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[PATCH AUTOSEL 6.9 01/20] ALSA: hda: cs35l56: Fix lifecycle of codec pointer
by Sasha Levin 01 Jul '24

01 Jul '24

From: Simon Trimmer <simont(a)opensource.cirrus.com> [ Upstream commit d339131bf02d4ed918415574082caf5e8af6e664 ] The codec should be cleared when the amp driver is unbound and when resuming it should be tested to prevent loading firmware into the device and ALSA in a partially configured system state. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20240531112716.25323-1-simont@opensource.cirrus.c… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/cs35l56_hda.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 11b0570ff56d4..0923e2589f5f7 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -735,6 +735,8 @@ static void cs35l56_hda_unbind(struct device *dev, struct device *master, void * if (comps[cs35l56->index].dev == dev) memset(&comps[cs35l56->index], 0, sizeof(*comps)); + cs35l56->codec = NULL; + dev_dbg(cs35l56->base.dev, "Unbound\n"); } @@ -840,6 +842,9 @@ static int cs35l56_hda_system_resume(struct device *dev) cs35l56->suspended = false; + if (!cs35l56->codec) + return 0; + ret = cs35l56_is_fw_reload_needed(&cs35l56->base); dev_dbg(cs35l56->base.dev, "fw_reload_needed: %d\n", ret); if (ret > 0) { -- 2.43.0

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[PATCH 0/7] qrb4210-rb2: add HDMI audio playback support
by Alexey Klimov 29 Jun '24

29 Jun '24

First time I am doing anything like this. This series adds mising pieces here and there to allow addition of sensible sound card in qrb4210-rb2.dts and enable HDMI audio playback there. It is planned in future to add support for other outputs and capture -- analog audio support. The series depends on Srini's sm4250 lpi pinctrl driver: c2e5a25e8d88 (pinctrl: qcom: Introduce SM4250 LPI pinctrl driver, 2024-06-22) 2ffa7a354662 (dt-bindings: pinctrl: qcom: Add SM4250 pinctrl, 2024-06-22) As far as I understood it was already accepted and will go through pinctrl tree. Alexey Klimov (7): ASoC: dt-bindings: qcom,sm8250: add qrb4210-rb2-sndcard ASoC: qcom: sm8250: add qrb4210-rb2-sndcard compatible string ASoC: qcom: sm8250: add handling of secondary MI2S clock arm64: dts: qcom: sm6115: add apr, its services and simple sound node arm64: dts: qcom: sm6115: add LPASS LPI pin controller arm64: dts: qcom: sm6115: add description of lpi_i2s2 pins arm64: dts: qcom: qrb4210-rb2: add HDMI audio playback support .../bindings/sound/qcom,sm8250.yaml | 1 + arch/arm64/boot/dts/qcom/qrb4210-rb2.dts | 73 ++++++++++++ arch/arm64/boot/dts/qcom/sm6115.dtsi | 112 ++++++++++++++++++ sound/soc/qcom/sm8250.c | 19 +++ 4 files changed, 205 insertions(+) -- 2.45.2

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[PATCH] dt-bindings: iio: stm32: dfsdm: fix dtbs warnings on dfsdm audio port
by Olivier Moysan 28 Jun '24

28 Jun '24

Fix warnings on DFSDM dtbs check Unevaluated properties are not allowed ('dfsdm-dai' was unexpected) 'port' does not match any of the regexes: 'pinctrl-[0-9]+' Fixes: 11183ac07a74 ("dt-bindings: stm32: convert dfsdm to json-schema") Signed-off-by: Olivier Moysan <olivier.moysan(a)foss.st.com> --- .../devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml b/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml index c1b1324fa132..2722edab1d9a 100644 --- a/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml +++ b/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml @@ -246,6 +246,10 @@ patternProperties: From common IIO binding. Used to pipe external sigma delta modulator or internal ADC output to DFSDM channel. + port: + $ref: /schemas/sound/audio-graph-port.yaml# + unevaluatedProperties: false + required: - compatible - "#sound-dai-cells" -- 2.25.1

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[PATCH] ASoC: meson: tdm: add sample rate support up to 768kHz
by Jerome Brunet 28 Jun '24

28 Jun '24

Add support for 705.6kHz and 768kHz sample rates Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-fifo.c | 2 +- sound/soc/meson/axg-frddr.c | 4 ++-- sound/soc/meson/axg-tdm.h | 2 +- sound/soc/meson/axg-toddr.c | 4 ++-- 4 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 59abe0b3c59f..7e6090af720b 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -32,7 +32,7 @@ static const struct snd_pcm_hardware axg_fifo_hw = { SNDRV_PCM_INFO_NO_PERIOD_WAKEUP), .formats = AXG_FIFO_FORMATS, .rate_min = 5512, - .rate_max = 384000, + .rate_max = 768000, .channels_min = 1, .channels_max = AXG_FIFO_CH_MAX, .period_bytes_min = AXG_FIFO_BURST, diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c index e97d43ae7fd2..e70c8c34c7db 100644 --- a/sound/soc/meson/axg-frddr.c +++ b/sound/soc/meson/axg-frddr.c @@ -112,7 +112,7 @@ static struct snd_soc_dai_driver axg_frddr_dai_drv = { .channels_max = AXG_FIFO_CH_MAX, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 5515, - .rate_max = 384000, + .rate_max = 768000, .formats = AXG_FIFO_FORMATS, }, .ops = &axg_frddr_ops, @@ -189,7 +189,7 @@ static struct snd_soc_dai_driver g12a_frddr_dai_drv = { .channels_max = AXG_FIFO_CH_MAX, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 5515, - .rate_max = 384000, + .rate_max = 768000, .formats = AXG_FIFO_FORMATS, }, .ops = &g12a_frddr_ops, diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h index daaca10fec9e..1a17f546ce6e 100644 --- a/sound/soc/meson/axg-tdm.h +++ b/sound/soc/meson/axg-tdm.h @@ -16,7 +16,7 @@ #define AXG_TDM_NUM_LANES 4 #define AXG_TDM_CHANNEL_MAX 128 #define AXG_TDM_RATES (SNDRV_PCM_RATE_5512 | \ - SNDRV_PCM_RATE_8000_384000) + SNDRV_PCM_RATE_8000_768000) #define AXG_TDM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_LE | \ diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c index e03a6e21c1c6..03512da4092b 100644 --- a/sound/soc/meson/axg-toddr.c +++ b/sound/soc/meson/axg-toddr.c @@ -131,7 +131,7 @@ static struct snd_soc_dai_driver axg_toddr_dai_drv = { .channels_max = AXG_FIFO_CH_MAX, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 5515, - .rate_max = 384000, + .rate_max = 768000, .formats = AXG_FIFO_FORMATS, }, .ops = &axg_toddr_ops, @@ -228,7 +228,7 @@ static struct snd_soc_dai_driver g12a_toddr_dai_drv = { .channels_max = AXG_FIFO_CH_MAX, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 5515, - .rate_max = 384000, + .rate_max = 768000, .formats = AXG_FIFO_FORMATS, }, .ops = &g12a_toddr_ops, -- 2.43.0

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[PATCH] ASoC: soc-utils: allow sample rate up to 768kHz for the dummy dai
by Jerome Brunet 28 Jun '24

28 Jun '24

The dummy DAI should allow any (reasonable) rates possible. Make the rate continuous for dummy and set range from 5512Hz to 768kHz The change is mostly cosmetic as dummy is skipped when setting the hwparams. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/soc-utils.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 11ba89c6b83c..303823dc45d7 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -144,7 +144,6 @@ static const struct snd_soc_component_driver dummy_codec = { .endianness = 1, }; -#define STUB_RATES SNDRV_PCM_RATE_8000_384000 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_U8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ @@ -198,14 +197,18 @@ static struct snd_soc_dai_driver dummy_dai = { .stream_name = "Playback", .channels_min = 1, .channels_max = 384, - .rates = STUB_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 768000, .formats = STUB_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 384, - .rates = STUB_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 768000, .formats = STUB_FORMATS, }, .ops = &dummy_dai_ops, -- 2.43.0

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[PATCH 0/4] ASoC: codecs: lpass-rx-macro: Few code cleanups
by Krzysztof Kozlowski 28 Jun '24

28 Jun '24

Hi, Improve a bit the Qualcomm LPASS RX macro driver and align similar parts of code with LPASS WSA macro driver for consistency. No external dependencies. Best regards, Krzysztof --- Krzysztof Kozlowski (4): ASoC: codecs: lpass-rx-macro: Simplify with devm allocations ASoC: codecs: lpass-rx-macro: Keep static regmap_config as const ASoC: dapm: Use unsigned for number of widgets in snd_soc_dapm_new_controls() ASoC: codecs: lpass-rx-macro: Use unsigned for number of widgets include/sound/soc-dapm.h | 2 +- sound/soc/codecs/lpass-rx-macro.c | 37 ++++++++++++++++++++++++------------- sound/soc/soc-dapm.c | 2 +- 3 files changed, 26 insertions(+), 15 deletions(-) --- base-commit: 47ec7f026f5db6a0fe26c6215554d043d04368eb change-id: 20240627-b4-qcom-audio-lpass-codec-cleanups-27175f1d069f Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

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Re: [PATCH] ASoC: fsl_xcvr: Improve suspend/resume flow in fsl_xcvr_trigger()
by Peter Jennings 28 Jun '24

28 Jun '24

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[PATCH v2 0/2] Add master clock handling for nau8824
by Maxim Kochetkov 28 Jun '24

28 Jun '24

nau8824 has external MCLK pin. So add enable/disable external clock management. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- Changes in v2: - move the devm_clk_get() to the nau8824_read_device() --- Maxim Kochetkov (2): ASoC: codecs: nau8824: Add master clock handling ASoC: dt-bindings: nau8824: Add master clock handling .../bindings/sound/nuvoton,nau8824.yaml | 8 +++++++ sound/soc/codecs/nau8824.c | 21 +++++++++++++++++-- sound/soc/codecs/nau8824.h | 1 + 3 files changed, 28 insertions(+), 2 deletions(-) -- 2.45.2

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[PATCH 0/6] ASoC: codecs: wsa88xx: add support for static port mapping.
by srinivas.kandagatla＠linaro.org 27 Jun '24

27 Jun '24

Existing way of allocating soundwire master ports on Qualcommm platforms is dynamic, and in linear order starting from 1 to MAX_PORTS. This will work as long as soundwire device ports are 1:1 mapped linearly. However on most Qcom SoCs like SM8550, SM8650, x1e80100, these are NOT mapped in that order. The result of this is that only one speaker among the pair of speakers is always silent, With recent changes for WSA codec to support codec versions and along with these patches we are able to get all speakers working on these SoCs. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Manikantan R (1): ASoC: dt-bindings: wsa883x: Document port mapping property Srinivas Kandagatla (5): ASoC: codecs: wsa883x: parse port-mapping information ASoC: dt-bindings: wsa8840: Document port mapping property ASoC: codecs: wsa884x: parse port-mapping information arm64: dts: x1e80100-crd: fix wsa soundwire port mapping arm64: dts: x1e80100-qcp: fix wsa soundwire port mapping Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml | 8 ++++++++ Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml | 8 ++++++++ arch/arm64/boot/dts/qcom/x1e80100-crd.dts | 4 ++++ arch/arm64/boot/dts/qcom/x1e80100-qcp.dts | 2 ++ sound/soc/codecs/wsa883x.c | 8 ++++++++ sound/soc/codecs/wsa884x.c | 8 ++++++++ 6 files changed, 38 insertions(+) --- base-commit: 9935be184a55dd84fc3275094f2df095491f6ea1 change-id: 20240626-port-map-ef50c3304d4a Best regards, -- Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>

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[PATCH 0/3] ASoC: imx-audmix: Split capture device to be a new device
by Shengjiu Wang 27 Jun '24

27 Jun '24

The transmitter and receiver part of the SAI interface need to be configured with different master/slave mode, especially to work with the audiomix module. The SAI1 TX is in master mode, but SAI1 RX is in slave mode. So add another two DAIs for TX and RX separately in fsl_sai driver. There will be three devices for audiomix sound card, hw:x,0 is the playback device for one SAI, hw:x,1 is the playback device for another SAI, hw:x,2 is the capture device for audmix output. Shengjiu Wang (3): ASoC: fsl_sai: Add separate DAI for transmitter and receiver ASoC: fsl_audmix: Split playback and capture stream to different DAI ASoC: imx-audmix: Split capture device for audmix sound/soc/fsl/fsl_audmix.c | 16 ++--- sound/soc/fsl/fsl_sai.c | 141 +++++++++++++++++++++++++++---------- sound/soc/fsl/fsl_sai.h | 4 +- sound/soc/fsl/imx-audmix.c | 79 ++++++++++++--------- 4 files changed, 155 insertions(+), 85 deletions(-) -- 2.34.1

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