January 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [GIT PULL] ASoC fixes for v5.0
by Mark Brown 18 Jan '19

18 Jan '19

The following changes since commit a7a850dba82498a1e050d8d153cae67ce0edb3b2: Merge branch 'asoc-4.21' into asoc-next (2018-12-18 12:23:59 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.0-rc2 for you to fetch changes up to 4cb79ef9c6c4413427cd70afbb1f3bc01e9b7abf: ASoC: amd: Fix potential NULL pointer dereference (2019-01-15 19:06:23 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.0 Quite a big batch of fixes here. There's a couple of things going on, the main one is that we found some issues with not deferring probe when we should, causing us to skip some driver initialization. The fixes for this then in turn exposed some issues with how we were searching for components which had previously gone unnoticed due to the original issue. There's also been the normal driver specific stuff and there's been what looks like several batches of automated scanning for issues which have generated quite a large set of smaller fixes for potential crashes and missed error handling. ---------------------------------------------------------------- Should probably have pointed out that the DaVinci runtime PM fix here is a big diff but mostly code motion, it fixes problems where the device would stop working after runtime PM. Ajit Pandey (1): ASoC: soc-core: defer card probe until all component is added to list Curtis Malainey (1): ASoC: soc-core: fix init platform memory handling Dan Carpenter (6): ASoC: dma-sh7760: cleanup a debug printk ASoC: pcm512x: Fix a double unlock in pcm512x_digital_mute() ASoC: qdsp6: q6asm-dai: Off by one in of_q6asm_parse_dai_data() ASoC: qdsp6: q6asm-dai: Fix a NULL vs IS_ERR() bug ASoC: qdsp6: q6asm-dai: Fix a small memory leak ALSA: compress: prevent potential divide by zero bugs Geert Uytterhoeven (1): ASoC: xlnx: Grammar s/the the/the/ Gustavo A. R. Silva (2): ASoC: rt5514-spi: Fix potential NULL pointer dereference ASoC: amd: Fix potential NULL pointer dereference Kangjie Lu (1): ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages Mac Chiang (1): ASoC: Intel: Boards: move the codec PLL configuration to _init Mark Brown (1): ASoC: core: Make snd_soc_find_component() more robust Maruthi Srinivas Bayyavarapu (1): ASoC: xlnx: change license header format style Matthias Reichl (1): ASoC: core: Don't defer probe on optional, NULL components Peter Ujfalusi (1): ASoC: ti: davinci-mcasp: Move context save/restore to runtime_pm callbacks Rander Wang (1): ASoC: Intel: Haswell/Broadwell: fix setting for .dynamic field Rohit kumar (2): ASoC: sdm845: set jack only for a specific backend ASoC: soc-core: Hold client_mutex around soc_init_dai_link() Shuming Fan (2): ASoC: rt5682: Fix recording no sound issue ASoC: rt5682: Fix PLL source register definitions Silvio Cesare (2): ASoC: dapm: change snprintf to scnprintf for possible overflow ASoC: imx-audmux: change snprintf to scnprintf for possible overflow Sinan Kaya (1): ASoC: Intel: atom: Make PCI dependency explicit Yizhuo (1): ASoC: Variable "val" in function rt274_i2c_probe() could be uninitialized b-ak (1): ASoC: tlv320aic32x4: Kernel OOPS while entering DAPM standby mode include/sound/soc.h | 6 ++ sound/core/compress_offload.c | 3 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 6 +- sound/soc/codecs/pcm512x.c | 11 +-- sound/soc/codecs/rt274.c | 5 +- sound/soc/codecs/rt5514-spi.c | 2 + sound/soc/codecs/rt5682.c | 1 + sound/soc/codecs/rt5682.h | 24 ++--- sound/soc/codecs/tlv320aic32x4.c | 4 + sound/soc/fsl/imx-audmux.c | 24 ++--- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 8 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 45 ++------- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 17 +++- sound/soc/qcom/sdm845.c | 31 ++++-- sound/soc/sh/dma-sh7760.c | 2 +- sound/soc/soc-core.c | 34 +++++-- sound/soc/soc-dapm.c | 10 +- sound/soc/ti/davinci-mcasp.c | 136 ++++++++++++-------------- sound/soc/xilinx/Kconfig | 2 +- sound/soc/xilinx/xlnx_i2s.c | 15 ++- 23 files changed, 208 insertions(+), 184 deletions(-)

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[alsa-devel] [PATCH v2 1/5] ASoC: imx-sgtl5000: put of nodes if finding codec fails
by Stefan Agner 18 Jan '19

18 Jan '19

Make sure to properly put the of node in case finding the codec fails. Fixes: 81e8e4926167 ("ASoC: fsl: add sgtl5000 clock support for imx-sgtl5000") Signed-off-by: Stefan Agner <stefan(a)agner.ch> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Changes in v2: - Reordered patches, make sure this is the first in the patchset -- Stefan sound/soc/fsl/imx-sgtl5000.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index c29200cf755a..594bde3b0ded 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -111,7 +111,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EPROBE_DEFER; + ret = -EPROBE_DEFER; + goto fail; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); -- 2.20.1

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[alsa-devel] [PATCH] alsamixer: define _GNU_SOURCE to get exp10
by makepost＠firemail.cc 18 Jan '19

18 Jan '19

Fixes Master, Headphones and Speaker stuck at -8 percentage after building with Clang 7.0.1 and getting warned about implicit declaration of exp10, which is behind _GNU_SOURCE as a non-standard feature. Thanks Takashi Iwai for the CFLAGS suggestion on the mailing list. GCC build is not affected, so not adding a compiler check to the configure script. uClibc-ng has got exp10 since 1.0.12, so the fallback macro is no longer needed. However, alsa-utils relies on gettext so might need further patches to actually work on uClibc systems. Signed-off-by: makepost <makepost(a)firemail.cc> diff --git a/alsamixer/volume_mapping.c b/alsamixer/volume_mapping.c index 94bd0fe..48cfbe2 100644 --- a/alsamixer/volume_mapping.c +++ b/alsamixer/volume_mapping.c @@ -36,11 +36,6 @@ #include <stdbool.h> #include "volume_mapping.h" -#ifdef __UCLIBC__ -/* 10^x = 10^(log e^x) = (e^x)^log10 = e^(x * log 10) */ -#define exp10(x) (exp((x) * log(10))) -#endif /* __UCLIBC__ */ - #define MAX_LINEAR_DB_SCALE 24 static inline bool use_linear_dB_scale(long dBmin, long dBmax) diff --git a/configure.ac b/configure.ac index 7938996..0d54942 100644 --- a/configure.ac +++ b/configure.ac @@ -350,6 +350,8 @@ if test x$alsamixer = xtrue; then AC_MSG_RESULT([$CURSESLIB]) + + CFLAGS="$CFLAGS -D_GNU_SOURCE" AC_SUBST(CURSESINC) -- 2.20.1

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[alsa-devel] Timing issues between ALSA and i915 drivers
by Pierre-Louis Bossart 18 Jan '19

18 Jan '19

Hi, I could use some feedback on HDMI audio issues exposed during the 4.21 merge window. By accident (misleading documentation) we ended up enabling the Skylake driver instead of the HDaudio legacy, and broke audio on a number of Skylake and ApolloLake devices where the HDMI/iDISP codec was not detected (bit 2 not set in the codec_mask). Linus' Dell XPS13 9350 was the first to be impacted of course... After debugging a bit, this issue can be resolved by either a) compiling both SOUND and DRM as built-ins (y instead of m) b) moving the calls snd_hdac_i915_init() to the probe function instead of the worker queue (code at https://github.com/plbossart/sound/commits/fix/skl-hdmi) Both solutions point to timing issues. During internal reviews I was alerted to the fact that the suggested fix essentially reverts patch ab1b732d53c18 ('ASoC: Intel: Skylake: Move i915 registration to worker thread') which was introduced to solve DRM lockup issues. In other words, we are at a point where we either have DRM lockups or can't detect the HDMI audio codec, none of which are too good. I don't have the background for the DRM lockup stuff, nor do I understand why snd_hdac_i915_init has to be called from a thread context. Is this really a requirement? I also don't get what sort of timing issues would come from an initialization. What happens on the i915 side and is there some sort of mandatory delay between the completion of the i915_init and issuing a snd_hdac_chip_readw(bus, STATESTS) to get the codec masks on the HDaudio links? Thanks for any pointers or comments. -Pierre

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[alsa-devel] [PATCH v2] ASoC: soc.h: add explanation of legacy/modern style of dai_link
by Kuninori Morimoto 18 Jan '19

18 Jan '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current ALSA SoC is assuming 1 CPU 1 Platform (= DMA) style system. Because of this background, it is directly using xxx_name / xxx_of_node / xxx_dai_name on dai_link. Let's call it as legacy style here. More complex style system like multi CPU multi Platform (= DMA) will coming. To supporting it, we can use snd_soc_dai_link_component on dai_link. Let's call it as modern style here. But current ALSA SoC can't support it so far. Thus, we need to have multi CPU / multi Codec / multi Platform style in the future on ALSA SoC. Currently we already have multi Codec support. Platform is starting to use modern style on dai_link, but still style only. Multi Platform is not yet implemented. And we still don't have multi CPU support on ALSA SoC, and not have modern style either. Currently, if driver is using legacy style Codec/Platform, it will be converted to modern style on soc-core. This means, we are using glue code for legacy vs modern style so far on ALSA SoC. We can fully switch to modern style on all drivers if ALSA SoC supported modern style for CPU, and then, legacy style code will be removed from ALSA SoC. Untile then, we need to keep both legacy/modern style and its glue code. This patch adds such future plan and background on soc.h Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - fixup "Driver" typo include/sound/soc.h | 36 ++++++++++++++++++++++++++++++++++++ sound/soc/soc-core.c | 20 ++++++++++++++++++-- 2 files changed, 54 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index e665f11..c31b6d1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -891,6 +891,18 @@ struct snd_soc_dai_link { /* config - must be set by machine driver */ const char *name; /* Codec name */ const char *stream_name; /* Stream name */ + + /* + * cpu_name + * cpu_of_node + * cpu_dai_name + * + * These are legacy style, and will be replaced to + * modern style (= snd_soc_dai_link_component) in the future, + * but, not yet supported so far. + * If modern style was supported for CPU, all driver will switch + * to use it, and, legacy style code will be removed from ALSA SoC. + */ /* * You MAY specify the link's CPU-side device, either by device name, * or by DT/OF node, but not both. If this information is omitted, @@ -906,6 +918,19 @@ struct snd_soc_dai_link { * only, which only works well when that device exposes a single DAI. */ const char *cpu_dai_name; + + /* + * codec_name + * codec_of_node + * codec_dai_name + * + * These are legacy style, it will be converted to modern style + * (= snd_soc_dai_link_component) automatically in soc-core + * if driver is using legacy style. + * Driver shouldn't use both legacy and modern style in the same time. + * If modern style was supported for CPU, all driver will switch + * to use it, and, legacy style code will be removed from ALSA SoC. + */ /* * You MUST specify the link's codec, either by device name, or by * DT/OF node, but not both. @@ -919,6 +944,17 @@ struct snd_soc_dai_link { unsigned int num_codecs; /* + * platform_name + * platform_of_node + * + * These are legacy style, it will be converted to modern style + * (= snd_soc_dai_link_component) automatically in soc-core + * if driver is using legacy style. + * Driver shouldn't use both legacy and modern style in the same time. + * If modern style was supported for CPU, all driver will switch + * to use it, and, legacy style code will be removed from ALSA SoC. + */ + /* * You MAY specify the link's platform/PCM/DMA driver, either by * device name, or by DT/OF node, but not both. Some forms of link * do not need a platform. diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c0aaff..bf405d6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1038,9 +1038,14 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link_component *platform = dai_link->platform; /* - * FIXME + * REMOVE ME * - * this function should be removed in the future + * This is glue code for Legacy vs Modern dai_link. + * This function will be removed if all derivers are switched to + * modern style dai_link. + * Driver shouldn't use both legacy and modern style in the same time. + * see + * soc.h :: struct snd_soc_dai_link */ /* convert Legacy platform link */ if (!platform || dai_link->legacy_platform) { @@ -1068,6 +1073,17 @@ static int snd_soc_init_platform(struct snd_soc_card *card, static int snd_soc_init_multicodec(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { + /* + * REMOVE ME + * + * This is glue code for Legacy vs Modern dai_link. + * This function will be removed if all derivers are switched to + * modern style dai_link. + * Driver shouldn't use both legacy and modern style in the same time. + * see + * soc.h :: struct snd_soc_dai_link + */ + /* Legacy codec/codec_dai link is a single entry in multicodec */ if (dai_link->codec_name || dai_link->codec_of_node || dai_link->codec_dai_name) { -- 2.7.4

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[alsa-devel] [PATCH] ASoC: soc.h: add explanation of legacy/modern style of dai_link
by Kuninori Morimoto 18 Jan '19

18 Jan '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current ALSA SoC is assuming 1 CPU 1 Platform (= DMA) style system. Because of this background, it is directly using xxx_name / xxx_of_node / xxx_dai_name on dai_link. Let's call it as legacy style here. More complex style system like multi CPU multi Platform (= DMA) will coming. To supporting it, we can use snd_soc_dai_link_component on dai_link. Let's call it as modern style here. But current ALSA SoC can't support it so far. Thus, we need to have multi CPU / multi Codec / multi Platform style in the future on ALSA SoC. Currently we already have multi Codec support. Platform is starting to use modern style on dai_link, but still style only. Multi Platform is not yet implemented. And we still don't have multi CPU support on ALSA SoC, and not have modern style either. Currently, if driver is using legacy style Codec/Platform, it will be converted to modern style on soc-core. This means, we are using glue code for legacy vs modern style so far on ALSA SoC. We can fully switch to modern style on all drivers if ALSA SoC supported modern style for CPU, and then, legacy style code will be removed from ALSA SoC. Untile then, we need to keep both legacy/modern style and its glue code. This patch adds such future plan and background on soc.h Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc.h | 36 ++++++++++++++++++++++++++++++++++++ sound/soc/soc-core.c | 20 ++++++++++++++++++-- 2 files changed, 54 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index e665f11..d17e368 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -891,6 +891,18 @@ struct snd_soc_dai_link { /* config - must be set by machine driver */ const char *name; /* Codec name */ const char *stream_name; /* Stream name */ + + /* + * cpu_name + * cpu_of_node + * cpu_dai_name + * + * These are legacy style, and will be replaced to + * modern style (= snd_soc_dai_link_component) in the future, + * but, not yet supported so far. + * If modern style was supported for CPU, all driver will switch + * to use it, and, legacy style code will be removed from ALSA SoC. + */ /* * You MAY specify the link's CPU-side device, either by device name, * or by DT/OF node, but not both. If this information is omitted, @@ -906,6 +918,19 @@ struct snd_soc_dai_link { * only, which only works well when that device exposes a single DAI. */ const char *cpu_dai_name; + + /* + * codec_name + * codec_of_node + * codec_dai_name + * + * These are legacy style, it will be converted to modern style + * (= snd_soc_dai_link_component) automatically in soc-core + * if driver is using legacy style. + * Dirver shouldn't use both legacy and modern style in the same time. + * If modern style was supported for CPU, all driver will switch + * to use it, and, legacy style code will be removed from ALSA SoC. + */ /* * You MUST specify the link's codec, either by device name, or by * DT/OF node, but not both. @@ -919,6 +944,17 @@ struct snd_soc_dai_link { unsigned int num_codecs; /* + * platform_name + * platform_of_node + * + * These are legacy style, it will be converted to modern style + * (= snd_soc_dai_link_component) automatically in soc-core + * if driver is using legacy style. + * Dirver shouldn't use both legacy and modern style in the same time. + * If modern style was supported for CPU, all driver will switch + * to use it, and, legacy style code will be removed from ALSA SoC. + */ + /* * You MAY specify the link's platform/PCM/DMA driver, either by * device name, or by DT/OF node, but not both. Some forms of link * do not need a platform. diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c0aaff..f01a223 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1038,9 +1038,14 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link_component *platform = dai_link->platform; /* - * FIXME + * REMOVE ME * - * this function should be removed in the future + * This is glue code for Legacy vs Modern dai_link. + * This function will be removed if all derivers are switched to + * modern style dai_link. + * Dirver shouldn't use both legacy and modern style in the same time. + * see + * soc.h :: struct snd_soc_dai_link */ /* convert Legacy platform link */ if (!platform || dai_link->legacy_platform) { @@ -1068,6 +1073,17 @@ static int snd_soc_init_platform(struct snd_soc_card *card, static int snd_soc_init_multicodec(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { + /* + * REMOVE ME + * + * This is glue code for Legacy vs Modern dai_link. + * This function will be removed if all derivers are switched to + * modern style dai_link. + * Dirver shouldn't use both legacy and modern style in the same time. + * see + * soc.h :: struct snd_soc_dai_link + */ + /* Legacy codec/codec_dai link is a single entry in multicodec */ if (dai_link->codec_name || dai_link->codec_of_node || dai_link->codec_dai_name) { -- 2.7.4

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[alsa-devel] [PATCH 1/5] ASoC: fsl_spdif: don't print EPROBE_DEFER as error
by Stefan Agner 17 Jan '19

17 Jan '19

Probe deferral is to be expected during normal operation, so avoid printing an error when it is encountered. Signed-off-by: Stefan Agner <stefan(a)agner.ch> --- sound/soc/fsl/fsl_spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 740b90df44bb..a26686e7281c 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1320,7 +1320,7 @@ static int fsl_spdif_probe(struct platform_device *pdev) } ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "imx_pcm_dma_init failed: %d\n", ret); return ret; -- 2.20.1

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[alsa-devel] [PATCH] ASoC: rt5682: add default pdata for i2s mode
by Bard liao 17 Jan '19

17 Jan '19

Add a default pdata which can fit most HW design. So we don't need to add a lot of DMI checking in this driver. Signed-off-by: Bard liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/codecs/rt5682.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 89c43b26c379..bd45d0343913 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -43,6 +43,12 @@ static const char *rt5682_supply_names[RT5682_NUM_SUPPLIES] = { "VBAT", }; +static const struct rt5682_platform_data i2s_default_platform_data = { + .dmic1_data_pin = RT5682_DMIC1_DATA_GPIO2, + .dmic1_clk_pin = RT5682_DMIC1_CLK_GPIO3, + .jd_src = RT5682_JD1, +}; + struct rt5682_priv { struct snd_soc_component *component; struct rt5682_platform_data pdata; @@ -2534,6 +2540,8 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5682); + rt5682->pdata = i2s_default_platform_data; + if (pdata) rt5682->pdata = *pdata; else -- 2.17.1

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[alsa-devel] [PATCH] ASoC: qcom: Kconfig: select max98927 for sdm845
by Cheng-Yi Chiang 16 Jan '19

16 Jan '19

Select SND_SOC_MAX98927 for SND_SOC_SDM845. Acked-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- This is part of the patch https://patchwork.kernel.org/patch/10696507/ RT5663 part was added in the patch e9dc919043c5 ("ASoC: sdm845: add rt5663 codec select") This patch adds max98927. Thanks! sound/soc/qcom/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 804ae0d93058..7948e993adba 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -102,6 +102,7 @@ config SND_SOC_SDM845 select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON select SND_SOC_RT5663 + select SND_SOC_MAX98927 help To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems. -- 2.20.1.97.g81188d93c3-goog

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[alsa-devel] [PATCH] ASoC: sdm845: Set DAI format for dmic codec
by Cheng-Yi Chiang 16 Jan '19

16 Jan '19

Set codec DAI format for dmic codec in startup. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/qcom/sdm845.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 6f66a58e23ca..882f52ed8231 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -248,12 +248,14 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream) break; case SECONDARY_MI2S_TX: + codec_dai_fmt |= SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_I2S; if (++(data->sec_mi2s_clk_count) == 1) { snd_soc_dai_set_sysclk(cpu_dai, Q6AFE_LPASS_CLK_ID_SEC_MI2S_IBIT, MI2S_BCLK_RATE, SNDRV_PCM_STREAM_CAPTURE); } snd_soc_dai_set_fmt(cpu_dai, fmt); + snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt); break; case QUATERNARY_TDM_RX_0: -- 2.20.1.97.g81188d93c3-goog

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