August 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] IEC61937 passthrough with BYT/CHT HdmiLpeAudio
by Pierre-Louis Bossart 16 Aug '17

16 Aug '17

Hi Takashi, While I was looking at IEC61937, I ran a quick test on my BYT and CHT boxes, and as I suspected IEC61937 works fine since your modifications to add s16_le support. I can get AC3 to my HDMI receiver without problems just by using the hw:0,<1,2,3> device, depending on which port is used. e.g. aplay -Dhw:0,2 -c2 -r48000 -fs16le ac3_surround_test.spdif To make things more usable for applications, and maybe support receivers who actually care about the AES0 non-audio bit or reduce the initial lock time, it'd be nice to support the traditional command line, e.g. aplay -D'hdmi:CARD=0,DEV=<0,1,2>,AES0=0x6' -c2 -r48000 -fs16_le ac3_surround_test.spdif However we seem to have a mix of alsa-lib configuration issues and maybe a miss at the driver level I added the missing devices in HdmiLpeAudio.conf with the patch below, which helps select the right output, but somehow the AES0 non-audio bit doesn't seem to be set. [plb@ZOTAC PASSTHROUGH]$ amixer -Dhw:0 cget numid=12 numid=12,iface=PCM,name='IEC958 Playback Default',device=2 ; type=IEC958,access=rw------,values=1 : values=[AES0=0x00 AES1=0x00 AES2=0x00 AES3=0x00] [plb@ZOTAC PASSTHROUGH]$ aplay -D'hdmi:CARD=0,DEV=2,AES0=0x6' -c2 -r48000 -fs16_le ac3_surround_test.spdif Playing raw data 'ac3_surround_test.spdif' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo ^CAborted by signal Interrupt... [plb@ZOTAC PASSTHROUGH]$ amixer -Dhw:0 cget numid=12numid=12,iface=PCM,name='IEC958 Playback Default',device=2 ; type=IEC958,access=rw------,values=1 : values=[AES0=0x00 AES1=0x00 AES2=0x00 AES3=0x00] Could it be that we are missing the 'index' which is used in the HDaudio case to access the 'IEC958 Playback Default' controls? Thanks -Pierre diff HdmiLpeAudio.conf /usr/share/alsa/cards/HdmiLpeAudio.conf 59,135d58 < device 0 < lock true < preserve true < value [ $AES0 $AES1 $AES2 $AES3 ] < } < ] < } < } < < HdmiLpeAudio.pcm.hdmi.1 { < @args [ CARD AES0 AES1 AES2 AES3 ] < @args.CARD { < type string < } < @args.AES0 { < type integer < } < @args.AES1 { < type integer < } < @args.AES2 { < type integer < } < @args.AES3 { < type integer < } < type hooks < slave.pcm { < type hw < card $CARD < device 1 < } < hooks.0 { < type ctl_elems < hook_args [ < { < interface PCM < name "IEC958 Playback Default" < device 1 < lock true < preserve true < value [ $AES0 $AES1 $AES2 $AES3 ] < } < ] < } < } < < HdmiLpeAudio.pcm.hdmi.2 { < @args [ CARD AES0 AES1 AES2 AES3 ] < @args.CARD { < type string < } < @args.AES0 { < type integer < } < @args.AES1 { < type integer < } < @args.AES2 { < type integer < } < @args.AES3 { < type integer < } < type hooks < slave.pcm { < type hw < card $CARD < device 2 < } < hooks.0 { < type ctl_elems < hook_args [ < { < interface PCM < name "IEC958 Playback Default" < device 2

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[alsa-devel] [PATCH] alsabat: fix the buffer size dependency issue
by Zhang Keqiao 16 Aug '17

16 Aug '17

In round trip latency test, the latency time is depends on the buffer size, before the test, it will playing one second 0 data to synchronize the playback thread and the recording thread, then begin playing a ~1000 HZ sine wave and record it. Stop counting and playing if the input's loudness is higher than the threshold. The issue here is before catching the sine wave, the buffer is full with 0 data need to deal with, so this latency is also counted. This patch is intends to use flag to do the thread synchronization instead of playing 0 data, once recording thread is ready, playing the ~1000 HZ sine wave and start over this process until the final test is over. Signed-off-by: Zhang Keqiao <keqiaox.k.zhang(a)linux.intel.com> --- bat/alsa.c | 26 +++++++++++++++++++++++--- bat/common.h | 4 +++- bat/latencytest.c | 47 +++++++++++++++++++++++++---------------------- 3 files changed, 51 insertions(+), 26 deletions(-) diff --git a/bat/alsa.c b/bat/alsa.c index 7613f44..5af7766 100644 --- a/bat/alsa.c +++ b/bat/alsa.c @@ -351,6 +351,14 @@ static int latencytest_process_output(struct pcm_container *sndpcm, bat->latency.is_playing = true; + /* set playback ready */ + bat->latency.state = LATENCY_STATE_PLAY_READY; + + /* wait for capture ready */ + while (bat->latency.state != LATENCY_STATE_RECORD_AND_LISTEN) { + usleep(10); + } + while (1) { /* generate output data */ err = handleoutput(bat, sndpcm->buffer, bytes, frames); @@ -365,10 +373,16 @@ static int latencytest_process_output(struct pcm_container *sndpcm, if (bat->latency.xrun_error == true) break; + /* Final test is over */ if (bat->latency.state == LATENCY_STATE_COMPLETE_SUCCESS) break; bat->periods_played++; + + /* One time success, start over */ + if (bat->latency.state == LATENCY_STATE_ONE_TIME_SUCCESS) + break; + } bat->latency.is_playing = false; @@ -441,6 +455,7 @@ void *playback_alsa(struct bat *bat) int err = 0; struct pcm_container sndpcm; +LATENCY_TEST_START: fprintf(bat->log, _("Entering playback thread (ALSA).\n")); retval_play = 0; @@ -501,6 +516,11 @@ exit3: free(sndpcm.buffer); exit2: snd_pcm_close(sndpcm.handle); + +/* Latency test is not finished, start over */ + if (bat->latency.is_capturing == true) + goto LATENCY_TEST_START; + exit1: pthread_exit(&retval_play); } @@ -641,15 +661,15 @@ static int latencytest_process_input(struct pcm_container *sndpcm, if (err != 0) break; - if (bat->latency.is_playing == false) - break; - /* write the chunk to file */ if (fwrite(sndpcm->buffer, 1, size, fp) != size) { err = -EIO; break; } + if (bat->latency.state == LATENCY_STATE_COMPLETE_SUCCESS) + break; + bytes_read += size; } diff --git a/bat/common.h b/bat/common.h index 1b07fbe..2ab3e86 100644 --- a/bat/common.h +++ b/bat/common.h @@ -158,8 +158,10 @@ enum latency_state { LATENCY_STATE_COMPLETE_FAILURE = -1, LATENCY_STATE_COMPLETE_SUCCESS = 0, LATENCY_STATE_MEASURE_FOR_1_SECOND, - LATENCY_STATE_PLAY_AND_LISTEN, + LATENCY_STATE_PLAY_READY, + LATENCY_STATE_RECORD_AND_LISTEN, LATENCY_STATE_WAITING, + LATENCY_STATE_ONE_TIME_SUCCESS, }; struct pcm { diff --git a/bat/latencytest.c b/bat/latencytest.c index fae191c..e7727e9 100644 --- a/bat/latencytest.c +++ b/bat/latencytest.c @@ -55,8 +55,6 @@ static void play_and_listen(struct bat *bat, void *buffer, int frames) int averageinput; int n = 0; float sum = 0; - float max = 0; - float min = 100000.0f; short int *input; int num = bat->latency.number; @@ -69,17 +67,17 @@ static void play_and_listen(struct bat *bat, void *buffer, int frames) /* Check the location when it became loud enough */ while (n < frames) { - if (*input++ > bat->latency.threshold) + if (*input > bat->latency.threshold) break; - *input += bat->channels; + input += bat->channels; n++; } /* Now we get the total round trip latency*/ bat->latency.samples += n; - /* Expect at least 1 buffer of round trip latency. */ - if (bat->latency.samples > frames) { + /* Round trip latency may less than one buffer */ + if (1) { bat->latency.result[num - 1] = (float) bat->latency.samples * 1000 / bat->rate; fprintf(bat->log, @@ -88,22 +86,14 @@ static void play_and_listen(struct bat *bat, void *buffer, int frames) (int) bat->latency.result[num - 1]); for (n = 0; n < num; n++) { - if (bat->latency.result[n] > max) - max = bat->latency.result[n]; - if (bat->latency.result[n] < min) - min = bat->latency.result[n]; sum += bat->latency.result[n]; } - - /* The maximum is higher than the minimum's double */ - if (max / min > 2.0f) { - bat->latency.state = - LATENCY_STATE_COMPLETE_FAILURE; - bat->latency.is_capturing = false; - return; + /* This is unnecessary, round trip lantency can be less than one + * period size, so this case is likely happen unless the period + * size is samll enough than the round trip latency. */ /* Final results */ - } else if (num == LATENCY_TEST_NUMBER) { + if (num == LATENCY_TEST_NUMBER) { bat->latency.final_result = (int) (sum / LATENCY_TEST_NUMBER); fprintf(bat->log, @@ -113,11 +103,12 @@ static void play_and_listen(struct bat *bat, void *buffer, int frames) bat->latency.state = LATENCY_STATE_COMPLETE_SUCCESS; bat->latency.is_capturing = false; + bat->latency.is_playing = false; return; /* Next step */ } else - bat->latency.state = LATENCY_STATE_WAITING; + bat->latency.state = LATENCY_STATE_ONE_TIME_SUCCESS; bat->latency.number++; @@ -194,17 +185,23 @@ int handleinput(struct bat *bat, void *buffer, int frames) /* 1 second elapsed */ if (bat->latency.samples >= bat->rate) { calculate_threshold(bat); - bat->latency.state = LATENCY_STATE_PLAY_AND_LISTEN; + /* Capture preparation is ready, start the latency test */ + bat->latency.state = LATENCY_STATE_RECORD_AND_LISTEN; bat->latency.samples = 0; bat->latency.sum = 0; } break; /* Playing sine wave and listening if it comes back */ - case LATENCY_STATE_PLAY_AND_LISTEN: + case LATENCY_STATE_RECORD_AND_LISTEN: play_and_listen(bat, buffer, frames); break; + /* playback is ready, trigger recording */ + case LATENCY_STATE_PLAY_READY: + bat->latency.state = LATENCY_STATE_MEASURE_FOR_1_SECOND; + break; + /* Waiting 1 second */ case LATENCY_STATE_WAITING: bat->latency.samples += frames; @@ -216,6 +213,12 @@ int handleinput(struct bat *bat, void *buffer, int frames) } break; + /* wait for next round playback ready */ + case LATENCY_STATE_ONE_TIME_SUCCESS: + if (bat->latency.state == LATENCY_STATE_PLAY_READY) + bat->latency.state = LATENCY_STATE_WAITING; + break; + default: return 0; } @@ -232,7 +235,7 @@ int handleoutput(struct bat *bat, void *buffer, int bytes, int frames) && bat->latency.is_capturing == false) return bat->latency.state; - if (bat->latency.state == LATENCY_STATE_PLAY_AND_LISTEN) + if (bat->latency.state == LATENCY_STATE_RECORD_AND_LISTEN) err = generate_sine_wave(bat, frames, buffer); else /* Output silence */ -- 2.9.3

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[alsa-devel] [PATCH v2] ASoC: samsung: i2s: Null pointer dereference on samsung_i2s_remove
by Anton Vasilyev 16 Aug '17

16 Aug '17

If (quirks & QUIRK_SEC_DAI == 0) then samsung_i2s_probe() doesn't allocate sec_dai and pri_dai->sec_dai remains Null, but samsung_i2s_remove() permorms pri_dai->sec_dai dereference in any case. The patch adds sec_dai check on Null before derefence at samsung_i2s_remove(). Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Anton Vasilyev <vasilyev(a)ispras.ru> --- v2: Drop initialization of sec_dai at samsung_i2s_remove as Sylwester Nawrocki suggest. --- sound/soc/samsung/i2s.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index af3ba4d..6f896e3 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1376,13 +1376,9 @@ static int samsung_i2s_probe(struct platform_device *pdev) static int samsung_i2s_remove(struct platform_device *pdev) { - struct i2s_dai *pri_dai, *sec_dai; + struct i2s_dai *pri_dai; pri_dai = dev_get_drvdata(&pdev->dev); - sec_dai = pri_dai->sec_dai; - - pri_dai->sec_dai = NULL; - sec_dai->pri_dai = NULL; pm_runtime_get_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); -- 2.7.4

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[alsa-devel] [PATCH] ASoC: compress: Set reasonable compress id string
by Peng Donglin 16 Aug '17

16 Aug '17

For dynamic compress rtd, the codec_dai may not have any actual meaning(like snd-soc-dummy-dai), so compress id can just use the value of dai_link->stream_name. But for others, its codec_dai has actual meaning, so compress id can include codec_dai name. Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> --- sound/soc/soc-compress.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 206f36b..b6a0714 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -737,9 +737,6 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) } /* check client and interface hw capabilities */ - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); - if (codec_dai->driver->playback.channels_min) playback = 1; if (codec_dai->driver->capture.channels_min) @@ -758,7 +755,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) return -EINVAL; } - if(playback) + if (playback) direction = SND_COMPRESS_PLAYBACK; else direction = SND_COMPRESS_CAPTURE; @@ -797,19 +794,18 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) else if (rtd->dai_link->dpcm_capture) be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); - } else + } else { + snprintf(new_name, sizeof(new_name), "%s %s-%d", + rtd->dai_link->stream_name, codec_dai->name, num); + memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); + } /* Add copy callback for not memory mapped DSPs */ if (platform->driver->compr_ops && platform->driver->compr_ops->copy) compr->ops->copy = soc_compr_copy; mutex_init(&compr->lock); - - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, - rtd->codec_dai->name, num); - ret = snd_compress_new(rtd->card->snd_card, num, direction, new_name, compr); if (ret < 0) { -- 1.9.1

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[alsa-devel] [PATCH 0/3] ALSA: make snd_kcontrol_new const
by Bhumika Goyal 16 Aug '17

16 Aug '17

Make these structures const. Done using Coccinelle. Bhumika Goyal (3): ALSA: aoa: make snd_kcontrol_new const ALSA: pcxhr: make snd_kcontrol_new const ALSA: hda: make snd_kcontrol_new const sound/aoa/codecs/onyx.c | 12 ++++++------ sound/pci/hda/patch_analog.c | 4 ++-- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/pcxhr/pcxhr_mixer.c | 4 ++-- 4 files changed, 11 insertions(+), 11 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Medfield: Delete an error message for a failed memory allocation in snd_mfld_mc_probe()
by SF Markus Elfring 16 Aug '17

16 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 11 Aug 2017 11:25:41 +0200 Omit an extra message for a memory allocation failure in this function. This issue was detected by using the Coccinelle software. Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_… Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/intel/boards/mfld_machine.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/mfld_machine.c b/sound/soc/intel/boards/mfld_machine.c index 4e08885f37aa..6f44acfb4aae 100644 --- a/sound/soc/intel/boards/mfld_machine.c +++ b/sound/soc/intel/boards/mfld_machine.c @@ -376,10 +376,8 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) /* audio interrupt base of SRAM location where * interrupts are stored by System FW */ mc_drv_ctx = devm_kzalloc(&pdev->dev, sizeof(*mc_drv_ctx), GFP_ATOMIC); - if (!mc_drv_ctx) { - pr_err("allocation failed\n"); + if (!mc_drv_ctx) return -ENOMEM; - } irq_mem = platform_get_resource_byname( pdev, IORESOURCE_MEM, "IRQ_BASE"); -- 2.14.0

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[alsa-devel] [PATCH] ASoC: codecs: make snd_compr_ops const
by Bhumika Goyal 16 Aug '17

16 Aug '17

Make these const as they are only stored in the compr_ops field of a snd_soc_platform_driver structure, which is of type const. Done using Coccinelle Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 505dbc9..e09fc8f 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1203,7 +1203,7 @@ static struct regmap *cs47l24_get_regmap(struct device *dev) }, }; -static struct snd_compr_ops cs47l24_compr_ops = { +static const struct snd_compr_ops cs47l24_compr_ops = { .open = cs47l24_open, .free = wm_adsp_compr_free, .set_params = wm_adsp_compr_set_params, diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index f500692..72486bf 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2017,7 +2017,7 @@ static struct regmap *wm5102_get_regmap(struct device *dev) }, }; -static struct snd_compr_ops wm5102_compr_ops = { +static const struct snd_compr_ops wm5102_compr_ops = { .open = wm5102_open, .free = wm_adsp_compr_free, .set_params = wm_adsp_compr_set_params, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index d6fae139..858a24f 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2372,7 +2372,7 @@ static struct regmap *wm5110_get_regmap(struct device *dev) }, }; -static struct snd_compr_ops wm5110_compr_ops = { +static const struct snd_compr_ops wm5110_compr_ops = { .open = wm5110_open, .free = wm_adsp_compr_free, .set_params = wm_adsp_compr_set_params, -- 1.9.1

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[alsa-devel] [PATCH 0/3] ASoC: soc-core: tidyup snd_soc_unregister_component()
by Kuninori Morimoto 16 Aug '17

16 Aug '17

Hi Mark Mainly these are for platform replace to component. 1), 2) are cosmetic patch. 3) is a littile bit aggressive patch which unregister all component by 1 call. Previous my patch-set tried to add new remove component function, but if we can unregister all component by 1 call, it is no longer needed. Kuninori Morimoto (3): 1) ASoC: soc-core: rename "cmpnt" to "component" 2) ASoC: soc-core: remove duplicate mutex_unlock from snd_soc_unregister_component() 3) ASoC: soc-core: snd_soc_unregister_component() unregister all component include/sound/soc.h | 4 ++-- sound/soc/soc-core.c | 60 +++++++++++++++++++++++++++++++--------------------- 2 files changed, 38 insertions(+), 26 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: compress: Set reasonable compress id string
by Donglin Peng 16 Aug '17

16 Aug '17

For dynamic compress rtd, the codec_dai may not have any actual meaning(like snd-soc-dummy-dai), so compress id can just use the value of dai_link->stream_name. But for others, its codec_dai has actual meaning, so compress id can include codec_dai name. Signed-off-by: pengdonglin <pengdonglin137(a)163.com> --- sound/soc/soc-compress.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 206f36bf43e8..b6a071461b10 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -737,9 +737,6 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) } /* check client and interface hw capabilities */ - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); - if (codec_dai->driver->playback.channels_min) playback = 1; if (codec_dai->driver->capture.channels_min) @@ -758,7 +755,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) return -EINVAL; } - if(playback) + if (playback) direction = SND_COMPRESS_PLAYBACK; else direction = SND_COMPRESS_CAPTURE; @@ -797,19 +794,18 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) else if (rtd->dai_link->dpcm_capture) be_pcm->streams[SNDRV_PCM_STRE AM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); - } else + } else { + snprintf(new_name, sizeof(new_name), "%s %s-%d", + rtd->dai_link->stream_name, codec_dai->name, num); + memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); + } /* Add copy callback for not memory mapped DSPs */ if (platform->driver->compr_ops && platform->driver->compr_ops->c opy) compr->ops->copy = soc_compr_copy; mutex_init(&compr->lock); - - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, - rtd->codec_dai->name, num); - ret = snd_compress_new(rtd->card->snd_card, num, direction, new_name, compr); if (ret < 0) { -- 2.11.0

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[alsa-devel] [PATCH] ASoC: Intel: kbl: Enabling ASRC for RT5663 codec on kabylake platform
by Naveen M 16 Aug '17

16 Aug '17

From: Harsha Priya N <harshapriya.n(a)intel.com> Patch fixes cracking noise in rt5663 headphones for kbl platform by calling rt5663_sel_asrc_clk_src() for RT5663_AD_STEREO_FILTER to set ASRC. The ASRC function is for asynchronous MCLK and LRCLK. For RT5663 ASRC should be enabled to support pcm format with 100fs. ASRC function will track i2s clock and generate corresponding system clock for codec. Calling this func helps select clock source for both RT5663_AD_STEREO_FILTER and RT5663_DA_STEREO_FILTER filters which fixes the crackling sound. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Shruthi Sudhakar <shruthi.sudhakar(a)intel.com> Signed-off-by: Naveen M <naveen.m(a)intel.com> diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index f9ba977..29b50e4 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -312,11 +312,13 @@ static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5663_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); /* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */ - rt5663_sel_asrc_clk_src(codec_dai->codec, RT5663_DA_STEREO_FILTER, 1); + rt5663_sel_asrc_clk_src(codec_dai->codec, + RT5663_DA_STEREO_FILTER | RT5663_AD_STEREO_FILTER, + RT5663_CLK_SEL_I2S1_ASRC); + ret = snd_soc_dai_set_sysclk(codec_dai, + RT5663_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); if (ret < 0) dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); -- 1.9.1

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