July 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH v2 0/3] ASoC: fsl-asoc-card: Support hp and mic detection
by Shengjiu Wang 17 Jul '20

17 Jul '20

Support hp and mic detection. Add a parameter for asoc_simple_init_jack. Shengjiu Wang (3): ASoC: simple-card-utils: Support configure pin_name for asoc_simple_init_jack ASoC: bindings: fsl-asoc-card: Support hp-det-gpio and mic-det-gpio ASoC: fsl-asoc-card: Support Headphone and Microphone Jack detection changes in v2: - Add more comments in third commit - Add Acked-by Nicolin. .../bindings/sound/fsl-asoc-card.txt | 3 + include/sound/simple_card_utils.h | 6 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl-asoc-card.c | 77 ++++++++++++++++++- sound/soc/generic/simple-card-utils.c | 7 +- 5 files changed, 86 insertions(+), 8 deletions(-) -- 2.27.0

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[PATCH v2] ASoC: soc-component: Add missed return for calling soc_component_ret
by Shengjiu Wang 17 Jul '20

17 Jul '20

Add missed return for calling soc_component_ret, otherwise the return value is wrong. Fixes: e2329eeba45f ("ASoC: soc-component: add soc_component_err()") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - add missed return in snd_soc_pcm_component_sync_stop and - snd_soc_pcm_component_new sound/soc/soc-component.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index af9909c5492f..9565a0dd7cb6 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -647,7 +647,7 @@ int snd_soc_pcm_component_sync_stop(struct snd_pcm_substream *substream) ret = component->driver->sync_stop(component, substream); if (ret < 0) - soc_component_ret(component, ret); + return soc_component_ret(component, ret); } } @@ -705,7 +705,7 @@ int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream, /* FIXME. it returns 1st mmap now */ for_each_rtd_components(rtd, i, component) if (component->driver->mmap) - soc_component_ret( + return soc_component_ret( component, component->driver->mmap(component, substream, vma)); @@ -723,7 +723,7 @@ int snd_soc_pcm_component_new(struct snd_soc_pcm_runtime *rtd) if (component->driver->pcm_construct) { ret = component->driver->pcm_construct(component, rtd); if (ret < 0) - soc_component_ret(component, ret); + return soc_component_ret(component, ret); } } -- 2.27.0

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[PATCH v3] ASoC: dt-bindings: renesas, fsi: use patternProperties for FSI-A/B
by Kuninori Morimoto 17 Jul '20

17 Jul '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> FSI has FSI-A and FSI-B, and has fsia-xxx/fsib-xxx properties. This patch uses patternProperties, and reduce verbose settings. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Link: https://lore.kernel.org/r/87o8osxlow.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87pn9v39eo.wl-kuninori.morimoto.gx@renesas.com --- v2 -> v3 - add Reviewd-by - add Link .../bindings/sound/renesas,fsi.yaml | 19 ++++--------------- 1 file changed, 4 insertions(+), 15 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml index 8a4406be387a..0dd3f7361399 100644 --- a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml @@ -43,30 +43,19 @@ properties: '#sound-dai-cells': const: 1 - fsia,spdif-connection: +patternProperties: + "^fsi(a|b),spdif-connection$": $ref: /schemas/types.yaml#/definitions/flag description: FSI is connected by S/PDIF - fsia,stream-mode-support: + "^fsi(a|b),stream-mode-support$": $ref: /schemas/types.yaml#/definitions/flag description: FSI supports 16bit stream mode - fsia,use-internal-clock: + "^fsi(a|b),use-internal-clock$": $ref: /schemas/types.yaml#/definitions/flag description: FSI uses internal clock when master mode - fsib,spdif-connection: - $ref: /schemas/types.yaml#/definitions/flag - description: same as fsia - - fsib,stream-mode-support: - $ref: /schemas/types.yaml#/definitions/flag - description: same as fsia - - fsib,use-internal-clock: - $ref: /schemas/types.yaml#/definitions/flag - description: same as fsia - required: - compatible - reg -- 2.25.1

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[PATCH] ASoC: rt5682: Report the button event in the headset type only
by Oder Chiou 17 Jul '20

17 Jul '20

The irq work will be manipulated by resume function, and it will report the wrong jack type while the jack type is headphone in the button event. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5682.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index b634e582ef7e..d7302f968a75 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1082,7 +1082,8 @@ void rt5682_jack_detect_handler(struct work_struct *work) /* jack was out, report jack type */ rt5682->jack_type = rt5682_headset_detect(rt5682->component, 1); - } else { + } else if ((rt5682->jack_type & SND_JACK_HEADSET) == + SND_JACK_HEADSET) { /* jack is already in, report button event */ rt5682->jack_type = SND_JACK_HEADSET; btn_type = rt5682_button_detect(rt5682->component); -- 2.27.0

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[PATCH][resend] ASoC: dt-bindings: renesas, fsi: use patternProperties for FSI-A/B
by Kuninori Morimoto 17 Jul '20

17 Jul '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> FSI has FSI-A and FSI-B, and has fsia-xxx/fsib-xxx properties. This patch uses patternProperties, and reduce verbose settings. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- - 2weeks passed, but nothing happen .../bindings/sound/renesas,fsi.yaml | 19 ++++--------------- 1 file changed, 4 insertions(+), 15 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml index 8a4406be387a..0dd3f7361399 100644 --- a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml @@ -43,30 +43,19 @@ properties: '#sound-dai-cells': const: 1 - fsia,spdif-connection: +patternProperties: + "^fsi(a|b),spdif-connection$": $ref: /schemas/types.yaml#/definitions/flag description: FSI is connected by S/PDIF - fsia,stream-mode-support: + "^fsi(a|b),stream-mode-support$": $ref: /schemas/types.yaml#/definitions/flag description: FSI supports 16bit stream mode - fsia,use-internal-clock: + "^fsi(a|b),use-internal-clock$": $ref: /schemas/types.yaml#/definitions/flag description: FSI uses internal clock when master mode - fsib,spdif-connection: - $ref: /schemas/types.yaml#/definitions/flag - description: same as fsia - - fsib,stream-mode-support: - $ref: /schemas/types.yaml#/definitions/flag - description: same as fsia - - fsib,use-internal-clock: - $ref: /schemas/types.yaml#/definitions/flag - description: same as fsia - required: - compatible - reg -- 2.25.1

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[PATCH 0/2] ALSA: hda: export snd_hda_codec_cleanup()
by Kai Vehmanen 17 Jul '20

17 Jul '20

Hi, this small series is preparation for a set of bugfix ASoC patches addressing a memleak at module unload for the HDA codec wrapper. Instead of duplicating HDA code in ASoC tree, I chose to export more functionality from hda_codec.c so it can be (re)used in ASoC's hdac_hda.c. Full series: https://github.com/thesofproject/linux/pull/2252 Takashi and Mark, feedback is welcome on how to best handle this kind of series where I have dependent patches both in sound/pci/hda and in ASoC. For this series, I'm sending the patches separately and when/if first set is merged by Takashi, I'll route the ASoC patches via our usually SOF set to Mark. Kai Vehmanen (2): ALSA: hda: export snd_hda_codec_cleanup_for_unbind() ALSA: hda: fix snd_hda_codec_cleanup() documentation include/sound/hda_codec.h | 2 ++ sound/pci/hda/hda_codec.c | 3 ++- 2 files changed, 4 insertions(+), 1 deletion(-) -- 2.27.0

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[PATCH v2 1/1] ASoC: fsl: fsl-asoc-card: Trivial: Fix misspelling of 'exists'
by Lee Jones 17 Jul '20

17 Jul '20

Cc: Timur Tabi <timur(a)kernel.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: linuxppc-dev(a)lists.ozlabs.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/fsl/fsl-asoc-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index faac6ce9a82cb..dbacdd25dfe76 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -92,7 +92,7 @@ struct fsl_asoc_card_priv { }; /* - * This dapm route map exits for DPCM link only. + * This dapm route map exists for DPCM link only. * The other routes shall go through Device Tree. * * Note: keep all ASRC routes in the second half -- 2.25.1

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[PATCH] ASoC: soc-jack: calling snd_soc_jack_report causes a null pointer access
by mnlife qiao 17 Jul '20

17 Jul '20

When snd_soc_card_jack_new is not called or the call fails, calling this function causes a null pointer access Signed-off-by: mnlife <mnlife(a)foxmail.com> --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0f1820f36b4d..51b799ee98b7 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -44,7 +44,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) unsigned int sync = 0; int enable; - if (!jack) + if (!jack || !jack->jack) return; trace_snd_soc_jack_report(jack, mask, status); -- 2.17.1

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[PATCH v6] ASoC: Intel: kbl_rt5663_rt5514_max98927: Fix kabylake_ssp_fixup function
by Harsha Priya 16 Jul '20

16 Jul '20

From: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> kabylake_ssp_fixup function uses snd_soc_dpcm to identify the codecs DAIs. The hw parameters are changed based on the codec DAI, the stream is intended for. The earlier approach to get snd_soc_dpcm was using container_of() macro on snd_pcm_hw_params. The structures have been modified over time and snd_soc_dpcm does not have snd_pcm_hw_params as a reference but as a copy. This causes the current driver to crash when used. This patch changes the way snd_soc_dpcm is extracted. The snd_soc_pcm_runtime holds 2 dpcm instances (one for playback and one for capture). The 2 codecs on this SSP are dmic and speakers. One is for capture and one is for playback respectively. Based on the direction of the stream, the snd_soc_dpcm is extracted from the snd_soc_pcm_runtime structure. Tested for all use cases of the driver. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> Tested-by: Lukasz Majczak <lma(a)semihalf.com> --- v1 -> v2: - Extract dmic from SSP0 as every BE should have own fixup function. v2 -> v3: - Restore naming in the dapm route table to not confuse with other drivers - Fixed indentations v3 -> v4: - Updated code and commit description according to solution proposed by Harsha v4 -> v5: - Cosmetic Changes v5 -> v6: - Dmic regression seen with v4 fixed - Using available routines for obtaining dpcm information --- --- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 26 ++++++++++++------- 1 file changed, 17 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 584e4f9cedc2..b261b1c466a8 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -379,22 +379,30 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + struct snd_soc_dpcm *dpcm, *rtd_dpcm; + + /* + * This macro will be called for playback stream + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) + rtd_dpcm = dpcm; + /* + * This macro will be called for capture stream + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) + rtd_dpcm = dpcm; /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; chan->min = chan->max = 2; snd_mask_none(fmt); snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { + } else if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio DMIC cap")) { if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) chan->min = chan->max = 2; @@ -405,7 +413,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) + if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; -- 2.17.1

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[PATCH v3] ASoC: Intel: kbl_rt5663_rt5514_max98927: Split be_hw_params_fixup function
by Lukasz Majczak 16 Jul '20

16 Jul '20

Split be_hw_params_fixup function for different codecs as current common function, leads to crash while trying to get snd_soc_dpcm with container_of() macro in kabylake_ssp_fixup(). The crash call path looks as below: soc_pcm_hw_params() snd_soc_dai_hw_params(codec_dai, substream, &codec_params); rtd->dai_link->be_hw_params_fixup(rtd, params) kabylake_ssp_fixup() In this case, codec_params is just a copy of an internal structure and is not embedded into struct snd_soc_dpcm thus we cannot use container_of() on it. v1 -> v2: - Extract dmic from SSP0 as every BE should have own fixup function. v2 -> v3: - Restore naming in the dapm route table to not confuse with other drivers - Fixed indentations Signed-off-by: Lukasz Majczak <lma(a)semihalf.com> --- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 130 +++++++++++------- 1 file changed, 84 insertions(+), 46 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 1b1f8d7a4ea3..b9596cf3a6c4 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -328,42 +328,51 @@ static const struct snd_soc_ops kabylake_rt5663_fe_ops = { .startup = kbl_fe_startup, }; -static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) +static void kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params, snd_pcm_format_t pcm_fmt) { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *chan = hw_param_interval(params, + struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { - rate->min = rate->max = 48000; - chan->min = chan->max = 2; - snd_mask_none(fmt); - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { - if (params_channels(params) == 2 || - DMIC_CH(dmic_constraints) == 2) - chan->min = chan->max = 2; - else - chan->min = chan->max = 4; - } - /* - * The speaker on the SSP0 supports S16_LE and not S24_LE. - * thus changing the mask here - */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); + + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + snd_mask_none(fmt); + snd_mask_set_format(fmt, pcm_fmt); +} + +static int kabylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + kabylake_ssp_fixup(rtd, params, SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int kabylake_ssp1_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + + kabylake_ssp_fixup(rtd, params, SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; return 0; } @@ -400,20 +409,6 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, int ret = 0, j; for_each_rtd_codec_dais(rtd, j, codec_dai) { - if (!strcmp(codec_dai->component->name, RT5514_DEV_NAME)) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0, 8, 16); - if (ret < 0) { - dev_err(rtd->dev, "set TDM slot err:%d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5514_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(rtd->dev, "set sysclk err: %d\n", ret); - return ret; - } - } if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); if (ret < 0) { @@ -433,10 +428,36 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, return ret; } +static int kabylake_dmic01_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret = 0; + + ret = snd_soc_dai_set_tdm_slot(rtd->codec_dai, 0xF, 0, 8, 16); + if (ret < 0) { + dev_err(rtd->dev, "set TDM slot err:%d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(rtd->codec_dai, + RT5514_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "set sysclk err: %d\n", ret); + return ret; + } + + return ret; +} + static struct snd_soc_ops kabylake_ssp0_ops = { .hw_params = kabylake_ssp0_hw_params, }; +static struct snd_soc_ops kabylake_dmic01_ops = { + .hw_params = kabylake_dmic01_hw_params, +}; + static const unsigned int channels_dmic[] = { 4, }; @@ -507,14 +528,19 @@ SND_SOC_DAILINK_DEF(ssp0_pin, SND_SOC_DAILINK_DEF(ssp0_codec, DAILINK_COMP_ARRAY( /* Left */ COMP_CODEC(MAXIM_DEV0_NAME, KBL_MAXIM_CODEC_DAI), - /* Right */COMP_CODEC(MAXIM_DEV1_NAME, KBL_MAXIM_CODEC_DAI), - /* dmic */ COMP_CODEC(RT5514_DEV_NAME, KBL_REALTEK_DMIC_CODEC_DAI))); + /* Right */COMP_CODEC(MAXIM_DEV1_NAME, KBL_MAXIM_CODEC_DAI))); SND_SOC_DAILINK_DEF(ssp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); SND_SOC_DAILINK_DEF(ssp1_codec, DAILINK_COMP_ARRAY(COMP_CODEC(RT5663_DEV_NAME, KBL_REALTEK_CODEC_DAI))); +SND_SOC_DAILINK_DEF(dmic01_pin, + DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); +SND_SOC_DAILINK_DEF(dmic01_codec, + DAILINK_COMP_ARRAY( + COMP_CODEC(RT5514_DEV_NAME, KBL_REALTEK_DMIC_CODEC_DAI))); + SND_SOC_DAILINK_DEF(idisp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); SND_SOC_DAILINK_DEF(idisp1_codec, @@ -618,9 +644,8 @@ static struct snd_soc_dai_link kabylake_dais[] = { SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, + .be_hw_params_fixup = kabylake_ssp0_fixup, .dpcm_playback = 1, - .dpcm_capture = 1, .ops = &kabylake_ssp0_ops, SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), }, @@ -632,12 +657,25 @@ static struct snd_soc_dai_link kabylake_dais[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, + .be_hw_params_fixup = kabylake_ssp1_fixup, .ops = &kabylake_rt5663_ops, .dpcm_playback = 1, .dpcm_capture = 1, SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), }, + { + .name = "dmic01", + .id = 2, + .no_pcm = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_dmic_fixup, + .dpcm_capture = 1, + .ops = &kabylake_dmic01_ops, + SND_SOC_DAILINK_REG(dmic01_pin, dmic01_codec, platform), + }, { .name = "iDisp1", .id = 3, base-commit: a4f6fc98cd2fa1774bcaeb248c67156ef9402a56 -- 2.25.1

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