September 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH v4 00/15] ASoC/qcom/arm64: Qualcomm ADSP DTS and binding fixes
by Krzysztof Kozlowski 23 Sep '22

23 Sep '22

Hi, Dependencies/merging ==================== 1. The DTS patches are independent. 2. The binding patches should come together, because of context changes. Could be one of: Qualcomm SoC, ASoC or DT tree. Changes since v3 ================ 1. Patch 9-10: re-order, so first apr.yaml is corrected and then we convert to DT schema. This makes patchset fully bisectable in expense of changing the same lines twice. 2. Patch 11: New patch. Changes since v2 ================ 1. Patch 9: rename and extend commit msg. 2. Add Rb tags. Changes since v1 ================ 1. Patch 9: New patch. 2. Patch 10: Correct also sound/qcom,q6apm-dai.yaml (Rob). 2. Patch 13: New patch. 3. Add Rb/Tb tags. Best regards, Krzysztof Krzysztof Kozlowski (15): arm64: dts: qcom: sdm630: align APR services node names with dtschema arm64: dts: qcom: sdm845: align APR services node names with dtschema arm64: dts: qcom: sm8250: align APR services node names with dtschema arm64: dts: qcom: msm8996: fix APR services nodes arm64: dts: qcom: sdm845: align dai node names with dtschema arm64: dts: qcom: msm8996: align dai node names with dtschema arm64: dts: qcom: qrb5165-rb5: align dai node names with dtschema arm64: dts: qcom: sm8250: use generic name for LPASS clock controller dt-bindings: soc: qcom: apr: correct service children ASoC: dt-bindings: qcom,q6asm: convert to dtschema ASoC: dt-bindings: qcom,q6adm: convert to dtschema ASoC: dt-bindings: qcom,q6dsp-lpass-ports: cleanup example ASoC: dt-bindings: qcom,q6dsp-lpass-clocks: cleanup example ASoC: dt-bindings: qcom,q6apm-dai: adjust indentation in example dt-bindings: soc: qcom: apr: add missing properties .../bindings/soc/qcom/qcom,apr.yaml | 112 ++++++++++++++++-- .../bindings/sound/qcom,q6adm-routing.yaml | 52 ++++++++ .../devicetree/bindings/sound/qcom,q6adm.txt | 39 ------ .../bindings/sound/qcom,q6apm-dai.yaml | 21 ++-- .../bindings/sound/qcom,q6asm-dais.yaml | 112 ++++++++++++++++++ .../devicetree/bindings/sound/qcom,q6asm.txt | 70 ----------- .../sound/qcom,q6dsp-lpass-clocks.yaml | 36 +++--- .../sound/qcom,q6dsp-lpass-ports.yaml | 64 +++++----- arch/arm64/boot/dts/qcom/msm8996.dtsi | 10 +- arch/arm64/boot/dts/qcom/qrb5165-rb5.dts | 4 +- arch/arm64/boot/dts/qcom/sdm630.dtsi | 8 +- arch/arm64/boot/dts/qcom/sdm845-db845c.dts | 2 +- .../boot/dts/qcom/sdm845-xiaomi-beryllium.dts | 2 +- .../boot/dts/qcom/sdm845-xiaomi-polaris.dts | 4 +- arch/arm64/boot/dts/qcom/sdm845.dtsi | 8 +- arch/arm64/boot/dts/qcom/sm8250.dtsi | 10 +- 16 files changed, 346 insertions(+), 208 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6adm-routing.yaml delete mode 100644 Documentation/devicetree/bindings/sound/qcom,q6adm.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm-dais.yaml delete mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm.txt -- 2.34.1

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[PATCH] Revert "ASoC: fsl_audmix: make clock and output src write only"
by Shengjiu Wang 23 Sep '22

23 Sep '22

This reverts commit 944c517b8c838832a166f1c89afbf8724f4a6b49. There is error after making clock and output src write only $amixer -c imxaudmix cset numid=1 1 amixer: Cannot read the given element from control sysdefault:3 Which is worse than before, so let's revert the change. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_audmix.c | 16 ++++------------ 1 file changed, 4 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index 43857b7a81c9..672148dd4b23 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -199,18 +199,10 @@ static int fsl_audmix_put_out_src(struct snd_kcontrol *kcontrol, static const struct snd_kcontrol_new fsl_audmix_snd_controls[] = { /* FSL_AUDMIX_CTR controls */ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Mixing Clock Source", - .info = snd_soc_info_enum_double, - .access = SNDRV_CTL_ELEM_ACCESS_WRITE, - .put = fsl_audmix_put_mix_clk_src, - .private_value = (unsigned long)&fsl_audmix_enum[0] }, - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Output Source", - .info = snd_soc_info_enum_double, - .access = SNDRV_CTL_ELEM_ACCESS_WRITE, - .put = fsl_audmix_put_out_src, - .private_value = (unsigned long)&fsl_audmix_enum[1] }, + SOC_ENUM_EXT("Mixing Clock Source", fsl_audmix_enum[0], + snd_soc_get_enum_double, fsl_audmix_put_mix_clk_src), + SOC_ENUM_EXT("Output Source", fsl_audmix_enum[1], + snd_soc_get_enum_double, fsl_audmix_put_out_src), SOC_ENUM("Output Width", fsl_audmix_enum[2]), SOC_ENUM("Frame Rate Diff Error", fsl_audmix_enum[3]), SOC_ENUM("Clock Freq Diff Error", fsl_audmix_enum[4]), -- 2.34.1

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[PATCH -next] ASoC: Intel: sof_da7219_mx98360a: Access num_codecs through dai_link
by Nathan Chancellor 23 Sep '22

23 Sep '22

After commit 3989ade2d1e7 ("ASoC: soc.h: remove num_cpus/codecs"), the following build error occurs: sound/soc/intel/boards/sof_da7219_max98373.c:198:27: error: no member named 'num_codecs' in 'struct snd_soc_pcm_runtime' for (j = 0; j < runtime->num_codecs; j++) { ~~~~~~~ ^ 1 error generated. This conversion was missed by the aforementioned change. Do it now to fix the build error. Fixes: 3989ade2d1e7 ("ASoC: soc.h: remove num_cpus/codecs") Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/intel/boards/sof_da7219_max98373.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c index 34cf849a8344..e048e789e633 100644 --- a/sound/soc/intel/boards/sof_da7219_max98373.c +++ b/sound/soc/intel/boards/sof_da7219_max98373.c @@ -195,7 +195,7 @@ static int ssp1_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *runtime = asoc_substream_to_rtd(substream); int ret, j; - for (j = 0; j < runtime->num_codecs; j++) { + for (j = 0; j < runtime->dai_link->num_codecs; j++) { struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, j); if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { base-commit: a921986f445ad611b441c8ee7749dc6dfc770481 -- 2.37.3

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[PATCH 1/2] ASoC: wcd9335: fix order of Slimbus unprepare/disable
by Krzysztof Kozlowski 23 Sep '22

23 Sep '22

Slimbus streams are first prepared and then enabled, so the cleanup path should reverse it. The unprepare sets stream->num_ports to 0 and frees the stream->ports. Calling disable after unprepare was not really effective (channels was not deactivated) and could lead to further issues due to making transfers on unprepared stream. Fixes: 20aedafdf492 ("ASoC: wcd9335: add support to wcd9335 codec") Cc: <stable(a)vger.kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/wcd9335.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 06c6adbe5920..d2548fdf9ae5 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -1972,8 +1972,8 @@ static int wcd9335_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - slim_stream_unprepare(dai_data->sruntime); slim_stream_disable(dai_data->sruntime); + slim_stream_unprepare(dai_data->sruntime); break; default: break; -- 2.34.1

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[PATCH 0/6] ASoC: mediatek: Allow separate handling of headphone and headset mic jack
by Nícolas F. R. A. Prado 23 Sep '22

23 Sep '22

This series allows the headphone and headset mic jack status to be handled separately by userspace on MT8192, MT8195 and MT8186. Changes based on commit d0508b4f1604 ("ASoC: rk3399_gru_sound: Add DAPM pins, kcontrols for jack detection"). Found while searching for an alternative for JackSwitch [1]. [1] https://lore.kernel.org/all/b98a8a77-7652-1995-27ba-eb7b6d30202a@gmail.com/ Nícolas F. R. A. Prado (6): ASoC: mediatek: mt8192-mt6359: Expose individual headset jack pins ASoC: mediatek: mt8195: Expose individual headset jack pins ASoC: mediatek: mt8186-da7219: Add headset widgets with switches ASoC: mediatek: mt8186-da7219: Expose individual headset jack pins ASoC: mediatek: mt8186-rt5682: Add headset widgets with switches ASoC: mediatek: mt8186-rt5682: Expose individual headset jack pins .../mt8186/mt8186-mt6366-da7219-max98357.c | 25 +++++++++++++++++-- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 25 +++++++++++++++++-- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 17 +++++++++++-- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 17 +++++++++++-- 4 files changed, 76 insertions(+), 8 deletions(-) -- 2.37.3

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[PATCH 00/10] ASoC: SOF: Intel: MTL: Fixes for suspend/resume
by Ranjani Sridharan 23 Sep '22

23 Sep '22

This series includes fixes for suspend/resume and module loading/unloading for the MTL platform. Fred Oh (4): ASoC: SOF: Intel: introduce new op to handle dsp power down ASoC: SOF: Intel: define and set power_down_dsp op for HDA platforms ASoC: SOF: Intel: mtl: define and set power_down_dsp op ASoC: SOF: Intel: use power_down_dsp op in hda_dsp_remove Ranjani Sridharan (5): ASoC: SOF: Intel: Add a new op for disabling interrupts ASoC: SOF: Intel: define and set the disable_interrupts op for cavs platforms ASoC: SOF: Intel: MTL: define and set the disable_interrupts op ASoC: SOF: Intel: HDA: use IPC version-specific ops ASoC: SOF: Intel: MTL: reuse the common ops for PM Yong Zhi (1): ASoC: SOF: Intel: MTL: remove the unnecessary snd_sof_dsp_read() sound/soc/sof/intel/apl.c | 2 + sound/soc/sof/intel/cnl.c | 4 + sound/soc/sof/intel/hda-dsp.c | 21 ++-- sound/soc/sof/intel/hda.c | 14 ++- sound/soc/sof/intel/hda.h | 2 + sound/soc/sof/intel/icl.c | 2 + sound/soc/sof/intel/mtl.c | 193 ++++++---------------------------- sound/soc/sof/intel/shim.h | 2 + sound/soc/sof/intel/skl.c | 2 + sound/soc/sof/intel/tgl.c | 8 ++ 10 files changed, 81 insertions(+), 169 deletions(-) -- 2.25.1

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[PATCH] ASoC: fsl: Remove unused inline function imx_pcm_dma_params_init_data()
by Gaosheng Cui 23 Sep '22

23 Sep '22

The imx_pcm_dma_params_init_data() are no longer used since commit c31da0b196f9 ("ASoC: imx-ssi: Remove unused driver"), and the function is used to initialize some members of "struct imx_dma_data", it's more readable to assign the value directly, imx_pcm_dma_params_init_data is useless, so remove it. Signed-off-by: Gaosheng Cui <cuigaosheng1(a)huawei.com> --- sound/soc/fsl/imx-pcm.h | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index 06b25f4b26b6..ac5f57c3cc55 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -18,15 +18,6 @@ #define IMX_DEFAULT_DMABUF_SIZE (64 * 1024) -static inline void -imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data, - int dma, enum sdma_peripheral_type peripheral_type) -{ - dma_data->dma_request = dma; - dma_data->priority = DMA_PRIO_HIGH; - dma_data->peripheral_type = peripheral_type; -} - struct imx_pcm_fiq_params { int irq; void __iomem *base; -- 2.25.1

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[PATCH] ASoC: mediatek: mt8195-mt6359: Use snd_soc_pm_ops instead of custom ops
by AngeloGioacchino Del Regno 23 Sep '22

23 Sep '22

It is possible to use the standard snd_soc_pm_ops for this card: remove the custom mt8195_mt6359_pm_ops. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> --- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359.c b/sound/soc/mediatek/mt8195/mt8195-mt6359.c index 961e769602d6..23bdde6acd1c 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359.c @@ -1532,16 +1532,11 @@ static const struct of_device_id mt8195_mt6359_dt_match[] = { {}, }; -static const struct dev_pm_ops mt8195_mt6359_pm_ops = { - .poweroff = snd_soc_poweroff, - .restore = snd_soc_resume, -}; - static struct platform_driver mt8195_mt6359_driver = { .driver = { .name = "mt8195_mt6359", .of_match_table = mt8195_mt6359_dt_match, - .pm = &mt8195_mt6359_pm_ops, + .pm = &snd_soc_pm_ops, }, .probe = mt8195_mt6359_dev_probe, }; -- 2.37.2

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[PATCH 00/12] ASoC: cs42l42: Add Soundwire support
by Richard Fitzgerald 23 Sep '22

23 Sep '22

The CS42L42 has a Soundwire interface for control and audio. This chain of patches adds support for this. Patches #1 .. #10 split out various changes to the existing code that are needed for adding Soundwire. These are mostly around clocking and supporting the separate probe and enumeration stages in Soundwire. Patches #11 and #12 actually add the Soundwire handling. Richard Fitzgerald (12): ASoC: cs42l42: Add SOFT_RESET_REBOOT register ASoC: cs42l42: Add bitclock frequency argument to cs42l42_pll_config() ASoC: cs42l42: Ensure MCLKint is a multiple of the sample rate ASoC: cs42l42: Separate ASP config from PLL config ASoC: cs42l42: Use cs42l42->dev instead of &i2c_client->dev ASoC: cs42l42: Split probe() and remove() into stages ASoC: cs42l42: Split cs42l42_resume into two functions ASoC: cs42l42: Pass component and dai defs into common probe ASoC: cs42l42: Split I2C identity into separate module ASoC: cs42l42: Export some functions for Soundwire ASoC: cs42l42: Add Soundwire support ASoC: cs42l42: Add support for Soundwire interrupts include/sound/cs42l42.h | 5 + sound/soc/codecs/Kconfig | 14 +- sound/soc/codecs/Makefile | 6 +- sound/soc/codecs/cs42l42-i2c.c | 107 +++++ sound/soc/codecs/cs42l42-sdw.c | 689 +++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l42.c | 359 +++++++++-------- sound/soc/codecs/cs42l42.h | 30 +- 7 files changed, 1049 insertions(+), 161 deletions(-) create mode 100644 sound/soc/codecs/cs42l42-i2c.c create mode 100644 sound/soc/codecs/cs42l42-sdw.c -- 2.30.2

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[Allen & Heath Qu-Pac] regression since 0.3.53+ / no audio inputs unless manual selection of config profile ("Pro") in pulse audio control
by GitHub issues - edited 23 Sep '22

23 Sep '22

alsa-project/alsa-lib issue #272 was edited from ianhundere:  - PipeWire version (`pipewire --version`): - `0.3.58` but also experienced with `0.3.53`, `0.3.55` and `0.3.56`. did not experience this prior to `0.3.53`. - Distribution and distribution version (`PRETTY_NAME` from `/etc/os-release`): - `Pop!_OS 22.04 LTS` - Desktop Environment: - `GNOME 42.3.1` - Kernel version (`uname -r`): - `5.19.0-76051900-generic` ## Description of Problem: have to manually set audio profile in pulse audio control. ## How Reproducible: reset interface / no inputs in daw (bitwig 4.3.8) ### Steps to Reproduce: 1. reset interface (allen heath qupac) 1. go to pulse audio control 1. unselect pro / reselect pro errors: `journalctl --no-hostname --user -u pipewire ` ``` Jul 30 14:39:43 systemd[1618]: Started PipeWire Multimedia Service. Jul 30 14:39:43 pipewire[1635]: spa.alsa: hw:acp: Rate doesn't match (requested 44100, got 48000) Jul 30 14:39:43 pipewire[1635]: spa.alsa: hw:acp: Rate doesn't match (requested 44100, got 48000) Jul 30 14:39:43 pipewire[1635]: spa.alsa: hw:acp: Rate doesn't match (requested 44100, got 48000) Jul 30 14:40:42 pipewire[1635]: spa.alsa: hw:3,0: snd_pcm_start: Broken pipe Jul 30 14:40:42 pipewire[1635]: spa.audioadapter: 0x56374b3cf218: can't send command 2: Broken pipe Jul 30 14:40:42 pipewire[1635]: pw.node: (alsa_input.usb-Allen_Heath_Ltd_QU-PAC-01.pro-input-0-110) suspended -> error (Start error: Broken> ``` then i unselect / reselect "Pro" via pulseaudio volume control: ``` Jul 30 22:54:35 pipewire[1632]: pw.link: 0x55618e69d5d0: one of the nodes is in error out:error in:suspended Jul 30 22:54:35 pipewire[1632]: pw.link: 0x55618e69e200: one of the nodes is in error out:error in:suspended Jul 30 22:54:35 pipewire[1632]: spa.alsa: hw:3,0: snd_pcm_start: Broken pipe Jul 30 22:54:35 pipewire[1632]: spa.audioadapter: 0x55618e007e58: can't send command 2: Broken pipe Jul 30 22:54:35 pipewire[1632]: spa.alsa: hw:3,0: snd_pcm_start: Broken pipe Jul 30 22:54:35 pipewire[1632]: spa.audioadapter: 0x55618e007e58: can't send command 2: Broken pipe Jul 30 22:54:35 pipewire[1632]: spa.alsa: hw:3,0: snd_pcm_start: Broken pipe Jul 30 22:54:35 pipewire[1632]: spa.audioadapter: 0x55618e007e58: can't send command 2: Broken pipe Jul 30 22:54:36 pipewire[1632]: spa.alsa: hw:3,0: snd_pcm_start: Broken pipe Jul 30 22:54:36 pipewire[1632]: spa.audioadapter: 0x55618e007e58: can't send command 2: Broken pipe Jul 30 22:54:36 pipewire[1632]: spa.alsa: hw:acp: Rate doesn't match (requested 44100, got 48000) Jul 30 22:54:36 pipewire[1632]: spa.alsa: hw:3,0: snd_pcm_start: Broken pipe Jul 30 22:54:36 pipewire[1632]: spa.audioadapter: 0x55618e007e58: can't send command 2: Broken pipe Jul 30 22:54:36 pipewire[1632]: spa.alsa: hw:3,0: snd_pcm_start: Broken pipe Jul 30 22:54:36 pipewire[1632]: spa.audioadapter: 0x55618e007e58: can't send command 2: Broken pipe Jul 30 22:54:46 pipewire[1632]: spa.alsa: hw:3,0: follower delay:520 target:256 thr:256, resync ``` ### Actual Results: no audio from external inputs ### Expected Results: you shouldn't need to reselect a profile # Additional Info (as attachments): - more info: https://github.com/pop-os/pipewire/issues/16, https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/2567 Issue URL : https://github.com/alsa-project/alsa-lib/issues/272 Repository URL: https://github.com/alsa-project/alsa-lib

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