October 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 0/9] ASoC: OMAP machine driver cleanups
by Peter Ujfalusi 10 Oct '11

10 Oct '11

Hello, Some cleanups for the OMAP ASoC machine drivers: - do not call soc_dapm_sync from dai_link->init callback since it is useless - Convert drivers to use table based DAPM initialization - Remove dapm_enable_pin calls from dai_link->init callback since pins are enabled by default Changelog since v1: - Do not touch ams-delta just remove the dapm_sync call. - Do not convert soc_dapm_sync calls to dapm_new_widgets call Regards, Peter --- Peter Ujfalusi (9): ASoC: OMAP machines: Remove soc_dapm_sync() call from init ASoC: am3517evm: Let core to deal with the DAPM widgets ASoC: n810: Let the core to register DAPM widgets/routes and controls ASoC: osk5912: Let core to deal with the DAPM widgets ASoC: sdp3430: Let core to deal with the DAPM widgets ASoC: zoom2: Let core to deal with the DAPM widgets ASoC: zoom2: No need to call dapm_pin_enable at init time ASoC: sdp4430: Let core to deal with the DAPM widgets ASoC: sdp4430: No need to call dapm_pin_enable at init time sound/soc/omap/am3517evm.c | 28 +++++----------------------- sound/soc/omap/ams-delta.c | 1 - sound/soc/omap/n810.c | 23 +++++++---------------- sound/soc/omap/omap3pandora.c | 8 ++------ sound/soc/omap/osk5912.c | 27 +++++---------------------- sound/soc/omap/rx51.c | 2 -- sound/soc/omap/sdp3430.c | 18 +++++------------- sound/soc/omap/sdp4430.c | 27 +++++---------------------- sound/soc/omap/zoom2.c | 26 ++++++-------------------- 9 files changed, 35 insertions(+), 125 deletions(-) -- 1.7.7

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[alsa-devel] ASoC updates for 3.2
by Mark Brown 10 Oct '11

10 Oct '11

The following changes since commit 734787550a60b768b675c26f93d134f6dc370bb5: ASoC: wm8995: Slightly optimize configure_clock (2011-10-04 13:09:55 +0100) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 Axel Lin (24): ASoC: wm8994: Fix setting rate_reg for wm8994-aif2 ASoC: Avoid writing to WM8971_RESET in wm8971_resume ASoC: wm8750: Use snd_soc_update_bits for read-modify-write ASoC: wm8988: Use snd_soc_update_bits for read-modify-write ASoC: wm8711: Use snd_soc_update_bits for read-modify-write ASoC: Remove unneeded hw_write initialisation in ak4671 ASoC: Remove unneeded hw_write initialisation in wm8523 ASoC: Remove unused function declaration in imx-ssi.h ASoC: imx: Remove unused variable 'dai' ASoC: Remove unused variable 'wm9090' in wm9090_probe ASoC: wm8990: Convert to snd_soc_cache_sync for sync reg_cache with the hardware ASoC: wm8990: Remove incorrect comments ASoC: Return early with -EINVAL if invalid dai format is detected ASoC: wm8510: Convert to snd_soc_cache_sync ASoC: wm8711: Convert to snd_soc_cache_sync ASoC: wm8731: Convert to snd_soc_cache_sync ASoC: wm8750: Convert to snd_soc_cache_sync ASoC: wm8776: Convert to snd_soc_cache_sync ASoC: wm8940: Convert to snd_soc_cache_sync ASoC: wm8960: Convert to snd_soc_cache_sync ASoC: wm8971: Convert to snd_soc_cache_sync ASoC: wm8974: Convert to snd_soc_cache_sync ASoC: wm8988: Convert to snd_soc_cache_sync ASoC: ak4535: convert to soc-cache Mark Brown (32): Merge branch 'for-3.1' into for-3.2 ASoC: Remove direct register cache accesses from WM8962 driver regulator: Fix return code from regulator_disable_deferred() ASoC: Factor write of widget power out into a separate function ASoC: Move bias level decision into main dapm_power_widgets() ASoC: Ensure all DAPM widgets have a power check callback ASoC: Factor out widget power check operation ASoC: Mark headphone, mic, speaker and line widgets as always connected ASoC: Do DAPM power checks only for widgets changed since last run ASoC: Reduce the number of neigbours we mark dirty when updating power ASoC: Add verbose debugging showing why widgets get marked dirty ASoC: Only run power_check() on a widget once per run ASoC: Don't mark the outputs of supplies as dirty on state changes ASoC: Stop checking for supplied widgets after we find the first ASoC: Use dapm_mark_dirty() for new DAPM widgets for consistency ASoC: Remove references to linux(a)wolfsonmicro.com ASoC: Ensure DAPM widgets are set up before we sync jacks Merge branch 'for-3.1' into for-3.2 regulator: Add WM1811 support ASoC: Suppress early calls to snd_soc_dapm_sync() ASoC: Remove redundant snd_soc_dapm_sync() calls from machine drivers ASoC: Ensure all DAPM widgets are instantiated with the card ASoC: Assign power_check when we allocate DAPM widgets ASoC: Cache connected input and output recursions ASoC: Squash error codes from regmap down to -1 on read ASoC: Fix DAPM sync for TLV320AIC3x custom DAPM widget ASoC: Convert Simtec machines to table based DAPM init ASoC: Convert H1940 to table based init ASoC: Convert RX1950 to table based init ASoC: Convert SmartQ to table based init ASoC: Convert SMDK WM8580 to table based DAPM init ASoC: Convert Jive to table based init Michael Opdenacker (1): ASoC: fix checkpatch.pl error in omap-mcbsp Peter Ujfalusi (15): ASoC: core: Introduce SOC_DOUBLE_VALUE macro ASoC: core: Introduce SOC_DOUBLE_R_VALUE macro ASoC: twl6040: Simplify custom put_volsw callback ASoC: Consolidate use of controls with custom get/put function ASoC: twl6040: Simplify custom get_volsw callback ASoC: twl6040: Prepare for core put_volsw/volsw_2r merger ASoC: core: Change SOC_SINGLE/DOUBLE_VALUE representation ASoC: core: Combine snd_soc_info_volsw/info_volsw_2r functions ASoC: core: Combine snd_soc_get_volsw/get_volsw_2r functions ASoC: core: Combine snd_soc_put_volsw/put_volsw_2r functions ASoC: twl6040: Simply call snd_soc_put_volsw form the custom code ASoC: core: Remove snd_soc_put_volsw_2r definition ASoC: twl6040: Warn user in twl6040_put_volsw for error case ASoC: Replace remaining use of *_volsw_2r with *_volsw ASoC: fix codec breakage caused by the volsw/volsw_2r merger Wolfram Sang (1): ASoC: tlv320aic32x4 fix initialization of micpga routing drivers/regulator/core.c | 9 +- drivers/regulator/wm8994-regulator.c | 13 + include/sound/soc-dapm.h | 8 + include/sound/soc.h | 85 ++++--- sound/soc/atmel/playpaq_wm8510.c | 1 - sound/soc/atmel/sam9g20_wm8731.c | 2 - sound/soc/atmel/snd-soc-afeb9260.c | 2 - sound/soc/codecs/ak4535.c | 100 ++------ sound/soc/codecs/ak4671.c | 2 - sound/soc/codecs/cs4270.c | 8 +- sound/soc/codecs/cs42l51.c | 9 +- sound/soc/codecs/lm4857.c | 2 +- sound/soc/codecs/tlv320aic23.c | 14 +- sound/soc/codecs/tlv320aic32x4.c | 4 +- sound/soc/codecs/tlv320aic3x.c | 4 + sound/soc/codecs/twl4030.c | 48 +--- sound/soc/codecs/twl6040.c | 76 ++---- sound/soc/codecs/wm8350.c | 43 ++-- sound/soc/codecs/wm8510.c | 13 +- sound/soc/codecs/wm8523.c | 1 - sound/soc/codecs/wm8580.c | 38 +-- sound/soc/codecs/wm8711.c | 19 +- sound/soc/codecs/wm8731.c | 15 +- sound/soc/codecs/wm8750.c | 42 +--- sound/soc/codecs/wm8776.c | 16 +- sound/soc/codecs/wm8940.c | 46 ++-- sound/soc/codecs/wm8960.c | 15 +- sound/soc/codecs/wm8962.c | 21 +- sound/soc/codecs/wm8971.c | 15 +- sound/soc/codecs/wm8974.c | 15 +- sound/soc/codecs/wm8988.c | 18 +-- sound/soc/codecs/wm8990.c | 36 ++-- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm9090.c | 1 - sound/soc/codecs/wm_hubs.c | 20 +- sound/soc/davinci/davinci-evm.c | 2 - sound/soc/imx/imx-pcm-fiq.c | 1 - sound/soc/imx/imx-ssi.c | 1 - sound/soc/imx/imx-ssi.h | 6 - sound/soc/kirkwood/kirkwood-t5325.c | 2 - sound/soc/mid-x86/mfld_machine.c | 2 - sound/soc/omap/omap-mcbsp.c | 3 +- sound/soc/pxa/corgi.c | 1 - sound/soc/pxa/e740_wm9705.c | 2 - sound/soc/pxa/e750_wm9705.c | 2 - sound/soc/pxa/e800_wm9712.c | 1 - sound/soc/pxa/magician.c | 1 - sound/soc/pxa/mioa701_wm9713.c | 1 - sound/soc/pxa/palm27x.c | 4 - sound/soc/pxa/saarb.c | 4 - sound/soc/pxa/tavorevb3.c | 4 - sound/soc/pxa/tosa.c | 1 - sound/soc/pxa/z2.c | 4 - sound/soc/pxa/zylonite.c | 1 - sound/soc/samsung/goni_wm8994.c | 2 - sound/soc/samsung/h1940_uda1380.c | 19 +- sound/soc/samsung/jive_wm8750.c | 17 +- sound/soc/samsung/neo1973_wm8753.c | 4 - sound/soc/samsung/rx1950_uda1380.c | 33 +-- sound/soc/samsung/s3c24xx_simtec_hermes.c | 11 +- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 11 +- sound/soc/samsung/smartq_wm8987.c | 25 +-- sound/soc/samsung/smdk_wm8580.c | 45 +--- sound/soc/samsung/smdk_wm8994.c | 2 - sound/soc/sh/sh7760-ac97.c | 7 - sound/soc/soc-core.c | 166 +++--------- sound/soc/soc-dapm.c | 346 +++++++++++++++++------- sound/soc/soc-io.c | 2 +- sound/soc/soc-jack.c | 2 + sound/soc/tegra/tegra_wm8903.c | 2 - sound/soc/tegra/trimslice.c | 2 - 72 files changed, 601 insertions(+), 903 deletions(-)

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[alsa-devel] [PATCH] ASoC: Instantiate card widgets immediately
by Mark Brown 10 Oct '11

10 Oct '11

This ensures they are available prior to the card late_probe(). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8dc9aba..d6af52b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1430,6 +1430,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); + snd_soc_dapm_new_widgets(&card->dapm); + for (i = 0; i < card->num_links; i++) { dai_link = &card->dai_link[i]; -- 1.7.6.3

1 0

[alsa-devel] amixer cget :Assertion `ctl && info && (info->id.name[0] || info->id.numid)' failed. Aborted
by Raymond Yau 10 Oct '11

10 Oct '11

There is assertion when specify iface=PCM or iface=PCM,name='' when using amixer -c0 cget according to the synthax [[iface=<iface>,][name='name',][index=<index>,][device=<device>,][subdevice=<subdevice>]]|[numid=<numid>] amixer -c0 cget iface=PCM amixer: control.c:258: snd_ctl_elem_info: Assertion `ctl && info && (info->id.name[0] || info->id.numid)' failed. Aborted amixer -c0 cget iface=PCM,name='' amixer: control.c:258: snd_ctl_elem_info: Assertion `ctl && info && (info->id.name[0] || info->id.numid)' failed. Aborted amixer -c0 cget Specify a full control identifier: [[iface=<iface>,][name='name',][index=<index>,][device=<device>,][subdevice=<subdevice>]]|[numid=<numid>] amixer -c0 cget iface=PCM,name='ILD' amixer: Cannot find the given element from control hw:0 amixer -c0 cget iface=PCM,name='ILD',device=2 numid=45,iface=PCM,name='ILD',device=2 ; type=INTEGER,access=rw------,values=2,min=0,max=2147483647,step=0 : values=0,0

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[alsa-devel] [PATCH] ALSA: remove the dead code from snd_pcm_open_file()
by Feng Tang 10 Oct '11

10 Oct '11

The rpcm_file parameter is never used in current ALSA code, so remove it to make it cleaner. Signed-off-by: Feng Tang <feng.tang(a)intel.com> --- sound/core/pcm_native.c | 12 +++--------- 1 files changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 1c6be91..2d3af5d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2058,16 +2058,12 @@ EXPORT_SYMBOL(snd_pcm_open_substream); static int snd_pcm_open_file(struct file *file, struct snd_pcm *pcm, - int stream, - struct snd_pcm_file **rpcm_file) + int stream) { struct snd_pcm_file *pcm_file; struct snd_pcm_substream *substream; int err; - if (rpcm_file) - *rpcm_file = NULL; - err = snd_pcm_open_substream(pcm, stream, file, &substream); if (err < 0) return err; @@ -2083,8 +2079,7 @@ static int snd_pcm_open_file(struct file *file, substream->pcm_release = pcm_release_private; } file->private_data = pcm_file; - if (rpcm_file) - *rpcm_file = pcm_file; + return 0; } @@ -2113,7 +2108,6 @@ static int snd_pcm_capture_open(struct inode *inode, struct file *file) static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream) { int err; - struct snd_pcm_file *pcm_file; wait_queue_t wait; if (pcm == NULL) { @@ -2131,7 +2125,7 @@ static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream) add_wait_queue(&pcm->open_wait, &wait); mutex_lock(&pcm->open_mutex); while (1) { - err = snd_pcm_open_file(file, pcm, stream, &pcm_file); + err = snd_pcm_open_file(file, pcm, stream); if (err >= 0) break; if (err == -EAGAIN) { -- 1.7.1

2 1

[alsa-devel] [PATCH 1/5] ASoC: Convert H1940 to table based init
by Mark Brown 10 Oct '11

10 Oct '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/h1940_uda1380.c | 17 +++++------------ 1 files changed, 5 insertions(+), 12 deletions(-) diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index cdaed90..03cfa5f 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -183,18 +183,6 @@ static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; int err; - /* Add h1940 specific widgets */ - err = snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets, - ARRAY_SIZE(uda1380_dapm_widgets)); - if (err) - return err; - - /* Set up h1940 specific audio path audio_mapnects */ - err = snd_soc_dapm_add_routes(dapm, audio_map, - ARRAY_SIZE(audio_map)); - if (err) - return err; - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Speaker"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); @@ -229,6 +217,11 @@ static struct snd_soc_card h1940_asoc = { .name = "h1940", .dai_link = h1940_uda1380_dai, .num_links = ARRAY_SIZE(h1940_uda1380_dai), + + .dapm_widgets = uda1380_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static int __init h1940_init(void) -- 1.7.6.3

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[alsa-devel] [PATCH v3 00/10] Convert wm8xxx codec drivers to use snd_soc_write/snd_soc_cache_sync
by Axel Lin 10 Oct '11

10 Oct '11

This serial converts the wm8xxx codec drivers to use snd_soc_cache_sync. I don't have these hardware, but I assume Mark has all these hardware. These patches are UNTESTED. I'd appreciate if someone can test it. (Current code is broken anyway, thus we need convert to use new API. ) v3: use snd_soc_cache_sync to replace customized xxxx_sync_cache() calls. Drop wm8990 patch in v2, which is already merged. I also change the subject line back to be the same as v1. ( we are using snd_soc_cache_sync now, not snd_soc_write ) v2: For 7, 9 CODECs that are write only, convert to using cache_bypass + snd_soc_write v2 also includes wm8990 that I had sent separately. Axel Lin (10): ASoC: wm8510: Convert to snd_soc_cache_sync ASoC: wm8711: Convert to snd_soc_cache_sync ASoC: wm8731: Convert to snd_soc_cache_sync ASoC: wm8750: Convert to snd_soc_cache_sync ASoC: wm8776: Convert to snd_soc_cache_sync ASoC: wm8940: Convert to snd_soc_cache_sync ASoC: wm8960: Convert to snd_soc_cache_sync ASoC: wm8971: Convert to snd_soc_cache_sync ASoC: wm8974: Convert to snd_soc_cache_sync ASoC: wm8988: Convert to snd_soc_cache_sync sound/soc/codecs/wm8510.c | 13 +---------- sound/soc/codecs/wm8711.c | 15 ++----------- sound/soc/codecs/wm8731.c | 15 +------------ sound/soc/codecs/wm8750.c | 16 +------------- sound/soc/codecs/wm8776.c | 16 +------------- sound/soc/codecs/wm8940.c | 46 +++++++++++++++++++++----------------------- sound/soc/codecs/wm8960.c | 14 +++--------- sound/soc/codecs/wm8971.c | 15 ++----------- sound/soc/codecs/wm8974.c | 13 +---------- sound/soc/codecs/wm8988.c | 17 +-------------- 10 files changed, 44 insertions(+), 136 deletions(-) -- 1.7.4.1

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[alsa-devel] [RFC][PATCH] ASoC: ak4535: convert to soc-cache
by Axel Lin 10 Oct '11

10 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- I don't have this hardware, this patch is UNTESTED!! ( But current code is broken anyway ) I'd appreciate if someone can test it. Axel sound/soc/codecs/ak4535.c | 100 ++++++++------------------------------------- 1 files changed, 18 insertions(+), 82 deletions(-) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index e1a214e..68df32d 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -34,7 +34,6 @@ struct ak4535_priv { unsigned int sysclk; enum snd_soc_control_type control_type; - void *control_data; }; /* @@ -47,63 +46,6 @@ static const u16 ak4535_reg[AK4535_CACHEREGNUM] = { 0x0000, 0x0000, 0x0057, 0x0000, }; -/* - * read ak4535 register cache - */ -static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg >= AK4535_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write ak4535 register cache - */ -static inline void ak4535_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= AK4535_CACHEREGNUM) - return; - cache[reg] = value; -} - -/* - * write to the AK4535 register space - */ -static int ak4535_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D8 AK4535 register offset - * D7...D0 register data - */ - data[0] = reg & 0xff; - data[1] = value & 0xff; - - ak4535_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -static int ak4535_sync(struct snd_soc_codec *codec) -{ - u16 *cache = codec->reg_cache; - int i, r = 0; - - for (i = 0; i < AK4535_CACHEREGNUM; i++) - r |= ak4535_write(codec, i, cache[i]); - - return r; -}; - static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"}; static const char *ak4535_mono_out[] = {"(L + R)/2", "Hi-Z"}; static const char *ak4535_hp_out[] = {"Stereo", "Mono"}; @@ -304,7 +246,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); - u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5); + u8 mode2 = snd_soc_read(codec, AK4535_MODE2) & ~(0x3 << 5); int rate = params_rate(params), fs = 256; if (rate) @@ -323,7 +265,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, } /* set rate */ - ak4535_write(codec, AK4535_MODE2, mode2); + snd_soc_write(codec, AK4535_MODE2, mode2); return 0; } @@ -348,44 +290,37 @@ static int ak4535_set_dai_fmt(struct snd_soc_dai *codec_dai, /* use 32 fs for BCLK to save power */ mode1 |= 0x4; - ak4535_write(codec, AK4535_MODE1, mode1); + snd_soc_write(codec, AK4535_MODE1, mode1); return 0; } static int ak4535_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC); + u16 mute_reg = snd_soc_read(codec, AK4535_DAC); if (!mute) - ak4535_write(codec, AK4535_DAC, mute_reg & ~0x20); + snd_soc_write(codec, AK4535_DAC, mute_reg & ~0x20); else - ak4535_write(codec, AK4535_DAC, mute_reg | 0x20); + snd_soc_write(codec, AK4535_DAC, mute_reg | 0x20); return 0; } static int ak4535_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 i, mute_reg; - switch (level) { case SND_SOC_BIAS_ON: - mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC); - ak4535_write(codec, AK4535_DAC, mute_reg & ~0x20); + snd_soc_update_bits(codec, AK4535_DAC, 0x20, 0); break; case SND_SOC_BIAS_PREPARE: - mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC); - ak4535_write(codec, AK4535_DAC, mute_reg | 0x20); + snd_soc_update_bits(codec, AK4535_DAC, 0x20, 0x20); break; case SND_SOC_BIAS_STANDBY: - i = ak4535_read_reg_cache(codec, AK4535_PM1); - ak4535_write(codec, AK4535_PM1, i | 0x80); - i = ak4535_read_reg_cache(codec, AK4535_PM2); - ak4535_write(codec, AK4535_PM2, i & (~0x80)); + snd_soc_update_bits(codec, AK4535_PM1, 0x80, 0x80); + snd_soc_update_bits(codec, AK4535_PM2, 0x80, 0); break; case SND_SOC_BIAS_OFF: - i = ak4535_read_reg_cache(codec, AK4535_PM1); - ak4535_write(codec, AK4535_PM1, i & (~0x80)); + snd_soc_update_bits(codec, AK4535_PM1, 0x80, 0); break; } codec->dapm.bias_level = level; @@ -428,7 +363,7 @@ static int ak4535_suspend(struct snd_soc_codec *codec, pm_message_t state) static int ak4535_resume(struct snd_soc_codec *codec) { - ak4535_sync(codec); + snd_soc_cache_sync(codec); ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } @@ -436,11 +371,15 @@ static int ak4535_resume(struct snd_soc_codec *codec) static int ak4535_probe(struct snd_soc_codec *codec) { struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); + int ret; printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION); - codec->control_data = ak4535->control_data; - + ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4535->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } /* power on device */ ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -461,8 +400,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4535 = { .remove = ak4535_remove, .suspend = ak4535_suspend, .resume = ak4535_resume, - .read = ak4535_read_reg_cache, - .write = ak4535_write, .set_bias_level = ak4535_set_bias_level, .reg_cache_size = ARRAY_SIZE(ak4535_reg), .reg_word_size = sizeof(u8), @@ -485,7 +422,6 @@ static __devinit int ak4535_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, ak4535); - ak4535->control_data = i2c; ak4535->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, -- 1.7.4.1

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[alsa-devel] [PATCH 0/2] omap: sound: style and warning fixes
by Michael Opdenacker 10 Oct '11

10 Oct '11

These patches apply to Tony Lindgren's linux-omap tree (they can also be applied to Linus' tree) They fix a warning at compile time and a coding style error.

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[alsa-devel] [PATCH] ASoC: Fix DAPM sync for TLV320AIC3x custom DAPM widget
by Mark Brown 10 Oct '11

10 Oct '11

We really should be doing this in the core, not in a driver... Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 3 +++ sound/soc/codecs/tlv320aic3x.c | 4 ++++ sound/soc/soc-dapm.c | 3 ++- 3 files changed, 9 insertions(+), 1 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e2853da..5b2202f 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -381,6 +381,9 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, const char *pin); +/* Mostly internal - should not normally be used */ +void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason); + /* dapm widget types */ enum snd_soc_dapm_type { snd_soc_dapm_input = 0, /* input pin */ diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index d877b39..be55b7f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -197,6 +197,10 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, else /* old connection must be powered down */ path->connect = invert ? 1 : 0; + + dapm_mark_dirty(path->source, "tlv320aic3x source"); + dapm_mark_dirty(path->sink, "tlv320aic3x sink"); + break; } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e49c56d..b9836ab 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -124,7 +124,7 @@ static bool dapm_dirty_widget(struct snd_soc_dapm_widget *w) return !list_empty(&w->dirty); } -static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) +void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) { if (!dapm_dirty_widget(w)) { dev_vdbg(w->dapm->dev, "Marking %s dirty due to %s\n", @@ -132,6 +132,7 @@ static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) list_add_tail(&w->dirty, &w->dapm->card->dapm_dirty); } } +EXPORT_SYMBOL_GPL(dapm_mark_dirty); /* create a new dapm widget */ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( -- 1.7.6.3

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