struct s3c2410_dma_client *)&pri_dai->dma_capture; + (struct s3c_dma_client *)&pri_dai->dma_capture; pri_dai->dma_capture.ch_name = "rx"; pri_dai->dma_playback.dma_size = 4; pri_dai->dma_capture.dma_size = 4; @@ -1233,7 +1233,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) } sec_dai->dma_playback.dma_addr = regs_base + I2STXDS; sec_dai->dma_playback.client = - (struct s3c2410_dma_client *)&sec_dai->dma_playback; + (struct s3c_dma_client *)&sec_dai->dma_playback; sec_dai->dma_playback.ch_name = "tx-sec"; if (!np) { diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 6a5e4bf6ac96..ab54e297957c 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -20,7 +20,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "pcm.h" @@ -132,11 +131,11 @@ struct s3c_pcm_info { struct s3c_dma_params *dma_capture; }; -static struct s3c2410_dma_client s3c_pcm_dma_client_out = { +static struct s3c_dma_client s3c_pcm_dma_client_out = { .name = "PCM Stereo out" }; -static struct s3c2410_dma_client s3c_pcm_dma_client_in = { +static struct s3c_dma_client s3c_pcm_dma_client_in = { .name = "PCM Stereo in" }; diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index d0794458963a..e9bb5d7a71ee 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -33,11 +33,11 @@ #include "regs-i2s-v2.h" #include "s3c2412-i2s.h" -static struct s3c2410_dma_client s3c2412_dma_client_out = { +static struct s3c_dma_client s3c2412_dma_client_out = { .name = "I2S PCM Stereo out" }; -static struct s3c2410_dma_client s3c2412_dma_client_in = { +static struct s3c_dma_client s3c2412_dma_client_in = { .name = "I2S PCM Stereo in" }; diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index f31e916dd8c4..d7b8457b5650 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -31,11 +31,11 @@ #include "dma.h" #include "s3c24xx-i2s.h" -static struct s3c2410_dma_client s3c24xx_dma_client_out = { +static struct s3c_dma_client s3c24xx_dma_client_out = { .name = "I2S PCM Stereo out" }; -static struct s3c2410_dma_client s3c24xx_dma_client_in = { +static struct s3c_dma_client s3c24xx_dma_client_in = { .name = "I2S PCM Stereo in" }; diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 28487dcc4538..cfe63b7bcc9f 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -18,7 +18,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "spdif.h" @@ -94,7 +93,7 @@ struct samsung_spdif_info { struct s3c_dma_params *dma_playback; }; -static struct s3c2410_dma_client spdif_dma_client_out = { +static struct s3c_dma_client spdif_dma_client_out = { .name = "S/PDIF Stereo out", }; -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: rt5640: Move cache sync() to resume()
by Oder Chiou 27 Mar '14

27 Mar '14

The patch fixes the defect in case of resume which doesn't sync the cache. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 0061ae6..0883c8a 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1890,11 +1890,9 @@ static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, static int rt5640_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); switch (level) { case SND_SOC_BIAS_STANDBY: if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { - regcache_cache_only(rt5640->regmap, false); snd_soc_update_bits(codec, RT5640_PWR_ANLG1, RT5640_PWR_VREF1 | RT5640_PWR_MB | RT5640_PWR_BG | RT5640_PWR_VREF2, @@ -1904,7 +1902,6 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, RT5640_PWR_ANLG1, RT5640_PWR_FV1 | RT5640_PWR_FV2, RT5640_PWR_FV1 | RT5640_PWR_FV2); - regcache_sync(rt5640->regmap); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); snd_soc_update_bits(codec, RT5640_MICBIAS, @@ -1979,6 +1976,9 @@ static int rt5640_resume(struct snd_soc_codec *codec) msleep(400); } + regcache_cache_only(rt5640->regmap, false); + regcache_sync(rt5640->regmap); + return 0; } #else -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH v2] ASoC: davinci-mcasp: set up channel status bits for S/PDIF mode
by Daniel Mack 27 Mar '14

27 Mar '14

In DIT (S/PDIF) mode, program the transmitted user bits to reflect the configured sample rate, along with some other details. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- Thanks to Peter for the review and bearing with my confusion :) sound/soc/davinci/davinci-mcasp.c | 49 +++++++++++++++++++++++++++++++++++++-- 1 file changed, 47 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 375e197..1fbded0 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -27,6 +27,7 @@ #include <linux/of_platform.h> #include <linux/of_device.h> +#include <sound/asoundef.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -576,8 +577,12 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) } /* S/PDIF */ -static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) +static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, + unsigned int rate) { + u32 val = 0; + u8 *bytes = (u8*) &val; + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15)); @@ -599,6 +604,46 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) /* Enable the DIT */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); + /* Set S/PDIF channel status bits */ + bytes[0] = IEC958_AES0_CON_NOT_COPYRIGHT; + bytes[1] = IEC958_AES1_CON_PCM_CODER; + + switch (rate) { + case 22050: + bytes[3] |= IEC958_AES3_CON_FS_22050; + break; + case 24000: + bytes[3] |= IEC958_AES3_CON_FS_24000; + break; + case 32000: + bytes[3] |= IEC958_AES3_CON_FS_32000; + break; + case 44100: + bytes[3] |= IEC958_AES3_CON_FS_44100; + break; + case 48000: + bytes[3] |= IEC958_AES3_CON_FS_48000; + break; + case 88200: + bytes[3] |= IEC958_AES3_CON_FS_88200; + break; + case 96000: + bytes[3] |= IEC958_AES3_CON_FS_96000; + break; + case 176400: + bytes[3] |= IEC958_AES3_CON_FS_176400; + break; + case 192000: + bytes[3] |= IEC958_AES3_CON_FS_192000; + break; + default: + printk(KERN_WARNING "unsupported sampling rate: %d\n", rate); + return -EINVAL; + } + + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG, val); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG, val); + return 0; } @@ -634,7 +679,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return ret; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) - ret = mcasp_dit_hw_param(mcasp); + ret = mcasp_dit_hw_param(mcasp, params_rate(params)); else ret = mcasp_i2s_hw_param(mcasp, substream->stream); -- 1.8.5.3

2 2

[alsa-devel] [PATCH] ASoC: davinci-mcasp: set up user bits for S/PDIF mode
by Daniel Mack 27 Mar '14

27 Mar '14

In DIT (S/PDIF) mode, program the transmitted user bits to reflect the configured sample rate, along with some other details. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/davinci/davinci-mcasp.c | 55 +++++++++++++++++++++++++++++++++++++-- 1 file changed, 53 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 712a7cd..ec0463a 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -27,6 +27,7 @@ #include <linux/of_platform.h> #include <linux/of_device.h> +#include <sound/asoundef.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -566,8 +567,11 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) } /* S/PDIF */ -static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) +static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, + unsigned int rate) { + u32 val = 0; + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15)); @@ -589,6 +593,53 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) /* Enable the DIT */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); + /* Set S/PDIF channel status bits */ + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG + 0, + IEC958_AES0_CON_NOT_COPYRIGHT); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG + 0, + IEC958_AES0_CON_NOT_COPYRIGHT); + + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG + 1, + IEC958_AES1_CON_PCM_CODER); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG + 1, + IEC958_AES1_CON_PCM_CODER); + + switch (rate) { + case 22050: + val |= IEC958_AES3_CON_FS_22050; + break; + case 24000: + val |= IEC958_AES3_CON_FS_24000; + break; + case 32000: + val |= IEC958_AES3_CON_FS_32000; + break; + case 44100: + val |= IEC958_AES3_CON_FS_44100; + break; + case 48000: + val |= IEC958_AES3_CON_FS_48000; + break; + case 88200: + val |= IEC958_AES3_CON_FS_88200; + break; + case 96000: + val |= IEC958_AES3_CON_FS_96000; + break; + case 176400: + val |= IEC958_AES3_CON_FS_176400; + break; + case 192000: + val |= IEC958_AES3_CON_FS_192000; + break; + default: + printk(KERN_WARNING "unsupported sampling rate: %d\n", rate); + return -EINVAL; + } + + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG + 3, val); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG + 3, val); + return 0; } @@ -621,7 +672,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, fifo_level = mcasp->rxnumevt * active_serializers; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) - ret = mcasp_dit_hw_param(mcasp); + ret = mcasp_dit_hw_param(mcasp, params_rate(params)); else ret = mcasp_i2s_hw_param(mcasp, substream->stream); -- 1.8.5.3

3 5

[alsa-devel] Lenovo G580 mic noise when recording
by Nevyan Neykov 27 Mar '14

27 Mar '14

Hello, my lenovo g580(2189, 20150) laptop is using intel hd audio conexant driver( Intel Corporation 7 Series/C210 ) and experiencing lots of noise when trying to record from either external or internal microphone. If I lower mic volume / boost the voice get barely recorded. I am using this line in my alsa conf: options snd-hda-intel model=auto position_fix=2 also tried with model=asus etc..., when using position_fix=1 recording receives lots of cracking noise. I must mention that both mics are working very well under win7/8. Regards.

1 0

[alsa-devel] [PATCH RFC v3 0/2] Move dai-link level properties away from dai subnodes
by Jyri Sarha 27 Mar '14

27 Mar '14

This patch is implemented on top of late patches from Jean-Francois Moine [1]. These patches implement the main part of the simple-card changes discussed in alsa-devel mailing list [2]. Since RFC v2 version of the patches, address remaining comments from Jean-Francois Moine [4]: - ASoC: simple-card: Move dai-link level properties away from dai subnodes - Only allow legacy parsing if there is no dai-link node. - Fix double unref in of_card parsing loop. Since RFC version of the patches: - ASoC: core: Update snd_soc_of_parse_daifmt() interface - Make minimal changes to simple-card for it to work with the updated snd_soc_of_parse_daifmt(). (Kuninori Morimoto's comment [3]). - ASoC: simple-card: Move dai-link level properties away from dai subnodes - Fix typo in DT binding document (Jean-Francois Moine's comment [4]). - Remove #define DEBUG - Remove accidental line wrap at simple-audio-card,name property descripition in the DT binding document [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074592.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074388.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074699.html [4] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074697.html Best regards, Jyri Jyri Sarha (2): ASoC: core: Update snd_soc_of_parse_daifmt() interface ASoC: simple-card: Move dai-link level properties away from dai subnodes .../devicetree/bindings/sound/simple-card.txt | 88 ++++---- include/sound/soc.h | 4 +- sound/soc/generic/simple-card.c | 238 ++++++++++++-------- sound/soc/soc-core.c | 8 +- 4 files changed, 200 insertions(+), 138 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: jack: export gpio detect
by cfreeman＠nvidia.com 26 Mar '14

26 Mar '14

From: Christopher Freeman <cfreeman(a)nvidia.com> Export the gpio detect function so machine drivers may call it. Interrupts for the jack may be disabled during sleep, so this allows a machine driver to have the jack status updated during resume. Signed-off-by: Christopher Freeman <cfreeman(a)nvidia.com> --- include/sound/soc.h | 1 + sound/soc/soc-jack.c | 3 ++- 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a6a059c..36383cb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -442,6 +442,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); +void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio); #endif /* codec register bit access */ diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0bb5ccc..fd97cc5 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -231,7 +231,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_notifier_unregister); #ifdef CONFIG_GPIOLIB /* gpio detect */ -static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) +void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) { struct snd_soc_jack *jack = gpio->jack; int enable; @@ -251,6 +251,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) snd_soc_jack_report(jack, report, gpio->report); } +EXPORT_SYMBOL_GPL(snd_soc_jack_gpio_detect); /* irq handler for gpio pin */ static irqreturn_t gpio_handler(int irq, void *data) -- 1.8.3.2 ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

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Re: [alsa-devel] linux-next: Tree for Mar 26 (sound/soc/codecs/cs42l52.c)
by Randy Dunlap 26 Mar '14

26 Mar '14

On 03/25/2014 11:35 PM, Stephen Rothwell wrote: > Hi all, > > This tree still fails (more than usual) the powerpc allyesconfig build. > > Changes since 20140325: > on x86_64: when CONFIG_INPUT=m and SNC_SOC_CS42L42=y: sound/built-in.o: In function `cs42l52_beep_set': cs42l52.c:(.text+0xd4b01): undefined reference to `input_event' sound/built-in.o: In function `cs42l52_probe': cs42l52.c:(.text+0xd4beb): undefined reference to `devm_input_allocate_device' cs42l52.c:(.text+0xd4d0c): undefined reference to `input_register_device' sound/built-in.o: In function `wm8962_beep_set': wm8962.c:(.text+0x116240): undefined reference to `input_event' sound/built-in.o: In function `wm8962_probe': wm8962.c:(.text+0x116cae): undefined reference to `devm_input_allocate_device' wm8962.c:(.text+0x116e0e): undefined reference to `input_register_device' Full randconfig file is attached. -- ~Randy

1 1

[alsa-devel] Fwd: Re: no reset_resume for driver snd-usb-audio for logitech headset H600
by baumber＠hotmail.com 26 Mar '14

26 Mar '14

FWD: Any news to the snd-usb-audio issue? Thanks, Bernhard Old Message: ------------- I have tested the patch with kernel 3.12.0 with suspend and hibernate, and the problem is gone. No messages "no reset_resume for driver snd-usb-audio?" or other negative effects/messages. After resume, the usb_audio_device (Headset) is the master channel in KDE Mixer, as it should be. => So the problem is solved for me. Can you please add the patch to the latest kernels and LTS-kernels or whitelist the workaround for my device (ID 046d:0a29 Logitech, Inc. H600 [Wireless Headset]) Thanks, Bernhard On 2013-10-20 23:35, Takashi Iwai wrote: > At Sun, 20 Oct 2013 23:18:13 +0200, > <baumber(a)hotmail.com> wrote: >> >> On 2013-10-09 11:16 (CEST,UTC+2), Takashi Iwai wrote: >>>>>> diff --git a/sound/usb/card.c b/sound/usb/card.c >>>>>> index 9d9de8d..c591a83 100644 >>>>>> --- a/sound/usb/card.c >>>>>> +++ b/sound/usb/card.c >>>>>> @@ -747,6 +747,7 @@ static struct usb_driver usb_audio_driver = { >>>>>> .disconnect = usb_audio_disconnect, >>>>>> .suspend = usb_audio_suspend, >>>>>> .resume = usb_audio_resume, >>>>>> + .reset_resume = usb_audio_resume, >>>>>> .id_table = usb_audio_ids, >>>>>> .supports_autosuspend = 1, >>>>>> }; >> >> Shall I add the line in b/sound/usb/card.c and try it? Which kernel version shall I use? > > It doesn't matter which kernel version is. > The error message should disappear, but I don't know whether any other > positive effect you get. As Clemens pointed, there can be some > negative effects by this, so it cannot be taken as is. But at least, > it'd be good to know the result... > > > thanks, > > Takashi

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[alsa-devel] [PATCH 1/3] ASoC: ak5386: add regulator consumer support
by Daniel Mack 26 Mar '14

26 Mar '14

The chip has two power supplies, VA and VDD. Register and enable them both. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/ak5386.c | 35 ++++++++++++++++++++++++++++++++--- 1 file changed, 32 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index 72e953b..99fb5db 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -14,12 +14,18 @@ #include <linux/of.h> #include <linux/of_gpio.h> #include <linux/of_device.h> +#include <linux/regulator/consumer.h> #include <sound/soc.h> #include <sound/pcm.h> #include <sound/initval.h> +static const char *supply_names[] = { + "va", "vd" +}; + struct ak5386_priv { int reset_gpio; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; static const struct snd_soc_dapm_widget ak5386_dapm_widgets[] = { @@ -122,6 +128,7 @@ static int ak5386_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct ak5386_priv *priv; + int ret, i; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) @@ -130,6 +137,19 @@ static int ak5386_probe(struct platform_device *pdev) priv->reset_gpio = -EINVAL; dev_set_drvdata(dev, priv); + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + priv->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret < 0) + return ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret < 0) + return ret; + if (of_match_device(of_match_ptr(ak5386_dt_ids), dev)) priv->reset_gpio = of_get_named_gpio(dev->of_node, "reset-gpio", 0); @@ -140,13 +160,22 @@ static int ak5386_probe(struct platform_device *pdev) "AK5386 Reset")) priv->reset_gpio = -EINVAL; - return snd_soc_register_codec(dev, &soc_codec_ak5386, - &ak5386_dai, 1); + ret = snd_soc_register_codec(dev, &soc_codec_ak5386, + &ak5386_dai, 1); + if (ret < 0) + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), + priv->supplies); + return ret; } static int ak5386_remove(struct platform_device *pdev) { - snd_soc_unregister_codec(&pdev->dev); + struct device *dev = &pdev->dev; + struct ak5386_priv *priv = dev_get_drvdata(dev); + + snd_soc_unregister_codec(dev); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; } -- 1.8.5.3

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