November 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASoC: samsung: Provide helper for DMA init
by Mark Brown 09 Dec '13

09 Dec '13

From: Mark Brown <broonie(a)linaro.org> In preparation for using the dmaengine helpers in ASoC rather than the dmaengine wrappers for the Samsung API wrap the configuration of dma_data. The dmaengine code expects different data to that used by the legacy API. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/ac97.c | 51 +++++++++++++++--------------------------------- sound/soc/samsung/dma.c | 8 ++++++++ sound/soc/samsung/dma.h | 3 +++ sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/pcm.c | 18 +++++++++-------- 5 files changed, 38 insertions(+), 44 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 350ba23..4a88e36 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -221,24 +221,6 @@ static struct snd_ac97_bus_ops s3c_ac97_ops = { .reset = s3c_ac97_cold_reset, }; -static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct s3c_dma_params *dma_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c_ac97_pcm_out; - else - dma_data = &s3c_ac97_pcm_in; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - return 0; -} - static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -279,21 +261,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return -ENODEV; - else - snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); - - return 0; -} - static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -329,15 +296,27 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops s3c_ac97_dai_ops = { - .hw_params = s3c_ac97_hw_params, .trigger = s3c_ac97_trigger, }; static const struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { - .hw_params = s3c_ac97_hw_mic_params, .trigger = s3c_ac97_mic_trigger, }; +static int s3c_ac97_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, &s3c_ac97_pcm_out, &s3c_ac97_pcm_in); + + return 0; +} + +static int s3c_ac97_mic_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, NULL, &s3c_ac97_mic_in); + + return 0; +} + static struct snd_soc_dai_driver s3c_ac97_dai[] = { [S3C_AC97_DAI_PCM] = { .name = "samsung-ac97", @@ -354,6 +333,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_dai_probe, .ops = &s3c_ac97_dai_ops, }, [S3C_AC97_DAI_MIC] = { @@ -365,6 +345,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 1, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_mic_dai_probe, .ops = &s3c_ac97_mic_dai_ops, }, }; diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index fe2748b..ee23194 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -441,6 +441,14 @@ static struct snd_soc_platform_driver samsung_asoc_platform = { .pcm_free = dma_free_dma_buffers, }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture) +{ + snd_soc_dai_init_dma_data(dai, playback, capture); +} +EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); + int samsung_asoc_dma_platform_register(struct device *dev) { return snd_soc_register_platform(dev, &samsung_asoc_platform); diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 0e86315..fb09a1c 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -22,6 +22,9 @@ struct s3c_dma_params { char *ch_name; }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture); int samsung_asoc_dma_platform_register(struct device *dev); void samsung_asoc_dma_platform_unregister(struct device *dev); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index a5cbdb4..67d9fa9 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -963,7 +963,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) } clk_prepare_enable(i2s->clk); - snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); + samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); if (other) { other->addr = i2s->addr; diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index e54256f..6a5e4bf 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -275,7 +275,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct s3c_dma_params *dma_data; void __iomem *regs = pcm->regs; struct clk *clk; int sclk_div, sync_div; @@ -284,13 +283,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(pcm->dev, "Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = pcm->dma_playback; - else - dma_data = pcm->dma_capture; - - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); - /* Strictly check for sample size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -461,10 +453,20 @@ static const struct snd_soc_dai_ops s3c_pcm_dai_ops = { .set_fmt = s3c_pcm_set_fmt, }; +static int s3c_pcm_dai_probe(struct snd_soc_dai *dai) +{ + struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, pcm->dma_playback, pcm->dma_capture); + + return 0; +} + #define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 #define S3C_PCM_DAI_DECLARE \ .symmetric_rates = 1, \ + .probe = s3c_pcm_dai_probe, \ .ops = &s3c_pcm_dai_ops, \ .playback = { \ .channels_min = 2, \ -- 1.8.4.3

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[alsa-devel] [PATCH 0/2] ARM: Rename ARCH_SHMOBILE to ARCH_SHMOBILE_LEGACY
by Laurent Pinchart 09 Dec '13

09 Dec '13

Hello, This patch series enables compilation of SH-Mobile drivers on both the non-multiplatform (ARCH_SHMOBILE) and the multiplatform (ARCH_SHMOBILE_MULTI) SH-Mobile architectures. To do so it renames ARCH_SHMOBILE to ARCH_SHMOBILE_LEGACY and creates a new ARCH_SHMOBILE symbol selected by both ARCH_SHMOBILE_MULTI and ARCH_SHMOBILE_LEGACY. The second patch adds a dependency on ARCH_SHMOBILE_LEGACY to the ASoC R-Car driver. It can be squashed with the first patch or applied separately to minimize the risk of conflicts if desired. Laurent Pinchart (2): ARM: Rename ARCH_SHMOBILE to ARCH_SHMOBILE_LEGACY ASoC: sh: rcar: Depend on ARCH_SHMOBILE_LEGACY arch/arm/Kconfig | 14 ++++++++------ arch/arm/Makefile | 1 - arch/arm/boot/compressed/Makefile | 2 +- arch/arm/boot/dts/Makefile | 2 +- arch/arm/mach-shmobile/Kconfig | 8 ++++++-- drivers/Makefile | 2 +- sound/soc/sh/Kconfig | 1 + 7 files changed, 18 insertions(+), 12 deletions(-) -- Regards, Laurent Pinchart

4 7

[alsa-devel] [PATCH][RESEND 0/8] Add and implement gen_pool_dma_alloc()
by Nicolin Chen 09 Dec '13

09 Dec '13

Previously, we don't have a specific gen_pool_alloc() for DMA usage; Instead, we need to use gen_pool_virt_to_phys() to convert the addr returned from gen_pool_alloc(). So each implementation of this has duplicated code. Thus add new helper function -- gen_pool_dma_alloc(). After gen_pool_dma_alloc() is introduced, we can replace the original combination of gen_pool_alloc()+gen_pool_virt_to_phys() with this new helper function. Thus this patch implement the helper function to all the current drivers which use gen_pool_virt_to_phys(). !!-------------------important------------------!! The later 7 patches need all related driver owner to test. We can here define a simple rule: 1, If one driver owner or maintainer doesn't like the mofication to his/her driver, just let me know. I would drop that patch. 2, If there's a bug and issue found after patch-testing, please reply the mail so that I can fix and refine the patch. 3, If one driver owner or maintainer is too busy and doesn't have bandwidth to test the patch, I would drop that patch from this series. We can reimplement it till there's someone test it. !!-------------------current progress-----------!! lib/genalloc: [Okay] ARM: davinci: [Untested] dma: mmp_tdma: [Untested] [media] coda: [Untested] uio: uio_pruss: [Untested] ALSA: memalloc: [Tested] by Nicolin Chen with i.MX6Q SabreSD ASoC: davinci: [Untested] ASoC: pxa: use [Untested] Nicolin Chen (8): lib/genalloc: add a helper function for DMA buffer allocation ARM: davinci: use gen_pool_dma_alloc() to sram.c dma: mmp_tdma: use gen_pool_dma_alloc() to allocate descriptor [media] coda: use gen_pool_dma_alloc() to allocate iram buffer uio: uio_pruss: use gen_pool_dma_alloc() to allocate sram memory ALSA: memalloc: use gen_pool_dma_alloc() to allocate iram buffer ASoC: davinci: use gen_pool_dma_alloc() in davinci-pcm.c ASoC: pxa: use gen_pool_dma_alloc() to allocate dma buffer arch/arm/mach-davinci/sram.c | 9 +-------- drivers/dma/mmp_tdma.c | 7 +------ drivers/media/platform/coda.c | 5 ++--- drivers/uio/uio_pruss.c | 6 ++---- include/linux/genalloc.h | 2 ++ lib/genalloc.c | 28 ++++++++++++++++++++++++++++ sound/core/memalloc.c | 6 +----- sound/soc/davinci/davinci-pcm.c | 3 +-- sound/soc/pxa/mmp-pcm.c | 3 +-- 9 files changed, 39 insertions(+), 30 deletions(-) -- 1.8.4

5 13

[alsa-devel] [PATCH] ASoC mc13783 codec: correct names
by Lambrecht Jürgen 06 Dec '13

06 Dec '13

Correct names to reality and for consistency: - "Speaker Amp Switch"->"Speaker Amp" - "Earpiece Amp Switch"->"Speaker Amp Switch" - "Loudspeaker enable"->"Loudspeaker Amp Switch" - "..Capture.."->"..Playback" (Line in..., CODEC..., ...Mixin Switch) - move "PCM Capture Volume" to bottom with other Capture controls Signed-off-by: Jürgen Lambrecht <J.Lambrecht(a)televic.com> --- sound/soc/codecs/mc13783.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 08fb022..878e3e5 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -534,7 +534,7 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 0, &cdcout_ctl), - SND_SOC_DAPM_SWITCH("Speaker Amp Switch", MC13783_AUDIO_RX0, 3, 0, + SND_SOC_DAPM_SWITCH("Speaker Amp", MC13783_AUDIO_RX0, 3, 0, &samp_ctl), SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), SND_SOC_DAPM_SWITCH("Headset Amp Left", MC13783_AUDIO_RX0, 10, 0, @@ -590,7 +590,7 @@ static struct snd_soc_dapm_route mc13783_routes[] = { { "HSR", NULL, "Headset Amp Right"}, { "RXOUTL", NULL, "Line out Amp Left"}, { "RXOUTR", NULL, "Line out Amp Right"}, - { "SP", "Speaker Amp Switch", "Speaker Amp Source MUX"}, + { "SP", "Speaker Amp", "Speaker Amp Source MUX"}, { "LSP", "Loudspeaker Amp", "Loudspeaker Amp Source MUX"}, { "HSL", "Headset Amp Left", "Headset Amp Source MUX"}, { "HSR", "Headset Amp Right", "Headset Amp Source MUX"}, @@ -609,26 +609,27 @@ static const struct soc_enum mc13783_enum_3d_mixer = mc13783_3d_mixer); static struct snd_kcontrol_new mc13783_control_list[] = { - SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0), SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0), SOC_SINGLE("PCM Playback Switch", MC13783_AUDIO_RX1, 5, 1, 0), - SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), SOC_ENUM("3D Control", mc13783_enum_3d_mixer), SOC_SINGLE("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 1, 0), - SOC_SINGLE("Earpiece Amp Switch", MC13783_AUDIO_RX0, 3, 1, 0), + SOC_SINGLE("Speaker Amp Switch", MC13783_AUDIO_RX0, 3, 1, 0), + SOC_SINGLE("Loudspeaker Amp Switch", MC13783_AUDIO_RX0, 5, 1, 0), SOC_DOUBLE("Headset Amp Switch", MC13783_AUDIO_RX0, 10, 9, 1, 0), SOC_DOUBLE("Line out Amp Switch", MC13783_AUDIO_RX0, 16, 15, 1, 0), - SOC_SINGLE("CODEC Capture Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0), - SOC_SINGLE("PCM Capture Mixin Switch", MC13783_AUDIO_RX0, 22, 1, 0), - SOC_SINGLE("Line in Capture Mixin Switch", MC13783_AUDIO_RX0, 23, 1, 0), + SOC_SINGLE("CODEC Playback Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0), + SOC_SINGLE("PCM Playback Mixin Switch", MC13783_AUDIO_RX0, 22, 1, 0), + SOC_SINGLE("Line in Playback Mixin Switch", MC13783_AUDIO_RX0, 23, 1, 0), + + SOC_SINGLE("CODEC Playback Volume", MC13783_AUDIO_RX1, 1, 15, 0), + SOC_SINGLE("CODEC Playback Switch", MC13783_AUDIO_RX1, 0, 1, 0), - SOC_SINGLE("CODEC Capture Volume", MC13783_AUDIO_RX1, 1, 15, 0), - SOC_SINGLE("CODEC Capture Switch", MC13783_AUDIO_RX1, 0, 1, 0), + SOC_SINGLE("Line in Playback Volume", MC13783_AUDIO_RX1, 12, 15, 0), + SOC_SINGLE("Line in Playback Switch", MC13783_AUDIO_RX1, 10, 1, 0), - SOC_SINGLE("Line in Capture Volume", MC13783_AUDIO_RX1, 12, 15, 0), - SOC_SINGLE("Line in Capture Switch", MC13783_AUDIO_RX1, 10, 1, 0), + SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), SOC_SINGLE("MC1 Capture Bias Switch", MC13783_AUDIO_TX, 0, 1, 0), SOC_SINGLE("MC2 Capture Bias Switch", MC13783_AUDIO_TX, 1, 1, 0), -- 1.7.10.4

2 2

[alsa-devel] [PATCH] ASoC: samsung: Initialize the dma_data for secondary dai
by Padmavathi Venna 05 Dec '13

05 Dec '13

This patch initializes the dma_data for the secondary dai. Without this patch there is crash due to NULL pointer dereference during boot time on systems where secondary dai is initialized. Below is the crash log: [ 1.630000] smdk-audio-wm8894 sound.7: wm8994-aif1 <-> 3830000.i2s mapping ok [ 1.630000] Unable to handle kernel NULL pointer dereference at virtual address 00000010 [ 1.630000] pgd = c0004000 [ 1.630000] [00000010] *pgd=00000000 [ 1.630000] Internal error: Oops: 5 [#1] PREEMPT SMP ARM [ 1.630000] Modules linked in: ... ... [<c0342d6c>] (dmaengine_pcm_new+0x284/0x2ac) from [<c0341ae8>] (soc_new_pcm+0x2fc/0x3f4) [<c0341ae8>] (soc_new_pcm+0x2fc/0x3f4) from [<c0338558>] (snd_soc_register_card+0x1898/0x18d8) [<c0338558>] (snd_soc_register_card+0x1898/0x18d8) from [<c034cc34>] (smdk_audio_probe+0xa8/0xe8) [<c034cc34>] (smdk_audio_probe+0xa8/0xe8) from [<c0252368>] (platform_drv_probe+0x18/0x1c) [<c0252368>] (platform_drv_probe+0x18/0x1c) from [<c02510a4>] (driver_probe_device+0xf4/0x204) [<c02510a4>] (driver_probe_device+0xf4/0x204) from [<c0251240>] (__driver_attach+0x8c/0x90) [<c0251240>] (__driver_attach+0x8c/0x90) from [<c024f70c>] (bus_for_each_dev+0x54/0x88) [<c024f70c>] (bus_for_each_dev+0x54/0x88) from [<c0250724>] (bus_add_driver+0xe0/0x2b8) [<c0250724>] (bus_add_driver+0xe0/0x2b8) from [<c0251868>] (driver_register+0x78/0xf4) [<c0251868>] (driver_register+0x78/0xf4) from [<c00086e8>] (do_one_initcall+0x34/0x150) [<c00086e8>] (do_one_initcall+0x34/0x150) from [<c0543c30>] (kernel_init_freeable+0x12c/0x1d0) [<c0543c30>] (kernel_init_freeable+0x12c/0x1d0) from [<c03ebf9c>] (kernel_init+0x8/0xe8) [<c03ebf9c>] (kernel_init+0x8/0xe8) from [<c000e978>] (ret_from_fork+0x14/0x3c) Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> --- This is made based on Mark Brown for-next branch of sound.git This patch is based on Mark Brown below patches for generic dma engine support for samsung audio IPs. ASoC: samsung: Use ASoC dmaengine code where possible ASoC: samsung: Provide helper for DMA init sound/soc/samsung/dmaengine.c | 3 ++- sound/soc/samsung/i2s.c | 5 ++++- 2 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c index ad0a371..3f60479 100644 --- a/sound/soc/samsung/dmaengine.c +++ b/sound/soc/samsung/dmaengine.c @@ -67,7 +67,8 @@ EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); int samsung_asoc_dma_platform_register(struct device *dev) { return snd_dmaengine_pcm_register(dev, &samsung_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_COMPAT); + SND_DMAENGINE_PCM_FLAG_COMPAT | + SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME); } EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 67d9fa9..0bcd7fb 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -946,8 +946,11 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai; - if (other && other->clk) /* If this is probe on secondary */ + if (other && other->clk) { /* If this is probe on secondary */ + samsung_asoc_init_dma_data(dai, &other->sec_dai->dma_playback, + NULL); goto probe_exit; + } i2s->addr = ioremap(i2s->base, 0x100); if (i2s->addr == NULL) { -- 1.7.4.4

3 3

[alsa-devel] alsa/asoundlib.h file not found
by Appalayagari Sreedhar 05 Dec '13

05 Dec '13

Hi, I have integrated the codec driver and the kernel log shows the sound card is detected. now i want to test the codec. i tried the sample test code for capture and playback. i am getting a compilation errors due to the asoundlib.h. please find the error log for my application. main.c:2:23: error: asoundlib.h: No such file or directory main.c: In function 'main': main.c:7: error: 'snd_pcm_t' undeclared (first use in this function) main.c:7: error: (Each undeclared identifier is reported only once main.c:7: error: for each function it appears in.) main.c:7: error: 'pcm_handle' undeclared (first use in this function) main.c:10: error: 'snd_pcm_stream_t' undeclared (first use in this function) main.c:10: error: expected ';' before 'stream' main.c:15: error: 'snd_pcm_hw_params_t' undeclared (first use in this function) main.c:15: error: 'hwparams' undeclared (first use in this function) main.c:22: warning: 'return' with a value, in function returning void main.c:34: error: 'snd_pcm_uframes_t' undeclared (first use in this function) main.c:34: error: expected ';' before 'periodsize' main.c:46: error: 'SND_PCM_ACCESS_RW_INTERLEAVED' undeclared (first use in this function) main.c:48: warning: 'return' with a value, in function returning void main.c:52: error: 'SND_PCM_FORMAT_S16_LE' undeclared (first use in this function) main.c:54: warning: 'return' with a value, in function returning void main.c:62: warning: 'return' with a value, in function returning void main.c:65:19: warning: missing terminating " character main.c:65: error: missing terminating " character main.c:66: error: expected expression before '==' token main.c:66: error: stray '\' in program main.c:66:31: warning: missing terminating " character main.c:66: error: missing terminating " character main.c:67: error: expected ';' before '}' token main.c:72: warning: 'return' with a value, in function returning void main.c:78: warning: 'return' with a value, in function returning void main.c:86: error: 'periodsize' undeclared (first use in this function) main.c:88: warning: 'return' with a value, in function returning void main.c:99: warning: 'return' with a value, in function returning void main.c:112: warning: incompatible implicit declaration of built-in function 'malloc' main.c:115: error: 'num_frames' undeclared (first use in this function) main.c:5: warning: return type of 'main' is not 'int' please suggest what needs to be done to resolve this issue asap. Thank you, Sreedhar.

5 4

[alsa-devel] [PATCH v2 0/8] ARM: s3c64xx: Let amba-pl08x driver handle DMA
by Tomasz Figa 04 Dec '13

04 Dec '13

Since the patches extending support of amba-pl08x DMA engine driver to PL080S DMA engine (PL080 modified by Samsung) found in Samsung S3C64xx SoCs got merged into Linux 3.12, it is a good time to migrate the platform to use this driver and drop the legacy DMA code. Due to changes scattered across different areas of kernel, patches are based on merged 4 branches: - for-next of Kukjin Kim's Samsung tree, - clk-next of Mike Turquette's clock tree, - for-next of Mark Brown's SPI tree, - for-next of Mark Brown's ASoC tree. To ease testing I have prepared a branch in my private tree for anyone willing to check the patches out: git://github.com/tom3q/linux.git s3c64xx-pl08x-patches On S3C64xx-based Mini6410 and SMDK6410 boards, with I2S audio playback and capture (including full duplex operation) and also SPI using spidev: Tested-by: Tomasz Figa <tomasz.figa(a)gmail.com> Changes since v1: - Dropped the now merged patches adding PL080s support amba-pl08x driver, - Rebased onto current for-next branches of respective trees. - Stopped hijacking pl330_filter() symbol to plug pl08_filter_id() as channel filtering callback on S3C64xx. Changes since RFC v2: - Added clkdev lookups to old clock driver. - Added patches removing the old DMA driver and any remaining code needed by it. - Fixed DMA support for SPI. - Added a word about PL080S to amba-pl08x.c file header. - Changed definition of LLI words from enums to macros. - Extended debugging messages to handle PL080S variant as well. - Little cleanup of LLI dumping code. - Added check for peripheral flow control, which is unsupported by PL080S to dma_set_runtime_config. - Corrected transfer size mask of PL080S. Changes since RFC v1: - Returned to original way of storing quirks as booleans, as suggested by Russell, Linus and Arnd. - Added reg_config field to pl08x_phy_chan struct, which stores variant-specific address of channel config register, as suggested by Russell. - Simplified handling of extended maximum transfer size of PL080S (no more conditional passing of 0 as length to pl08x_cctl_bits()). - Reworked LLI handling in the driver to stop casting arbitrary memory to a struct and allow different word count of LLI entry, as suggested by Linus. - Removed AMBA ID override from S3C64xx PL080 initialization code. - Fixed brokenness of Samsung DMA wrapper API, which caused cyclic buffers to be queued multiple times when DMA engine is used. - Included patch adding clock aliases for DMA engines (depends on Common Clock Framework driver for S3C64xx). - Fixed several minor stylistic issues. Tomasz Figa (8): spi: s3c64xx: Do not require legacy DMA API in case of S3C64XX ASoC: Samsung: Do not require legacy DMA API in case of S3C64XX ARM: s3c64xx: Add support for DMA using generic amba-pl08x driver clk: samsung: s3c64xx: Add aliases for DMA clocks ARM: s3c64xx: Remove legacy DMA driver clk: samsung: s3c64xx: Remove clock aliases of old DMA driver spi: s3c64xx: Always select S3C64XX_PL080 when ARCH_S3C64XX is enabled ASoC: Samsung: Always select S3C64XX_PL080 when ARCH_S3C64XX is enabled arch/arm/Kconfig | 1 + arch/arm/mach-s3c64xx/Kconfig | 7 +- arch/arm/mach-s3c64xx/Makefile | 2 +- arch/arm/mach-s3c64xx/common.h | 5 + arch/arm/mach-s3c64xx/dma.c | 762 ------------------------------- arch/arm/mach-s3c64xx/include/mach/dma.h | 144 ++---- arch/arm/mach-s3c64xx/pl080.c | 244 ++++++++++ arch/arm/plat-samsung/devs.c | 10 +- arch/arm/plat-samsung/dma-ops.c | 8 +- drivers/clk/samsung/clk-s3c64xx.c | 4 +- drivers/spi/Kconfig | 2 +- sound/soc/samsung/Kconfig | 2 +- 12 files changed, 316 insertions(+), 875 deletions(-) delete mode 100644 arch/arm/mach-s3c64xx/dma.c create mode 100644 arch/arm/mach-s3c64xx/pl080.c -- 1.8.3.2

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[alsa-devel] [PATCH 07/10] ASoC: ux500: Store DMA data in the DAI differently in the pdata and DT case
by Lee Jones 04 Dec '13

04 Dec '13

In this patch we do two things. Firstly, instead of open coding the store of DMA data in to the DAI for later use, we use the API provided. Secondly we create and store similar DMA data for the DT case, only this time we use 'struct snd_dmaengine_dai_dma_data' which is provided by the core for this very reason. Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)kernel.org> Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/ux500_msp_dai.c | 42 +++++++++++++++++++++++++++++++++++++++-- 1 file changed, 40 insertions(+), 2 deletions(-) diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index c6fb5cc..8d1b01f 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -17,12 +17,14 @@ #include <linux/bitops.h> #include <linux/platform_device.h> #include <linux/clk.h> +#include <linux/of.h> #include <linux/regulator/consumer.h> #include <linux/mfd/dbx500-prcmu.h> #include <linux/platform_data/asoc-ux500-msp.h> #include <sound/soc.h> #include <sound/soc-dai.h> +#include <sound/dmaengine_pcm.h> #include "ux500_msp_i2s.h" #include "ux500_msp_dai.h" @@ -654,16 +656,52 @@ static int ux500_msp_dai_trigger(struct snd_pcm_substream *substream, return ret; } +static int ux500_msp_dai_of_probe(struct snd_soc_dai *dai) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + struct snd_dmaengine_dai_dma_data *playback_dma_data; + struct snd_dmaengine_dai_dma_data *capture_dma_data; + + playback_dma_data = devm_kzalloc(dai->dev, + sizeof(*playback_dma_data), + GFP_KERNEL); + if (!playback_dma_data) + return -ENOMEM; + + capture_dma_data = devm_kzalloc(dai->dev, + sizeof(*capture_dma_data), + GFP_KERNEL); + if (!capture_dma_data) + return -ENOMEM; + + playback_dma_data->addr = drvdata->msp->playback_dma_data.tx_rx_addr; + capture_dma_data->addr = drvdata->msp->capture_dma_data.tx_rx_addr; + + playback_dma_data->maxburst = 4; + capture_dma_data->maxburst = 4; + + snd_soc_dai_init_dma_data(dai, playback_dma_data, capture_dma_data); + + return 0; +} + static int ux500_msp_dai_probe(struct snd_soc_dai *dai) { struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + struct device_node *np = dai->dev->of_node; + int ret; - dai->playback_dma_data = &drvdata->msp->playback_dma_data; - dai->capture_dma_data = &drvdata->msp->capture_dma_data; + if (np) { + ret = ux500_msp_dai_of_probe(dai); + return ret; + } drvdata->msp->playback_dma_data.data_size = drvdata->slot_width; drvdata->msp->capture_dma_data.data_size = drvdata->slot_width; + snd_soc_dai_init_dma_data(dai, + &drvdata->msp->playback_dma_data, + &drvdata->msp->capture_dma_data); return 0; } -- 1.8.1.2

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[alsa-devel] [TINYCOMPRESS PATCH 1/2] crec: Initial version of a compressed capture utility
by Charles Keepax 03 Dec '13

03 Dec '13

This version only supports capture of PCM streams over a compressed device and saves these as WAV files. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Android.mk | 10 ++ crec.c | 425 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ makefile.linux | 20 ++- 3 files changed, 448 insertions(+), 7 deletions(-) create mode 100644 crec.c diff --git a/Android.mk b/Android.mk index 104dfef..3cdd3d0 100644 --- a/Android.mk +++ b/Android.mk @@ -20,3 +20,13 @@ LOCAL_MODULE_TAGS := optional include $(BUILD_EXECUTABLE) +include $(CLEAR_VARS) + +LOCAL_C_INCLUDES:= $(LOCAL_PATH)/include +LOCAL_SRC_FILES:= crec.c +LOCAL_MODULE := crec +LOCAL_SHARED_LIBRARIES:= libcutils libutils libtinycompress +LOCAL_MODULE_TAGS := optional + +include $(BUILD_EXECUTABLE) + diff --git a/crec.c b/crec.c new file mode 100644 index 0000000..d2fd873 --- /dev/null +++ b/crec.c @@ -0,0 +1,425 @@ +/* + * BSD LICENSE + * + * crec command line recorder for compress audio record in alsa + * Copyright (c) 2011-2012, Intel Corporation + * Copyright (c) 2013, Wolfson Microelectronic Ltd. + * All rights reserved. + * + * Author: Vinod Koul <vinod.koul(a)linux.intel.com> + * Author: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions are met: + * + * Redistributions of source code must retain the above copyright notice, + * this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * Neither the name of Intel Corporation nor the names of its contributors + * may be used to endorse or promote products derived from this software + * without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + * LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF + * THE POSSIBILITY OF SUCH DAMAGE. + * + * LGPL LICENSE + * + * crec command line recorder for compress audio record in alsa + * Copyright (c) 2011-2012, Intel Corporation + * Copyright (c) 2013, Wolfson Microelectronic Ltd. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU Lesser General Public License, + * version 2.1, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public + * License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to + * the Free Software Foundation, Inc., + * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include <stdint.h> +#include <linux/types.h> +#include <fcntl.h> +#include <errno.h> +#include <unistd.h> +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <signal.h> +#include <stdbool.h> +#include <getopt.h> +#include <sys/time.h> +#define __force +#define __bitwise +#define __user +#include "sound/compress_params.h" +#include "sound/compress_offload.h" +#include "tinycompress/tinycompress.h" + +static int verbose; + +static const unsigned int DEFAULT_CHANNELS = 1; +static const unsigned int DEFAULT_RATE = 44100; +static const unsigned int DEFAULT_FORMAT = SNDRV_PCM_FORMAT_S16_LE; + +struct riff_chunk { + char desc[4]; + uint32_t size; +} __attribute__((__packed__)); + +struct wave_header { + struct { + struct riff_chunk chunk; + char format[4]; + } __attribute__((__packed__)) riff; + + struct { + struct riff_chunk chunk; + uint16_t type; + uint16_t channels; + uint32_t rate; + uint32_t byterate; + uint16_t blockalign; + uint16_t samplebits; + } __attribute__((__packed__)) fmt; + + struct { + struct riff_chunk chunk; + } __attribute__((__packed__)) data; +} __attribute__((__packed__)); + +const struct wave_header blank_wave_header = { + .riff = { + .chunk = { + .desc = "RIFF", + }, + .format = "WAVE", + }, + .fmt = { + .chunk = { + .desc = "fmt ", /* Note the space is important here */ + .size = sizeof(blank_wave_header.fmt) - + sizeof(blank_wave_header.fmt.chunk), + }, + .type = 0x01, /* PCM */ + }, + .data = { + .chunk = { + .desc = "data", + }, + }, +}; + +static void init_wave_header(struct wave_header *header, uint16_t channels, + uint32_t rate, uint16_t samplebits) +{ + memcpy(header, &blank_wave_header, sizeof(blank_wave_header)); + + header->fmt.channels = channels; + header->fmt.rate = rate; + header->fmt.byterate = channels * rate * (samplebits / 8); + header->fmt.blockalign = channels * (samplebits / 8); + header->fmt.samplebits = samplebits; +} + +static void size_wave_header(struct wave_header *header, uint32_t size) +{ + header->riff.chunk.size = sizeof(*header) - + sizeof(header->riff.chunk) + size; + header->data.chunk.size = size; +} + +static void usage(void) +{ + fprintf(stderr, "usage: crec [OPTIONS] filename\n" + "-c\tcard number\n" + "-d\tdevice node\n" + "-b\tbuffer size\n" + "-f\tfragments\n" + "-v\tverbose mode\n" + "-l\tlength of record in seconds\n" + "-h\tPrints this help list\n\n" + "-C\tSpecify the number of channels (default %u)\n" + "-R\tSpecify the sample rate (default %u)\n" + "-F\tSpecify the format: S16_LE, S32_LE (default S16_LE)\n\n" + "Example:\n" + "\tcrec -c 1 -d 2 test.wav\n" + "\tcrec -f 5 test.wav\n", + DEFAULT_CHANNELS, DEFAULT_RATE); + + exit(EXIT_FAILURE); +} + +static int print_time(struct compress *compress) +{ + unsigned int avail; + struct timespec tstamp; + + if (compress_get_hpointer(compress, &avail, &tstamp) != 0) { + fprintf(stderr, "Error querying timestamp\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + return -1; + } else { + printf("DSP recorded %jd.%jd\n", + (intmax_t)tstamp.tv_sec, (intmax_t)tstamp.tv_nsec*1000); + } + return 0; +} + +void capture_samples(char *name, unsigned int card, unsigned int device, + unsigned long buffer_size, unsigned int frag, + unsigned int length, unsigned int rate, + unsigned int channels, unsigned int format) +{ + struct compr_config config; + struct snd_codec codec; + struct compress *compress; + struct wave_header header; + FILE *file; + char *buffer; + size_t written; + int read, ret; + unsigned int size, total_read = 0; + unsigned int samplebits; + + switch (format) { + case SNDRV_PCM_FORMAT_S32_LE: + samplebits = 32; + break; + default: + samplebits = 16; + break; + } + + /* Convert length from seconds to bytes */ + length = length * rate * (samplebits / 8) * channels; + + if (verbose) + printf("%s: entry, reading %u bytes\n", __func__, length); + + file = fopen(name, "w+b"); + if (!file) { + fprintf(stderr, "Unable to open file '%s'\n", name); + exit(EXIT_FAILURE); + } + + /* Write a header, will update with size once record is complete */ + init_wave_header(&header, channels, rate, samplebits); + written = fwrite(&header, sizeof(header), 1, file); + if (written != 1) { + fprintf(stderr, "Error writing output file header: %d\n", + ferror(file)); + goto file_exit; + } + + codec.id = SND_AUDIOCODEC_PCM; + codec.ch_in = channels; + codec.ch_out = channels; + codec.sample_rate = compress_get_alsa_rate(rate); + if (!codec.sample_rate) { + fprintf(stderr, "invalid sample rate %d\n", rate); + goto file_exit; + } + codec.bit_rate = 0; + codec.rate_control = 0; + codec.profile = 0; + codec.level = 0; + codec.ch_mode = 0; + codec.format = format; + if ((buffer_size != 0) && (frag != 0)) { + config.fragment_size = buffer_size/frag; + config.fragments = frag; + } else { + /* use driver defaults */ + config.fragment_size = 0; + config.fragments = 0; + } + config.codec = &codec; + + compress = compress_open(card, device, COMPRESS_OUT, &config); + if (!compress || !is_compress_ready(compress)) { + fprintf(stderr, "Unable to open Compress device %d:%d\n", + card, device); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto file_exit; + }; + + if (verbose) + printf("%s: Opened compress device\n", __func__); + + size = config.fragment_size; + buffer = malloc(size * config.fragments); + if (!buffer) { + fprintf(stderr, "Unable to allocate %d bytes\n", size); + goto comp_exit; + } + + printf("Recording file %s On Card %u device %u, with buffer of %lu bytes\n", + name, card, device, buffer_size); + printf("Format %u Channels %u, %u Hz\n", + codec.id, codec.ch_out, rate); + + compress_start(compress); + + if (verbose) + printf("%s: Capturing audio NOW!!!\n", __func__); + + do { + if (length && size > length - total_read) + size = length - total_read; + + read = compress_read(compress, buffer, size); + if (read < 0) { + fprintf(stderr, "Error reading sample\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto buf_exit; + } + if ((unsigned int)read != size) { + fprintf(stderr, "We read %d, DSP sent %d\n", + size, read); + } + + if (read > 0) { + total_read += read; + + written = fwrite(buffer, 1, read, file); + if (written != (size_t)read) { + fprintf(stderr, "Error writing output file: %d\n", + ferror(file)); + goto buf_exit; + } + if (verbose) { + print_time(compress); + printf("%s: read %d\n", __func__, read); + } + } + } while (!length || total_read < length); + + ret = compress_stop(compress); + if (ret < 0) { + fprintf(stderr, "Error closing stream\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + } + + /* Update file header now we know file size */ + size_wave_header(&header, total_read); + ret = fseek(file, 0, SEEK_SET); + if (ret < 0) { + fprintf(stderr, "Error seeking: %s\n", stderror(errno)); + goto buf_exit; + } + written = fwrite(&header, sizeof(header), 1, file); + if (written != 1) { + fprintf(stderr, "Error updating output file header: %d\n", + ferror(file)); + goto buf_exit; + } + + if (verbose) + printf("%s: exit success\n", __func__); + + free(buffer); + fclose(file); + + compress_close(compress); + + return; +buf_exit: + free(buffer); +comp_exit: + compress_close(compress); +file_exit: + fclose(file); + + if (verbose) + printf("%s: exit failure\n", __func__); + + exit(EXIT_FAILURE); +} + +int main(int argc, char **argv) +{ + char *file; + unsigned long buffer_size = 0; + int c; + unsigned int card = 0, device = 0, frag = 0, length = 0; + unsigned int rate = DEFAULT_RATE, channels = DEFAULT_CHANNELS; + unsigned int format = DEFAULT_FORMAT; + + if (argc < 2) + usage(); + + verbose = 0; + while ((c = getopt(argc, argv, "hvl:R:C:F:b:f:c:d:")) != -1) { + switch (c) { + case 'h': + usage(); + break; + case 'b': + buffer_size = strtol(optarg, NULL, 0); + break; + case 'f': + frag = strtol(optarg, NULL, 10); + break; + case 'c': + card = strtol(optarg, NULL, 10); + break; + case 'd': + device = strtol(optarg, NULL, 10); + break; + case 'v': + verbose = 1; + break; + case 'l': + length = strtol(optarg, NULL, 10); + break; + case 'R': + rate = strtol(optarg, NULL, 10); + break; + case 'C': + channels = strtol(optarg, NULL, 10); + break; + case 'F': + if (strcmp(optarg, "S16_LE") == 0) { + format = SNDRV_PCM_FORMAT_S16_LE; + } else if (strcmp(optarg, "S32_LE") == 0) { + format = SNDRV_PCM_FORMAT_S32_LE; + } else { + fprintf(stderr, "Unrecognised format: %s\n", + optarg); + usage(); + } + break; + default: + exit(EXIT_FAILURE); + } + } + if (optind >= argc) + usage(); + + file = argv[optind]; + + capture_samples(file, card, device, buffer_size, frag, length, + rate, channels, format); + + printf("Finish capturing... Close Normally\n"); + exit(EXIT_SUCCESS); +} + diff --git a/makefile.linux b/makefile.linux index 677878d..81782d0 100644 --- a/makefile.linux +++ b/makefile.linux @@ -1,10 +1,13 @@ LIB = libtinycompress -BIN = cplay VER = 0.1 LIBSRC = compress.c -SRC = cplay.c utils.c +PLAYBIN = cplay +RECBIN = crec +PLAYSRC = cplay.c utils.c +RECSRC = crec.c utils.c LIBOBJ = ${LIBSRC:.c=.o} -OBJ = ${SRC:.c=.o} +PLAYOBJ = ${PLAYSRC:.c=.o} +RECOBJ = ${RECSRC:.c=.o} CC = gcc CROSS_COMPILE = @@ -13,23 +16,26 @@ CFLAGS += -std=c99 -Wall -Wextra -Wunused -DVERSION=\"${VER}\" -I./include LDFLAGS += -L. -ltinycompress LIBLDFLAGS = -lasound -all: libtinycompress cplay +all: libtinycompress cplay crec $(LIB): ${LIBOBJ} @echo " LD "$@ @${CROSS_COMPILE}${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} -$(BIN): ${OBJ} ${LIB} +$(PLAYBIN): ${PLAYOBJ} ${LIB} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${OBJ} ${LDFLAGS} + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${PLAYOBJ} ${LDFLAGS} +$(RECBIN): ${RECOBJ} ${LIB} + @echo " LD "$@ + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${RECOBJ} ${LDFLAGS} %.o: %.c @echo " CC "$< @${CROSS_COMPILE}${CC} ${CFLAGS} -c -fPIC -o $@ $< clean: - @rm -rf ${BIN} ${OBJ} ${LIB}.so ${LIBOBJ} + @rm -rf ${BIN} ${PLAYOBJ} ${RECOBJ} ${LIB}.so ${LIBOBJ} .PHONY: all clean -- 1.7.2.5

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Re: [alsa-devel] [PATCH v2] ASoC: simple-card: add Device Tree support
by Mark Brown 03 Dec '13

03 Dec '13

On Thu, Oct 03, 2013 at 02:33:55AM -0700, Kuninori Morimoto wrote: > ping ? Don't send contentless pings. I'm waiting for review from the device tree guys but I see now that you didn't CC them so they're unlikely to respond - please resend CCing them.

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