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[PATCH -next] ASoC: codecs: wcd938x: constify static struct snd_soc_dai_ops
by Pu Lehui 15 Jun '21

15 Jun '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Signed-off-by: Pu Lehui <pulehui(a)huawei.com> --- sound/soc/codecs/wcd938x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 2cf6145b7219..8ec9357fd855 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3425,7 +3425,7 @@ static int wcd938x_codec_set_sdw_stream(struct snd_soc_dai *dai, } -static struct snd_soc_dai_ops wcd938x_sdw_dai_ops = { +static const struct snd_soc_dai_ops wcd938x_sdw_dai_ops = { .hw_params = wcd938x_codec_hw_params, .hw_free = wcd938x_codec_free, .set_sdw_stream = wcd938x_codec_set_sdw_stream, -- 2.17.1

2 1

[PATCH 0/3] ASoC: tlv320aic32x4: Add support for TAS2505
by Claudius Heine 15 Jun '21

15 Jun '21

Hi, this is an early patchset that provides support for the TAS2505 by the tlv320aic32x4 driver. Playback with aplay and the right mixer settings seems to be working, but some mixer settings either cause distortions or popping. Any input on how to prevent that is welcome. Also with this change TAS2505 is compiled in unconditionally whenever SND_SOC_TLV320AIC32X4_I2C is enabled. Maybe an additional option for the TAS2505 should be added, but then we probably also need a setting to provide only support for the TAS2505 and not the TLV320AIC32X4. Not sure how to set that up in the best way. This is my first time working with the sound subsystem, so expect stupidity and obvious mistakes. regards, Claudius Claudius Heine (3): ASoC: tlv320aic32x4: prepare driver for different device variants ASoC: tlv320aic32x4: add support for TAS2505 ASoC: tlv320aic32x4: dt-bindings: add TAS2505 to compatible .../bindings/sound/tlv320aic32x4.txt | 1 + sound/soc/codecs/tlv320aic32x4-i2c.c | 22 ++- sound/soc/codecs/tlv320aic32x4-spi.c | 23 ++- sound/soc/codecs/tlv320aic32x4.c | 173 +++++++++++++++++- sound/soc/codecs/tlv320aic32x4.h | 10 + 5 files changed, 219 insertions(+), 10 deletions(-) base-commit: 009c9aa5be652675a06d5211e1640e02bbb1c33d -- 2.32.0

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[PATCH] ASoC: dt-bindings: fsl-sai: Add compatible string for imx8mm/8mn/8mp/8ulp
by Shengjiu Wang 15 Jun '21

15 Jun '21

Add compatible string for imx8mm/8mn/8mp/8ulp, these platforms all support SAI IP. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl-sai.txt | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 0dc83cc4a236..c71c5861d787 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -9,8 +9,10 @@ Required properties: - compatible : Compatible list, contains "fsl,vf610-sai", "fsl,imx6sx-sai", "fsl,imx6ul-sai", - "fsl,imx7ulp-sai", "fsl,imx8mq-sai" or - "fsl,imx8qm-sai". + "fsl,imx7ulp-sai", "fsl,imx8mq-sai", + "fsl,imx8qm-sai", "fsl,imx8mm-sai", + "fsl,imx8mn-sai", "fsl,imx8mp-sai", or + "fsl,imx8ulp-sai". - reg : Offset and length of the register set for the device. -- 2.27.0

3 2

[PATCH v6 0/4] Unify NVIDIA Tegra ASoC machine drivers
by Dmitry Osipenko 15 Jun '21

15 Jun '21

This series squashes all the ASoC machine drivers into a single one, this change was suggested by Jon Hunter. It also sets driver_name and components string of each card, allowing userspace alsa-lib to find UCMs at predictable path. Changelog: v6: - Fixed missed configuration of AC97 clock rate for the WM9712 codec in the unified driver. - Added new patch that removes now obsolete "utils" helpers and moves code into the unified driver. ASoC: tegra: Squash utils into common machine driver v5: - The v4 removed the customization of components string for Nexus 7, but I missed to remove the "components" hook which is unused now, it's removed in v5 for consistency. - Slightly improved naming of the common 12MHz MCLK rate function to make it more consistent with the rest of the driver functions. v4: - Moved out mclk_rate callback that is currently used only by WM8903 machine driver from the common driver. This was suggested by Jon Hunter. - Dropped patch which was setting custom components string for Nexus 7. Jaroslav Kysela wants it to be specified in a device-tree, but the components string doesn't have a firm specification for today. It's better to drop this change for now since it's optional anyways. - Fixed compilation error that was reported by kernel robot for v3. - Jaroslav Kysela merged alsa-ucm-conf PR [1] which added UCMs for Nexus 7 and Acer A500. The UCMs are fully working using a combination of updated kernel + alsa-ucm-conf master + alsa-lib master, meaning that they will work with the next releases of kernel and ALSA userspace upstream packages. - Added ack from Jaroslav Kysela to the "Specify components string for each card" patch that he gave to v3. v3: - Added components string as was suggested by Jaroslav Kysela to v2. - Renamed MCLK rate function that is used by max98090 and other codecs to make it look more generic. Added option for specifying CLK ID per device. This all was suggested by Jon Hunter to v2. v2: - Dropped use of of_device_compatible_match(), like it was suggested by Rob Herring in a review comment to v1. - Added patch that sets card's driver_name of as Tegra ASoC drivers. In a comment to v1 Jaroslav Kysela suggested that the Tegra drivers don't set the card name properly and he was right. I opened pull request with the new Tegra UCMs and updated lookup paths for older UCMs [1]. [1] https://github.com/alsa-project/alsa-ucm-conf/pull/92 Dmitry Osipenko (4): ASoC: tegra: Set driver_name=tegra for all machine drivers ASoC: tegra: Unify ASoC machine drivers ASoC: tegra: Specify components string for each card ASoC: tegra: Squash utils into common machine driver sound/soc/tegra/Kconfig | 12 + sound/soc/tegra/Makefile | 19 +- sound/soc/tegra/tegra_alc5632.c | 259 -------- sound/soc/tegra/tegra_asoc_machine.c | 854 +++++++++++++++++++++++++++ sound/soc/tegra/tegra_asoc_machine.h | 49 ++ sound/soc/tegra/tegra_asoc_utils.c | 225 ------- sound/soc/tegra/tegra_asoc_utils.h | 38 -- sound/soc/tegra/tegra_max98090.c | 276 --------- sound/soc/tegra/tegra_rt5640.c | 222 ------- sound/soc/tegra/tegra_rt5677.c | 324 ---------- sound/soc/tegra/tegra_sgtl5000.c | 211 ------- sound/soc/tegra/tegra_wm8753.c | 185 ------ sound/soc/tegra/tegra_wm8903.c | 351 +++-------- sound/soc/tegra/tegra_wm9712.c | 166 ------ sound/soc/tegra/trimslice.c | 172 ------ 15 files changed, 996 insertions(+), 2367 deletions(-) delete mode 100644 sound/soc/tegra/tegra_alc5632.c create mode 100644 sound/soc/tegra/tegra_asoc_machine.c create mode 100644 sound/soc/tegra/tegra_asoc_machine.h delete mode 100644 sound/soc/tegra/tegra_asoc_utils.c delete mode 100644 sound/soc/tegra/tegra_asoc_utils.h delete mode 100644 sound/soc/tegra/tegra_max98090.c delete mode 100644 sound/soc/tegra/tegra_rt5640.c delete mode 100644 sound/soc/tegra/tegra_rt5677.c delete mode 100644 sound/soc/tegra/tegra_sgtl5000.c delete mode 100644 sound/soc/tegra/tegra_wm8753.c delete mode 100644 sound/soc/tegra/tegra_wm9712.c delete mode 100644 sound/soc/tegra/trimslice.c -- 2.30.2

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[PATCH] ASoC: dt-bindings: fsl, spdif: Add compatible string for imx8ulp
by Shengjiu Wang 15 Jun '21

15 Jun '21

Add compatible string for imx8ulp, which supports spdif module Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,spdif.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml index 4454aca34d56..f226ec13167a 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -25,6 +25,7 @@ properties: - fsl,imx8mq-spdif - fsl,imx8mm-spdif - fsl,imx8mn-spdif + - fsl,imx8ulp-spdif reg: maxItems: 1 -- 2.27.0

3 2

Xone43C: USB communication
by Roman Muller 15 Jun '21

15 Jun '21

Hello Gerardo, hello all of you, > Is your capture complete from a software standpoint? i.e. are you > starting Serato or Traktor on the VM while doing the usbmon capture? Not tested yet, I need to get a few items for that and investigate the matter! It is complete up to loading the Win10 driver (which has been supplied by the manufacturer). Maybe I am lucky and this is already sufficient. Have a nice day, Roman

2 1

[PATCH -next] ASoC: axi-spdif: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/adi/axi-spdif.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/adi/axi-spdif.c b/sound/soc/adi/axi-spdif.c index 9b3d81c41c8c..8d4a6cb4e5c5 100644 --- a/sound/soc/adi/axi-spdif.c +++ b/sound/soc/adi/axi-spdif.c @@ -189,8 +189,7 @@ static int axi_spdif_probe(struct platform_device *pdev) platform_set_drvdata(pdev, spdif); - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap_resource(&pdev->dev, res); + base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(base)) return PTR_ERR(base); -- 2.25.1

2 1

[PATCH -next] ASoC: codecs: wcd: Remove unneeded semicolon
by Yang Li 14 Jun '21

14 Jun '21

Eliminate the following coccicheck warning: ./sound/soc/codecs/wcd-mbhc-v2.c:990:2-3: Unneeded semicolon. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/codecs/wcd-mbhc-v2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd-mbhc-v2.c b/sound/soc/codecs/wcd-mbhc-v2.c index dee9410..405128c 100644 --- a/sound/soc/codecs/wcd-mbhc-v2.c +++ b/sound/soc/codecs/wcd-mbhc-v2.c @@ -987,7 +987,7 @@ static void wcd_mbhc_adc_update_fsm_source(struct wcd_mbhc *mbhc, wcd_mbhc_write_field(mbhc, WCD_MBHC_BTN_ISRC_CTL, 0); break; - }; + } } static void wcd_mbhc_bcs_enable(struct wcd_mbhc *mbhc, int plug_type, bool enable) -- 1.8.3.1

3 2

[PATCH -next] ASoC: mchp-i2s-mcc: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/atmel/mchp-i2s-mcc.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 673bc16cb46a..8988f024a732 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -1008,8 +1008,7 @@ static int mchp_i2s_mcc_probe(struct platform_device *pdev) if (!dev) return -ENOMEM; - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap_resource(&pdev->dev, mem); + base = devm_platform_get_and_ioremap_resource(pdev, 0, &mem); if (IS_ERR(base)) return PTR_ERR(base); -- 2.25.1

2 1

[RESEND PATCH] ASoC: rt5640: Make codec selectable
by Martin Blumenstingl 14 Jun '21

14 Jun '21

The Realtek rt5640 codec driver can be used with the generic sound card drivers, so it should be selectable. For example, with the addition of #sound-dai-cells = <0> property in DT, it can be used with simple and graph card drivers. Signed-off-by: Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> --- Re-sent because my last mail did not make it to the list This will be used in arch/arm/boot/dts/meson8b-ec100.dts sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2a7b3e363069..96fc0144f9fc 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1180,7 +1180,7 @@ config SND_SOC_RT5631 depends on I2C config SND_SOC_RT5640 - tristate + tristate "Realtek RT5640/RT5639 Codec" depends on I2C config SND_SOC_RT5645 -- 2.31.1

2 1

[PATCH] ASoC: soc-pcm: fix the return value in dpcm_apply_symmetry()
by Jaroslav Kysela 14 Jun '21

14 Jun '21

In case, where the loops are not executed for a reason, the uninitialized variable 'err' is returned to the caller. Make code fully predictible and assign zero in the declaration. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Mark Brown <broonie(a)kernel.org> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8659089a87a0..46513bb97904 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1700,7 +1700,7 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, struct snd_soc_dpcm *dpcm; struct snd_soc_pcm_runtime *fe = asoc_substream_to_rtd(fe_substream); struct snd_soc_dai *fe_cpu_dai; - int err; + int err = 0; int i; /* apply symmetry for FE */ -- 2.30.2

2 1

Re: [PATCH] ASoC:fsl_easrc:Remove superfluous error message around platform_get_irq()
by Mark Brown 14 Jun '21

14 Jun '21

On Thu, 10 Jun 2021 20:50:52 +0800, Zhongjun Tan wrote: > Clean up the check for irq.dev_err is superfluous as platform_get_irq() > already prints an error.Remove curly braces to confirm to styling > requirements. Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC:fsl_easrc:Remove superfluous error message around platform_get_irq() commit: 4d5f3a096f3d9e7067c7c2e730d989668e06d552 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

1 0

[PATCH v8 0/9] ASoC: codecs: add wcd938x support
by Srinivas Kandagatla 14 Jun '21

14 Jun '21

This patchset adds support for Qualcomm WCD938X codec. Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 4 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 7 x TX diff inputs, 8 DMICs and MBHC. Eventhough this device has two SoundWire devices, only tx device has access to main codec Control/Status Registers! For codec driver to be functional it would need both tx and rx Soundwire devices to be up and this is taken care by using device component framework and device-links are used to ensure proper pm dependencies. Ex tx does not enter suspend before rx or codec is suspended. This patchset along with other SoundWire patches on the list have been tested on SM8250 MTP device. Thanks, srini Changes since v7: - Fixed errors reported by kernel test robot by moving SoundWire specific bits to sdw driver. Srinivas Kandagatla (9): ASoC: dt-bindings: wcd938x: add bindings for wcd938x ASoC: codecs: wcd-clsh: add new version support ASoC: codecs: wcd938x: add basic driver ASoC: dt-bindings: wcd938x-sdw: add bindings for wcd938x-sdw ASoC: codecs: wcd938x-sdw: add SoundWire driver ASoC: codecs: wcd938x: add basic controls ASoC: codecs: wcd938x: add playback dapm widgets ASoC: codecs: wcd938x: add capture dapm widgets ASoC: codecs: wcd938x: add audio routing .../bindings/sound/qcom,wcd938x-sdw.yaml | 70 + .../bindings/sound/qcom,wcd938x.yaml | 146 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/wcd-clsh-v2.c | 348 +- sound/soc/codecs/wcd-clsh-v2.h | 16 + sound/soc/codecs/wcd938x-sdw.c | 315 ++ sound/soc/codecs/wcd938x.c | 3743 +++++++++++++++++ sound/soc/codecs/wcd938x.h | 720 ++++ 9 files changed, 5365 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml create mode 100644 sound/soc/codecs/wcd938x-sdw.c create mode 100644 sound/soc/codecs/wcd938x.c create mode 100644 sound/soc/codecs/wcd938x.h -- 2.21.0

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[PATCH -next] ASoC: atmel-i2s: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/atmel/atmel-i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 584656cc7d3c..e8fe1a7a4d83 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -610,8 +610,7 @@ static int atmel_i2s_probe(struct platform_device *pdev) dev->caps = match->data; /* Map I/O registers. */ - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap_resource(&pdev->dev, mem); + base = devm_platform_get_and_ioremap_resource(pdev, 0, &mem); if (IS_ERR(base)) return PTR_ERR(base); -- 2.25.1

2 1

[PATCH] dt-bindings: ASoC: sgtl5000: Add audio-graph-card port
by Marek Vasut 14 Jun '21

14 Jun '21

The SGTL5000 codec can be connected via audio-graph-card, add the missing port: entry into the bindings. Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: kernel(a)dh-electronics.com Cc: alsa-devel(a)alsa-project.org To: devicetree(a)vger.kernel.org --- Documentation/devicetree/bindings/sound/sgtl5000.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.yaml b/Documentation/devicetree/bindings/sound/sgtl5000.yaml index 70b4a8831073..e762c320b574 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.yaml +++ b/Documentation/devicetree/bindings/sound/sgtl5000.yaml @@ -75,6 +75,10 @@ properties: $ref: "/schemas/types.yaml#/definitions/uint32" enum: [ 0, 1, 2, 3 ] + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + required: - compatible - reg -- 2.30.2

2 1

[PATCH v9 0/9] ASoC: codecs: add wcd938x support
by Srinivas Kandagatla 14 Jun '21

14 Jun '21

This patchset adds support for Qualcomm WCD938X codec. Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 4 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 7 x TX diff inputs, 8 DMICs and MBHC. Eventhough this device has two SoundWire devices, only tx device has access to main codec Control/Status Registers! For codec driver to be functional it would need both tx and rx Soundwire devices to be up and this is taken care by using device component framework and device-links are used to ensure proper pm dependencies. Ex tx does not enter suspend before rx or codec is suspended. This patchset along with other SoundWire patches on the list have been tested on SM8250 MTP device. Thanks, srini Changes since v8: - moved Kconfig and Makefile changes to last patch as suggested by Mark - removed array of enums and used static entries instead. Suggested by Mark - return true if put succeeds, Suggested by Mark - removed some unneeded semi-colons in switch Srinivas Kandagatla (9): ASoC: dt-bindings: wcd938x: add bindings for wcd938x ASoC: codecs: wcd-clsh: add new version support ASoC: codecs: wcd938x: add basic driver ASoC: dt-bindings: wcd938x-sdw: add bindings for wcd938x-sdw ASoC: codecs: wcd938x-sdw: add SoundWire driver ASoC: codecs: wcd938x: add basic controls ASoC: codecs: wcd938x: add playback dapm widgets ASoC: codecs: wcd938x: add capture dapm widgets ASoC: codecs: wcd938x: add audio routing and Kconfig .../bindings/sound/qcom,wcd938x-sdw.yaml | 70 + .../bindings/sound/qcom,wcd938x.yaml | 146 + sound/soc/codecs/Kconfig | 14 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/wcd-clsh-v2.c | 348 +- sound/soc/codecs/wcd-clsh-v2.h | 16 + sound/soc/codecs/wcd938x-sdw.c | 315 ++ sound/soc/codecs/wcd938x.c | 3753 +++++++++++++++++ sound/soc/codecs/wcd938x.h | 720 ++++ 9 files changed, 5376 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml create mode 100644 sound/soc/codecs/wcd938x-sdw.c create mode 100644 sound/soc/codecs/wcd938x.c create mode 100644 sound/soc/codecs/wcd938x.h -- 2.21.0

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[PATCH] ASoC: rt5645: Avoid upgrading static warnings to errors
by Mark Brown 14 Jun '21

14 Jun '21

One of the fixes reverted as part of the UMN fallout was actually fine, however rather than undoing the revert the process that handled all this stuff resulted in a patch which attempted to add extra error checks instead. Unfortunately this new change wasn't really based on a good understanding of the subsystem APIs and bypassed the usual patch flow without ensuring it was reviewed by people with subsystem knowledge and was merged as a fix rather than during the merge window. The effect of the new fix is to upgrade what were previously warnings on static data in the code to hard errors on that data. If this actually happens then it would break existing systems, if it doesn't happen then the change has no effect so this was not a safe change to apply as a fix to the release candidates. Since the new code has not been tested and doesn't in practice improve error handling revert it instead, and also drop the original revert since the original fix was fine. This takes the driver back to what it was in -rc1. Fixes: 5e70b8e22b64e ("ASoC: rt5645: add error checking to rt5645_probe function") Fixes: 1e0ce84215dbf ("Revert "ASoC: rt5645: fix a NULL pointer dereference") Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Phillip Potter <phil(a)philpotter.co.uk> --- sound/soc/codecs/rt5645.c | 49 +++++++++------------------------------ 1 file changed, 11 insertions(+), 38 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 438fa18bcb55..9408ee63cb26 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3388,44 +3388,30 @@ static int rt5645_probe(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); - int ret = 0; rt5645->component = component; switch (rt5645->codec_type) { case CODEC_TYPE_RT5645: - ret = snd_soc_dapm_new_controls(dapm, + snd_soc_dapm_new_controls(dapm, rt5645_specific_dapm_widgets, ARRAY_SIZE(rt5645_specific_dapm_widgets)); - if (ret < 0) - goto exit; - - ret = snd_soc_dapm_add_routes(dapm, + snd_soc_dapm_add_routes(dapm, rt5645_specific_dapm_routes, ARRAY_SIZE(rt5645_specific_dapm_routes)); - if (ret < 0) - goto exit; - if (rt5645->v_id < 3) { - ret = snd_soc_dapm_add_routes(dapm, + snd_soc_dapm_add_routes(dapm, rt5645_old_dapm_routes, ARRAY_SIZE(rt5645_old_dapm_routes)); - if (ret < 0) - goto exit; } break; case CODEC_TYPE_RT5650: - ret = snd_soc_dapm_new_controls(dapm, + snd_soc_dapm_new_controls(dapm, rt5650_specific_dapm_widgets, ARRAY_SIZE(rt5650_specific_dapm_widgets)); - if (ret < 0) - goto exit; - - ret = snd_soc_dapm_add_routes(dapm, + snd_soc_dapm_add_routes(dapm, rt5650_specific_dapm_routes, ARRAY_SIZE(rt5650_specific_dapm_routes)); - if (ret < 0) - goto exit; break; } @@ -3433,17 +3419,9 @@ static int rt5645_probe(struct snd_soc_component *component) /* for JD function */ if (rt5645->pdata.jd_mode) { - ret = snd_soc_dapm_force_enable_pin(dapm, "JD Power"); - if (ret < 0) - goto exit; - - ret = snd_soc_dapm_force_enable_pin(dapm, "LDO2"); - if (ret < 0) - goto exit; - - ret = snd_soc_dapm_sync(dapm); - if (ret < 0) - goto exit; + snd_soc_dapm_force_enable_pin(dapm, "JD Power"); + snd_soc_dapm_force_enable_pin(dapm, "LDO2"); + snd_soc_dapm_sync(dapm); } if (rt5645->pdata.long_name) @@ -3454,14 +3432,9 @@ static int rt5645_probe(struct snd_soc_component *component) GFP_KERNEL); if (!rt5645->eq_param) - ret = -ENOMEM; -exit: - /* - * If there was an error above, everything will be cleaned up by the - * caller if we return an error here. This will be done with a later - * call to rt5645_remove(). - */ - return ret; + return -ENOMEM; + + return 0; } static void rt5645_remove(struct snd_soc_component *component) -- 2.20.1

2 2

[PATCH -next] ASoC: bcm: cygnus_ssp: Use devm_platform_ioremap_resource_byname()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use the devm_platform_ioremap_resource_byname() helper instead of calling platform_get_resource_byname() and devm_ioremap_resource() separately. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/bcm/cygnus-ssp.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index ba03bb62ba96..fca5a3f2eec5 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -1308,7 +1308,6 @@ static int cygnus_ssp_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct device_node *child_node; - struct resource *res; struct cygnus_audio *cygaud; int err; int node_count; @@ -1320,13 +1319,11 @@ static int cygnus_ssp_probe(struct platform_device *pdev) dev_set_drvdata(dev, cygaud); - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "aud"); - cygaud->audio = devm_ioremap_resource(dev, res); + cygaud->audio = devm_platform_ioremap_resource_byname(pdev, "aud"); if (IS_ERR(cygaud->audio)) return PTR_ERR(cygaud->audio); - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "i2s_in"); - cygaud->i2s_in = devm_ioremap_resource(dev, res); + cygaud->i2s_in = devm_platform_ioremap_resource_byname(pdev, "i2s_in"); if (IS_ERR(cygaud->i2s_in)) return PTR_ERR(cygaud->i2s_in); -- 2.25.1

2 1

[PATCH] ASoC: fsl_spdif: Fix error handler with pm_runtime_enable
by Shengjiu Wang 14 Jun '21

14 Jun '21

There is error message when defer probe happens: fsl-spdif-dai 2dab0000.spdif: Unbalanced pm_runtime_enable! Fix the error handler with pm_runtime_enable and add fsl_spdif_remove() for pm_runtime_disable. Fixes: 9cb2b3796e08 ("ASoC: fsl_spdif: Add pm runtime function") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_spdif.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 2a76714eb8e6..76a7c9581333 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1441,16 +1441,29 @@ static int fsl_spdif_probe(struct platform_device *pdev) &spdif_priv->cpu_dai_drv, 1); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); - return ret; + goto err_pm_disable; } ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE); - if (ret && ret != -EPROBE_DEFER) - dev_err(&pdev->dev, "imx_pcm_dma_init failed: %d\n", ret); + if (ret) { + dev_err_probe(&pdev->dev, ret, "imx_pcm_dma_init failed\n"); + goto err_pm_disable; + } + + return ret; +err_pm_disable: + pm_runtime_disable(&pdev->dev); return ret; } +static int fsl_spdif_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + + return 0; +} + #ifdef CONFIG_PM static int fsl_spdif_runtime_suspend(struct device *dev) { @@ -1554,6 +1567,7 @@ static struct platform_driver fsl_spdif_driver = { .pm = &fsl_spdif_pm, }, .probe = fsl_spdif_probe, + .remove = fsl_spdif_remove, }; module_platform_driver(fsl_spdif_driver); -- 2.27.0

2 1

[PATCH -next] ASoC: atmel-pdmic: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/atmel/atmel-pdmic.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c index 8e1d8230b180..42117de299e7 100644 --- a/sound/soc/atmel/atmel-pdmic.c +++ b/sound/soc/atmel/atmel-pdmic.c @@ -620,8 +620,7 @@ static int atmel_pdmic_probe(struct platform_device *pdev) return ret; } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - io_base = devm_ioremap_resource(dev, res); + io_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(io_base)) return PTR_ERR(io_base); -- 2.25.1

2 1

[PATCH v3] ASoC: qcom: Add four speaker support on MI2S secondary
by Srinivasa Rao Mandadapu 14 Jun '21

14 Jun '21

Add four speaker support on MI2S secondary block by using I2S SD1 line on gpio52 pin, and add channel map control support in the lpass-cpu audio driver. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes Since V2: -- Added empty entry at the end of channel maps array. Changes Since V1: -- removed set_channel_map/get_channel_map implementation as default kcontrols added in pcm_new API. sound/soc/qcom/lpass-cpu.c | 33 +++++++++++++++++++++++++++++++++ sound/soc/qcom/lpass-sc7180.c | 1 + sound/soc/qcom/lpass.h | 2 ++ 3 files changed, 36 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c62d2612e8f5..aff39c9f6326 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -29,6 +29,15 @@ #define LPASS_CPU_I2S_SD0_1_2_MASK GENMASK(2, 0) #define LPASS_CPU_I2S_SD0_1_2_3_MASK GENMASK(3, 0) +/* + * Channel maps for Quad channel playbacks on MI2S Secondary + */ +static struct snd_pcm_chmap_elem lpass_quad_chmaps[] = { + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_RL, + SNDRV_CHMAP_FR, SNDRV_CHMAP_RR } }, + { } +}; static int lpass_cpu_init_i2sctl_bitfields(struct device *dev, struct lpaif_i2sctl *i2sctl, struct regmap *map) { @@ -324,6 +333,25 @@ const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int ret; + struct snd_soc_dai_driver *drv = dai->driver; + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + + if (drvdata->mi2s_playback_sd_mode[dai->id] == LPAIF_I2SCTL_MODE_QUAD01) { + ret = snd_pcm_add_chmap_ctls(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, + lpass_quad_chmaps, drv->playback.channels_max, 0, + NULL); + if (ret < 0) + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(lpass_cpu_pcm_new); + int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); @@ -856,6 +884,11 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) PTR_ERR(drvdata->mi2s_bit_clk[dai_id])); return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]); } + if (drvdata->mi2s_playback_sd_mode[dai_id] == + LPAIF_I2SCTL_MODE_QUAD01) { + variant->dai_driver[dai_id].playback.channels_min = 4; + variant->dai_driver[dai_id].playback.channels_max = 4; + } } /* Allocation for i2sctl regmap fields */ diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 8c168d3c589e..77a556b27cf0 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -58,6 +58,7 @@ static struct snd_soc_dai_driver sc7180_lpass_cpu_dai_driver[] = { }, .probe = &asoc_qcom_lpass_cpu_dai_probe, .ops = &asoc_qcom_lpass_cpu_dai_ops, + .pcm_new = lpass_cpu_pcm_new, }, { .id = LPASS_DP_RX, .name = "Hdmi", diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..623ddccdafff 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -259,5 +259,7 @@ void asoc_qcom_lpass_cpu_platform_shutdown(struct platform_device *pdev); int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev); int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai); extern const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops; +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai); #endif /* __LPASS_H__ */ -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

4 3

[PATCH -next] ASoC: axi-i2s: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/adi/axi-i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index aa082131fb90..1289cb4e2988 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -198,8 +198,7 @@ static int axi_i2s_probe(struct platform_device *pdev) axi_i2s_parse_of(i2s, pdev->dev.of_node); - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap_resource(&pdev->dev, res); + base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(base)) return PTR_ERR(base); -- 2.25.1

2 1

[PATCH -next] ASoC: atmel-classd: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 14 Jun '21

14 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/atmel/atmel-classd.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index 6023369e0f1a..a9f9f449c48c 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -558,8 +558,7 @@ static int atmel_classd_probe(struct platform_device *pdev) return ret; } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - io_base = devm_ioremap_resource(dev, res); + io_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(io_base)) return PTR_ERR(io_base); -- 2.25.1

2 1

Re: [PATCH] ASoC:fsl_spdif:Remove superfluous error message around platform_get_irq()
by Mark Brown 14 Jun '21

14 Jun '21

On Thu, 10 Jun 2021 12:00:37 +0800, Zhongjun Tan wrote: > The platform_get_irq() prints error message telling that interrupt is > missing, hence there is no need to duplicated that message. Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC:fsl_spdif:Remove superfluous error message around platform_get_irq() commit: 2e8a8adb96a335a04f1697dd4314f5569521328f All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

1 0

ALSA 1.2.5.1 bugfix release
by Jaroslav Kysela 14 Jun '21

14 Jun '21

Hello all, the ALSA userspace alsa-lib, alsa-utils and alsa-ucm-conf packages were released. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.5_v1.2.5.1 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 0

USB sound card issues
by Muni Sekhar 14 Jun '21

14 Jun '21

Hi all, I am using a USB sound card(snd-usb-audio), and it fails while doing the audio recording "arecord: pcm_read:2032: read error: Input/output error". What could be the problem? and is there a workaround for this? $ arecord -l **** List of CAPTURE Hardware Devices **** card 1: BT_AG [BT_002], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord --device hw:1,0 --channels 1 --format S16_LE --rate 32000_Hz x.wav Recording WAVE 'x.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono arecord: pcm_read:2032: read error: Input/output error -- Thanks, Sekhar

3 10

added an option to aseqnet to optionally set the midi process name
by GitHub pull_request - reopened 14 Jun '21

14 Jun '21

alsa-project/alsa-utils pull request #95 was reopened from BrainDamage: This option allows to run multiple instances of aseqnet without having to double check the assigned port number, since each one can get spawned with a unique name. The patch is pretty minimal and clean, thopefully here shouldn't be troubles merging it. Request URL : https://github.com/alsa-project/alsa-utils/pull/95 Patch URL : https://github.com/alsa-project/alsa-utils/pull/95.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] alsa: control_led - fix initialization in the mode show callback
by Jaroslav Kysela 14 Jun '21

14 Jun '21

The str variable should be always initialized before use even if the switch covers all cases. This is a minimalistic fix: Assign NULL, the sprintf() may print '(null)' if something is corrupted. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/core/control_led.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 861f71f2fab5..764058cc345d 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -397,7 +397,7 @@ static ssize_t mode_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_ctl_led *led = container_of(dev, struct snd_ctl_led, dev); - const char *str; + const char *str = NULL; switch (led->mode) { case MODE_FOLLOW_MUTE: str = "follow-mute"; break; -- 2.30.2

2 1

[PATCH 0/7] Prepare EP93xx drivers for Common Clock Framework
by Alexander Sverdlin 14 Jun '21

14 Jun '21

Nikita posted a patch converting EP93xx to use Common Clock Framework. It turns out some cleanup is necessary in the EP93xx drivers to avoid "Enabling unprepared" clock warnings. Patches with stack traces in the commit messages are tested on EP9302. Link: https://lore.kernel.org/patchwork/patch/1435884/ Alexander Sverdlin (7): iio: ep93xx: Prepare clock before using it spi: spi-ep93xx: Prepare clock before using it Input: ep93xx_keypad: Prepare clock before using it video: ep93xx: Prepare clock before using it dmaengine: ep93xx: Prepare clock before using it ASoC: cirrus: i2s: Prepare clock before using it pwm: ep93xx: Prepare clock before using it drivers/dma/ep93xx_dma.c | 6 +++--- drivers/iio/adc/ep93xx_adc.c | 6 +++--- drivers/input/keyboard/ep93xx_keypad.c | 4 ++-- drivers/pwm/pwm-ep93xx.c | 12 ++++++------ drivers/spi/spi-ep93xx.c | 4 ++-- drivers/video/fbdev/ep93xx-fb.c | 4 ++-- sound/soc/cirrus/ep93xx-i2s.c | 12 ++++++------ 7 files changed, 24 insertions(+), 24 deletions(-) -- 2.32.0

3 3

[PATCH] ALSA: firewire: fill Kconfig with entries for devices based on DICE ASICs
by Takashi Sakamoto 14 Jun '21

14 Jun '21

Nowadays a lot of devices are supported by ALSA dice driver. This commit fills Kconfig with entries for them. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 21 +++++++++++++++++++-- 1 file changed, 19 insertions(+), 2 deletions(-) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index fc6858131b59..a08a0dbfc951 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -18,8 +18,25 @@ config SND_DICE select SND_HWDEP select SND_FIREWIRE_LIB help - Say Y here to include support for many DACs based on the DICE - chip family (DICE-II/Jr/Mini) which TC Applied Technologies produces. + Say Y here to include support for devices based on the DICE chip family + (DICE-II/TCD2210(Mini)/TCD2220(Jr.)) which TC Applied Technologies (TCAT) produced. + * Allen and Heath Zed R16 + * Alesis iO 14/26 FireWire, MasterControl, MultiMix 8/12/16 FireWire + * Avid Mbox 3 Pro + * FlexRadio Systems FLEX-3000, FLEX-5000 + * Focusrite Liquid Saffire 56 + * Focusrite Saffire Pro 14, Pro 24, Pro 24 DSP, Pro 26, Pro 40(TCD2220) + * Harman Music Group Lexicon I-ONIX FW810S + * Loud Technologies Mackie Onyx Blackbird, Onyx 820i/1220i/1620i/1640i (latter models) + * M-Audio ProFire 610/2626 + * Mytek Stereo192-DSD DAC + * Midas Klark Teknik VeniceF series + * PreSonus FireStudio, FireStudio Mobile, FireStudio Project, FireStudio Tube + * PreSonus StudioLive 16.4.2, 16.0.2, 24.4.2, 32.4.2 + * Solid State Logic Duende Classic, Duende Mini + * TC Electronic Studio Konnekt 48, Konnekt 24D, Konnekt Live, Impact Twin + * TC Electronic Digital Konnekt x32, Desktop Konnekt 6 + * Weiss Engineering ADC2, Vesta, Minerva, AFI1, DAC1, INT202, DAC202 To compile this driver as a module, choose M here: the module will be called snd-dice. -- 2.30.2

2 1

Android porting patches
by GitHub pull_request - opened 14 Jun '21

14 Jun '21

alsa-project/alsa-lib pull request #156 was opened from cwhuang: This is a series of patches to make alsa-lib buildable with Android source tree. Test on with Android 8.1 and later. Request URL : https://github.com/alsa-project/alsa-lib/pull/156 Patch URL : https://github.com/alsa-project/alsa-lib/pull/156.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v4] soundwire: intel: move to auxiliary bus
by Bard Liao 14 Jun '21

14 Jun '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Now that the auxiliary_bus exists, there's no reason to use platform devices as children of a PCI device any longer. This patch refactors the code by extending a basic auxiliary device with Intel link-specific structures that need to be passed between controller and link levels. This refactoring is much cleaner with no need for cross-pointers between device and link structures. Note that the auxiliary bus API has separate init and add steps, which requires more attention in the error unwinding paths. The main loop needs to deal with kfree() and auxiliary_device_uninit() for the current iteration before jumping to the common label which releases everything allocated in prior iterations. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- v2: - add link_dev_register for all kzalloc, device_init, and device_add. v3: - Modify the function description of sdw_intel_probe() which was missing in previous version. v4: - Move intel_link_dev_unregister(ldev) before pm_runtime_put_noidle( ldev->link_res.dev) to fix use-after-free reported by KASAN. --- drivers/soundwire/Kconfig | 1 + drivers/soundwire/intel.c | 56 ++++--- drivers/soundwire/intel.h | 14 +- drivers/soundwire/intel_init.c | 232 +++++++++++++++++++--------- include/linux/soundwire/sdw_intel.h | 6 +- 5 files changed, 202 insertions(+), 107 deletions(-) diff --git a/drivers/soundwire/Kconfig b/drivers/soundwire/Kconfig index 016e74230bb7..2b7795233282 100644 --- a/drivers/soundwire/Kconfig +++ b/drivers/soundwire/Kconfig @@ -25,6 +25,7 @@ config SOUNDWIRE_INTEL tristate "Intel SoundWire Master driver" select SOUNDWIRE_CADENCE select SOUNDWIRE_GENERIC_ALLOCATION + select AUXILIARY_BUS depends on ACPI && SND_SOC help SoundWire Intel Master driver. diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index fd95f94630b1..c11e3d8cd308 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -11,7 +11,7 @@ #include <linux/module.h> #include <linux/interrupt.h> #include <linux/io.h> -#include <linux/platform_device.h> +#include <linux/auxiliary_bus.h> #include <sound/pcm_params.h> #include <linux/pm_runtime.h> #include <sound/soc.h> @@ -1327,11 +1327,14 @@ static int intel_init(struct sdw_intel *sdw) } /* - * probe and init + * probe and init (aux_dev_id argument is required by function prototype but not used) */ -static int intel_master_probe(struct platform_device *pdev) +static int intel_link_probe(struct auxiliary_device *auxdev, + const struct auxiliary_device_id *aux_dev_id) + { - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; + struct sdw_intel_link_dev *ldev = auxiliary_dev_to_sdw_intel_link_dev(auxdev); struct sdw_intel *sdw; struct sdw_cdns *cdns; struct sdw_bus *bus; @@ -1344,14 +1347,14 @@ static int intel_master_probe(struct platform_device *pdev) cdns = &sdw->cdns; bus = &cdns->bus; - sdw->instance = pdev->id; - sdw->link_res = dev_get_platdata(dev); + sdw->instance = auxdev->id; + sdw->link_res = &ldev->link_res; cdns->dev = dev; cdns->registers = sdw->link_res->registers; cdns->instance = sdw->instance; cdns->msg_count = 0; - bus->link_id = pdev->id; + bus->link_id = auxdev->id; sdw_cdns_probe(cdns); @@ -1384,10 +1387,10 @@ static int intel_master_probe(struct platform_device *pdev) return 0; } -int intel_master_startup(struct platform_device *pdev) +int intel_link_startup(struct auxiliary_device *auxdev) { struct sdw_cdns_stream_config config; - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); struct sdw_bus *bus = &cdns->bus; @@ -1524,9 +1527,9 @@ int intel_master_startup(struct platform_device *pdev) return ret; } -static int intel_master_remove(struct platform_device *pdev) +static void intel_link_remove(struct auxiliary_device *auxdev) { - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); struct sdw_bus *bus = &cdns->bus; @@ -1542,19 +1545,17 @@ static int intel_master_remove(struct platform_device *pdev) snd_soc_unregister_component(dev); } sdw_bus_master_delete(bus); - - return 0; } -int intel_master_process_wakeen_event(struct platform_device *pdev) +int intel_link_process_wakeen_event(struct auxiliary_device *auxdev) { - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; struct sdw_intel *sdw; struct sdw_bus *bus; void __iomem *shim; u16 wake_sts; - sdw = platform_get_drvdata(pdev); + sdw = dev_get_drvdata(dev); bus = &sdw->cdns.bus; if (bus->prop.hw_disabled) { @@ -1976,17 +1977,22 @@ static const struct dev_pm_ops intel_pm = { SET_RUNTIME_PM_OPS(intel_suspend_runtime, intel_resume_runtime, NULL) }; -static struct platform_driver sdw_intel_drv = { - .probe = intel_master_probe, - .remove = intel_master_remove, +static const struct auxiliary_device_id intel_link_id_table[] = { + { .name = "soundwire_intel.link" }, + {}, +}; +MODULE_DEVICE_TABLE(auxiliary, intel_link_id_table); + +static struct auxiliary_driver sdw_intel_drv = { + .probe = intel_link_probe, + .remove = intel_link_remove, .driver = { - .name = "intel-sdw", + /* auxiliary_driver_register() sets .name to be the modname */ .pm = &intel_pm, - } + }, + .id_table = intel_link_id_table }; - -module_platform_driver(sdw_intel_drv); +module_auxiliary_driver(sdw_intel_drv); MODULE_LICENSE("Dual BSD/GPL"); -MODULE_ALIAS("platform:intel-sdw"); -MODULE_DESCRIPTION("Intel Soundwire Master Driver"); +MODULE_DESCRIPTION("Intel Soundwire Link Driver"); diff --git a/drivers/soundwire/intel.h b/drivers/soundwire/intel.h index 06bac8ba14e9..0b47b148da3f 100644 --- a/drivers/soundwire/intel.h +++ b/drivers/soundwire/intel.h @@ -7,7 +7,6 @@ /** * struct sdw_intel_link_res - Soundwire Intel link resource structure, * typically populated by the controller driver. - * @pdev: platform_device * @mmio_base: mmio base of SoundWire registers * @registers: Link IO registers base * @shim: Audio shim pointer @@ -23,7 +22,6 @@ * @list: used to walk-through all masters exposed by the same controller */ struct sdw_intel_link_res { - struct platform_device *pdev; void __iomem *mmio_base; /* not strictly needed, useful for debug */ void __iomem *registers; void __iomem *shim; @@ -48,7 +46,15 @@ struct sdw_intel { #endif }; -int intel_master_startup(struct platform_device *pdev); -int intel_master_process_wakeen_event(struct platform_device *pdev); +int intel_link_startup(struct auxiliary_device *auxdev); +int intel_link_process_wakeen_event(struct auxiliary_device *auxdev); + +struct sdw_intel_link_dev { + struct auxiliary_device auxdev; + struct sdw_intel_link_res link_res; +}; + +#define auxiliary_dev_to_sdw_intel_link_dev(auxiliary_dev) \ + container_of(auxiliary_dev, struct sdw_intel_link_dev, auxdev) #endif /* __SDW_INTEL_LOCAL_H */ diff --git a/drivers/soundwire/intel_init.c b/drivers/soundwire/intel_init.c index 30ce95ec2d70..9e283bef53d2 100644 --- a/drivers/soundwire/intel_init.c +++ b/drivers/soundwire/intel_init.c @@ -12,7 +12,7 @@ #include <linux/interrupt.h> #include <linux/io.h> #include <linux/module.h> -#include <linux/platform_device.h> +#include <linux/auxiliary_bus.h> #include <linux/pm_runtime.h> #include <linux/soundwire/sdw_intel.h> #include "cadence_master.h" @@ -24,28 +24,108 @@ #define SDW_LINK_BASE 0x30000 #define SDW_LINK_SIZE 0x10000 +static void intel_link_dev_release(struct device *dev) +{ + struct auxiliary_device *auxdev = to_auxiliary_dev(dev); + struct sdw_intel_link_dev *ldev = auxiliary_dev_to_sdw_intel_link_dev(auxdev); + + kfree(ldev); +} + +/* alloc, init and add link devices */ +static struct sdw_intel_link_dev *intel_link_dev_register(struct sdw_intel_res *res, + struct sdw_intel_ctx *ctx, + struct fwnode_handle *fwnode, + const char *name, + int link_id) +{ + struct sdw_intel_link_dev *ldev; + struct sdw_intel_link_res *link; + struct auxiliary_device *auxdev; + int ret; + + ldev = kzalloc(sizeof(*ldev), GFP_KERNEL); + if (!ldev) + return ERR_PTR(-ENOMEM); + + auxdev = &ldev->auxdev; + auxdev->name = name; + auxdev->dev.parent = res->parent; + auxdev->dev.fwnode = fwnode; + auxdev->dev.release = intel_link_dev_release; + + /* we don't use an IDA since we already have a link ID */ + auxdev->id = link_id; + + /* + * keep a handle on the allocated memory, to be used in all other functions. + * Since the same pattern is used to skip links that are not enabled, there is + * no need to check if ctx->ldev[i] is NULL later on. + */ + ctx->ldev[link_id] = ldev; + + /* Add link information used in the driver probe */ + link = &ldev->link_res; + link->mmio_base = res->mmio_base; + link->registers = res->mmio_base + SDW_LINK_BASE + + (SDW_LINK_SIZE * link_id); + link->shim = res->mmio_base + SDW_SHIM_BASE; + link->alh = res->mmio_base + SDW_ALH_BASE; + + link->ops = res->ops; + link->dev = res->dev; + + link->clock_stop_quirks = res->clock_stop_quirks; + link->shim_lock = &ctx->shim_lock; + link->shim_mask = &ctx->shim_mask; + link->link_mask = ctx->link_mask; + + /* now follow the two-step init/add sequence */ + ret = auxiliary_device_init(auxdev); + if (ret < 0) { + dev_err(res->parent, "failed to initialize link dev %s link_id %d\n", + name, link_id); + kfree(ldev); + return ERR_PTR(ret); + } + + ret = auxiliary_device_add(&ldev->auxdev); + if (ret < 0) { + dev_err(res->parent, "failed to add link dev %s link_id %d\n", + ldev->auxdev.name, link_id); + /* ldev will be freed with the put_device() and .release sequence */ + auxiliary_device_uninit(&ldev->auxdev); + return ERR_PTR(ret); + } + + return ldev; +} + +static void intel_link_dev_unregister(struct sdw_intel_link_dev *ldev) +{ + auxiliary_device_delete(&ldev->auxdev); + auxiliary_device_uninit(&ldev->auxdev); +} + static int sdw_intel_cleanup(struct sdw_intel_ctx *ctx) { - struct sdw_intel_link_res *link = ctx->links; + struct sdw_intel_link_dev *ldev; u32 link_mask; int i; - if (!link) - return 0; - link_mask = ctx->link_mask; - for (i = 0; i < ctx->count; i++, link++) { + for (i = 0; i < ctx->count; i++) { if (!(link_mask & BIT(i))) continue; - if (link->pdev) { - pm_runtime_disable(&link->pdev->dev); - platform_device_unregister(link->pdev); - } + ldev = ctx->ldev[i]; - if (!link->clock_stop_quirks) - pm_runtime_put_noidle(link->dev); + pm_runtime_disable(&ldev->auxdev.dev); + if (!ldev->link_res.clock_stop_quirks) + pm_runtime_put_noidle(ldev->link_res.dev); + + intel_link_dev_unregister(ldev); } return 0; @@ -91,9 +171,8 @@ EXPORT_SYMBOL_NS(sdw_intel_thread, SOUNDWIRE_INTEL_INIT); static struct sdw_intel_ctx *sdw_intel_probe_controller(struct sdw_intel_res *res) { - struct platform_device_info pdevinfo; - struct platform_device *pdev; struct sdw_intel_link_res *link; + struct sdw_intel_link_dev *ldev; struct sdw_intel_ctx *ctx; struct acpi_device *adev; struct sdw_slave *slave; @@ -116,67 +195,60 @@ static struct sdw_intel_ctx count = res->count; dev_dbg(&adev->dev, "Creating %d SDW Link devices\n", count); - ctx = devm_kzalloc(&adev->dev, sizeof(*ctx), GFP_KERNEL); + /* + * we need to alloc/free memory manually and can't use devm: + * this routine may be called from a workqueue, and not from + * the parent .probe. + * If devm_ was used, the memory might never be freed on errors. + */ + ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); if (!ctx) return NULL; ctx->count = count; - ctx->links = devm_kcalloc(&adev->dev, ctx->count, - sizeof(*ctx->links), GFP_KERNEL); - if (!ctx->links) + + /* + * allocate the array of pointers. The link-specific data is allocated + * as part of the first loop below and released with the auxiliary_device_uninit(). + * If some links are disabled, the link pointer will remain NULL. Given that the + * number of links is small, this is simpler than using a list to keep track of links. + */ + ctx->ldev = kcalloc(ctx->count, sizeof(*ctx->ldev), GFP_KERNEL); + if (!ctx->ldev) { + kfree(ctx); return NULL; + } - ctx->count = count; ctx->mmio_base = res->mmio_base; ctx->link_mask = res->link_mask; ctx->handle = res->handle; mutex_init(&ctx->shim_lock); - link = ctx->links; link_mask = ctx->link_mask; INIT_LIST_HEAD(&ctx->link_list); - /* Create SDW Master devices */ - for (i = 0; i < count; i++, link++) { - if (!(link_mask & BIT(i))) { - dev_dbg(&adev->dev, - "Link %d masked, will not be enabled\n", i); + for (i = 0; i < count; i++) { + if (!(link_mask & BIT(i))) continue; - } - link->mmio_base = res->mmio_base; - link->registers = res->mmio_base + SDW_LINK_BASE - + (SDW_LINK_SIZE * i); - link->shim = res->mmio_base + SDW_SHIM_BASE; - link->alh = res->mmio_base + SDW_ALH_BASE; - - link->ops = res->ops; - link->dev = res->dev; - - link->clock_stop_quirks = res->clock_stop_quirks; - link->shim_lock = &ctx->shim_lock; - link->shim_mask = &ctx->shim_mask; - link->link_mask = link_mask; - - memset(&pdevinfo, 0, sizeof(pdevinfo)); - - pdevinfo.parent = res->parent; - pdevinfo.name = "intel-sdw"; - pdevinfo.id = i; - pdevinfo.fwnode = acpi_fwnode_handle(adev); - pdevinfo.data = link; - pdevinfo.size_data = sizeof(*link); - - pdev = platform_device_register_full(&pdevinfo); - if (IS_ERR(pdev)) { - dev_err(&adev->dev, - "platform device creation failed: %ld\n", - PTR_ERR(pdev)); + /* + * init and add a device for each link + * + * The name of the device will be soundwire_intel.link.[i], + * with the "soundwire_intel" module prefix automatically added + * by the auxiliary bus core. + */ + ldev = intel_link_dev_register(res, + ctx, + acpi_fwnode_handle(adev), + "link", + i); + if (IS_ERR(ldev)) goto err; - } - link->pdev = pdev; - link->cdns = platform_get_drvdata(pdev); + + link = &ldev->link_res; + link->cdns = dev_get_drvdata(&ldev->auxdev.dev); if (!link->cdns) { dev_err(&adev->dev, "failed to get link->cdns\n"); @@ -194,8 +266,7 @@ static struct sdw_intel_ctx num_slaves++; } - ctx->ids = devm_kcalloc(&adev->dev, num_slaves, - sizeof(*ctx->ids), GFP_KERNEL); + ctx->ids = kcalloc(num_slaves, sizeof(*ctx->ids), GFP_KERNEL); if (!ctx->ids) goto err; @@ -213,8 +284,14 @@ static struct sdw_intel_ctx return ctx; err: - ctx->count = i; - sdw_intel_cleanup(ctx); + while (i--) { + if (!(link_mask & BIT(i))) + continue; + ldev = ctx->ldev[i]; + intel_link_dev_unregister(ldev); + } + kfree(ctx->ldev); + kfree(ctx); return NULL; } @@ -222,7 +299,7 @@ static int sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) { struct acpi_device *adev; - struct sdw_intel_link_res *link; + struct sdw_intel_link_dev *ldev; u32 caps; u32 link_mask; int i; @@ -241,27 +318,28 @@ sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) return -EINVAL; } - if (!ctx->links) + if (!ctx->ldev) return -EINVAL; - link = ctx->links; link_mask = ctx->link_mask; /* Startup SDW Master devices */ - for (i = 0; i < ctx->count; i++, link++) { + for (i = 0; i < ctx->count; i++) { if (!(link_mask & BIT(i))) continue; - intel_master_startup(link->pdev); + ldev = ctx->ldev[i]; - if (!link->clock_stop_quirks) { + intel_link_startup(&ldev->auxdev); + + if (!ldev->link_res.clock_stop_quirks) { /* * we need to prevent the parent PCI device * from entering pm_runtime suspend, so that * power rails to the SoundWire IP are not * turned off. */ - pm_runtime_get_noresume(link->dev); + pm_runtime_get_noresume(ldev->link_res.dev); } } @@ -272,8 +350,8 @@ sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) * sdw_intel_probe() - SoundWire Intel probe routine * @res: resource data * - * This registers a platform device for each Master handled by the controller, - * and SoundWire Master and Slave devices will be created by the platform + * This registers an auxiliary device for each Master handled by the controller, + * and SoundWire Master and Slave devices will be created by the auxiliary * device probe. All the information necessary is stored in the context, and * the res argument pointer can be freed after this step. * This function will be called after sdw_intel_acpi_scan() by SOF probe. @@ -306,27 +384,31 @@ EXPORT_SYMBOL_NS(sdw_intel_startup, SOUNDWIRE_INTEL_INIT); void sdw_intel_exit(struct sdw_intel_ctx *ctx) { sdw_intel_cleanup(ctx); + kfree(ctx->ids); + kfree(ctx->ldev); + kfree(ctx); } EXPORT_SYMBOL_NS(sdw_intel_exit, SOUNDWIRE_INTEL_INIT); void sdw_intel_process_wakeen_event(struct sdw_intel_ctx *ctx) { - struct sdw_intel_link_res *link; + struct sdw_intel_link_dev *ldev; u32 link_mask; int i; - if (!ctx->links) + if (!ctx->ldev) return; - link = ctx->links; link_mask = ctx->link_mask; /* Startup SDW Master devices */ - for (i = 0; i < ctx->count; i++, link++) { + for (i = 0; i < ctx->count; i++) { if (!(link_mask & BIT(i))) continue; - intel_master_process_wakeen_event(link->pdev); + ldev = ctx->ldev[i]; + + intel_link_process_wakeen_event(&ldev->auxdev); } } EXPORT_SYMBOL_NS(sdw_intel_process_wakeen_event, SOUNDWIRE_INTEL_INIT); diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index 3a5446ac014a..1ebea7764011 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -58,7 +58,7 @@ struct sdw_intel_acpi_info { u32 link_mask; }; -struct sdw_intel_link_res; +struct sdw_intel_link_dev; /* Intel clock-stop/pm_runtime quirk definitions */ @@ -109,7 +109,7 @@ struct sdw_intel_slave_id { * Controller * @num_slaves: total number of devices exposed across all enabled links * @handle: ACPI parent handle - * @links: information for each link (controller-specific and kept + * @ldev: information for each link (controller-specific and kept * opaque here) * @ids: array of slave_id, representing Slaves exposed across all enabled * links @@ -123,7 +123,7 @@ struct sdw_intel_ctx { u32 link_mask; int num_slaves; acpi_handle handle; - struct sdw_intel_link_res *links; + struct sdw_intel_link_dev **ldev; struct sdw_intel_slave_id *ids; struct list_head link_list; struct mutex shim_lock; /* lock for access to shared SHIM registers */ -- 2.17.1

5 14

[PATCH] ASoC: rsnd: adg: clearly handle clock error / NULL case
by Kuninori Morimoto 14 Jun '21

14 Jun '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This driver is assuming that all adg->clk[i] is not NULL. Because of this prerequisites, for_each_rsnd_clk() is possible to work for all clk without checking NULL. In other words, all adg->clk[i] should not NULL. Some SoC might doesn't have clk_a/b/c/i. devm_clk_get() returns error in such case. This driver calls rsnd_adg_null_clk_get() and use null_clk instead of NULL in such cases. But devm_clk_get() might returns NULL even though such clocks exist, but it doesn't mean error (user deliberately chose to disable the feature). NULL clk itself is not error from clk point of view, but is error from this driver point of view because it is not assuming such case. But current code is using IS_ERR() which doesn't care NULL. This driver uses IS_ERR_OR_NULL() instead of IS_ERR() for clk check. And it uses ERR_CAST() to clarify null_clk error. One concern here is that it unconditionally uses null_clk if clk_a/b/c/i was error. It is correct if it doesn't exist, but is not correct if it returns error even though it exist. It needs to check "clock-names" from DT before calling devm_clk_get() to handling such case. But let's assume it is overkill so far. Link: https://lore.kernel.org/r/YMCmhfQUimHCSH/n@mwanda Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 0ebee1ed06a9..abe9d539709b 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -393,7 +393,7 @@ static struct clk *rsnd_adg_create_null_clk(struct rsnd_priv *priv, clk = clk_register_fixed_rate(dev, name, parent, 0, 0); if (IS_ERR(clk)) { dev_err(dev, "create null clk error\n"); - return NULL; + return ERR_CAST(clk); } return clk; @@ -430,9 +430,9 @@ static int rsnd_adg_get_clkin(struct rsnd_priv *priv) for (i = 0; i < CLKMAX; i++) { clk = devm_clk_get(dev, clk_name[i]); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) clk = rsnd_adg_null_clk_get(priv); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) goto err; adg->clk[i] = clk; -- 2.25.1

2 2

Some minor fixes for Topology2.0
by GitHub pull_request - opened 13 Jun '21

13 Jun '21

alsa-project/alsa-utils pull request #98 was opened from ranj063: Request URL : https://github.com/alsa-project/alsa-utils/pull/98 Patch URL : https://github.com/alsa-project/alsa-utils/pull/98.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH -next 0/9] ASoC: fsl: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 13 Jun '21

13 Jun '21

patch #1 ~ #8: Use devm_platform_get_and_ioremap_resource() patch #9 check return value of platform_get_resource_byname() Yang Yingliang (9): ASoC: fsl_asrc: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_aud2htx: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_easrc: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_esai: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_micfil: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_sai: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_spdif: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_ssi: Use devm_platform_get_and_ioremap_resource() ASoC: fsl_xcvr: check return value after calling platform_get_resource_byname() sound/soc/fsl/fsl_asrc.c | 3 +-- sound/soc/fsl/fsl_aud2htx.c | 3 +-- sound/soc/fsl/fsl_easrc.c | 3 +-- sound/soc/fsl/fsl_esai.c | 3 +-- sound/soc/fsl/fsl_micfil.c | 3 +-- sound/soc/fsl/fsl_sai.c | 3 +-- sound/soc/fsl/fsl_spdif.c | 3 +-- sound/soc/fsl/fsl_ssi.c | 3 +-- sound/soc/fsl/fsl_xcvr.c | 4 ++++ 9 files changed, 12 insertions(+), 16 deletions(-) -- 2.25.1

2 10

alsa-lib 1.2.5 breaks espeak
by GitHub issues - opened 13 Jun '21

13 Jun '21

alsa-project/alsa-lib issue #155 was opened from MrBitBucket: In archlinux amd64 upgrading alsa-lib from 1.2.4 - 1.2.5 breaks the espeak application https://sourceforge.net/projects/espeak/. I see extra errors in espeak output eg ```Expression 'framesAvail' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 3629 Expression 'PaAlsaStreamComponent_GetAvailableFrames( self, &framesAvail, xrun )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 3989 Expression 'PaAlsaStreamComponent_RegisterChannels( &self->playback, &self->bufferProcessor, &playbackFrames, &xrun )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4118 Expression 'PaAlsaStream_SetUpBuffers( stream, &framesGot, &xrun )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4359``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/155 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/6] ALSA: bebob/fireworks: device entry corrections
by Takashi Sakamoto 12 Jun '21

12 Jun '21

Hi, After working for hardware database of systemd[1], I realize that current device entries of ALSA bebob/fireworks drivers includes some issues. This patchset corrects them. [1] https://github.com/systemd/systemd/pull/19124 Takashi Sakamoto (6): ALSA: bebob: fulfil device entries ALSA: fireworks: fulfil device entries ALSA: bebob: correct device entry for Mackie D.2 FireWire option card ALSA: bebob: correct device entry for Acoustic Reality eAR Master One, Eroica, Figaro, and Ciaccona ALSA: bebob: code refactoring for M-Audio models ALSA: bebob: correct device entries for Phonic Helix Board and FireFly series sound/firewire/bebob/bebob.c | 92 ++++++++++++++-------------- sound/firewire/bebob/bebob.h | 9 --- sound/firewire/fireworks/fireworks.c | 15 +++++ sound/firewire/fireworks/fireworks.h | 8 --- 4 files changed, 61 insertions(+), 63 deletions(-) -- 2.30.2

2 7

[PATCH][next] ALSA: i2c: tea6330t: Remove redundant initialization of variable err
by Colin King 12 Jun '21

12 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> The variable err is being initialized with a value that is never read, it is being updated later on. The assignment is redundant and can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/i2c/tea6330t.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/i2c/tea6330t.c b/sound/i2c/tea6330t.c index 742d0f724375..037d6293f728 100644 --- a/sound/i2c/tea6330t.c +++ b/sound/i2c/tea6330t.c @@ -284,7 +284,7 @@ int snd_tea6330t_update_mixer(struct snd_card *card, struct tea6330t *tea; const struct snd_kcontrol_new *knew; unsigned int idx; - int err = -ENOMEM; + int err; u8 default_treble, default_bass; unsigned char bytes[7]; -- 2.31.1

2 1

[PATCH] ALSA: firewire-lib: replace in_interrupt() with in_softirq()
by Takashi Sakamoto 12 Jun '21

12 Jun '21

Tasklet is used to handle isochronous context of 1394 OHCI. The explicit usage of in_softirq() may be preferable than in_interrupt(). Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream-trace.h | 2 +- sound/firewire/amdtp-stream.c | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h index aa53c13b89d3..5fd2aeccdfc2 100644 --- a/sound/firewire/amdtp-stream-trace.h +++ b/sound/firewire/amdtp-stream-trace.h @@ -49,7 +49,7 @@ TRACE_EVENT(amdtp_packet, __entry->data_blocks = data_blocks; __entry->data_block_counter = data_block_counter, __entry->packet_index = packet_index; - __entry->irq = !!in_interrupt(); + __entry->irq = !!in_softirq(); __entry->index = index; ), TP_printk( diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 1d9bc7b07df1..aad9778d1c4d 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -610,7 +610,7 @@ static void update_pcm_pointers(struct amdtp_stream *s, // buffer associated to PCM substream to process PCM frames in the buffer, instead // of receiving notification of period elapsed by poll wait. if (!pcm->runtime->no_period_wakeup) { - if (in_interrupt()) { + if (in_softirq()) { // In software IRQ context for 1394 OHCI. snd_pcm_period_elapsed(pcm); } else { @@ -1026,7 +1026,7 @@ static void generate_pkt_descs(struct amdtp_stream *s, const __be32 *ctx_header, static inline void cancel_stream(struct amdtp_stream *s) { s->packet_index = -1; - if (in_interrupt()) + if (in_softirq()) amdtp_stream_pcm_abort(s); WRITE_ONCE(s->pcm_buffer_pointer, SNDRV_PCM_POS_XRUN); } @@ -1738,7 +1738,7 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d, if (irq_target && amdtp_stream_running(irq_target)) { // In software IRQ context, the call causes dead-lock to disable the tasklet // synchronously. - if (!in_interrupt()) + if (!in_softirq()) fw_iso_context_flush_completions(irq_target->context); } -- 2.30.2

2 1

[PATCH -next] ALSA: n64: check return value after calling platform_get_resource()
by Yang Yingliang 12 Jun '21

12 Jun '21

It will cause null-ptr-deref if platform_get_resource() returns NULL, we need check the return value. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/mips/snd-n64.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/mips/snd-n64.c b/sound/mips/snd-n64.c index e35e93157755..463a6fe589eb 100644 --- a/sound/mips/snd-n64.c +++ b/sound/mips/snd-n64.c @@ -338,6 +338,10 @@ static int __init n64audio_probe(struct platform_device *pdev) strcpy(card->longname, "N64 Audio"); res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!res) { + err = -EINVAL; + goto fail_dma_alloc; + } if (devm_request_irq(&pdev->dev, res->start, n64audio_isr, IRQF_SHARED, "N64 Audio", priv)) { err = -EBUSY; -- 2.25.1

2 1

[PATCH -next] ALSA: hda/tegra: Use devm_platform_get_and_ioremap_resource()
by Yang Yingliang 12 Jun '21

12 Jun '21

Use devm_platform_get_and_ioremap_resource() to simplify code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/pci/hda/hda_tegra.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 6f2b743b9d75..5aeef6123781 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -235,11 +235,9 @@ static int hda_tegra_init_chip(struct azx *chip, struct platform_device *pdev) { struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); struct hdac_bus *bus = azx_bus(chip); - struct device *dev = hda->dev; struct resource *res; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - hda->regs = devm_ioremap_resource(dev, res); + hda->regs = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(hda->regs)) return PTR_ERR(hda->regs); -- 2.25.1

2 1

[PATCH -next] ALSA: trident: Fix build error
by YueHaibing 12 Jun '21

12 Jun '21

sound/pci/trident/trident_memory.c: In function ‘set_tlb_bus’: sound/pci/trident/trident_memory.c:85:35: error: ‘pagetr’ undeclared (first use in this function); did you mean ‘page’? for (i = 0; i < UNIT_PAGES; i++, pagetr++) { ^~~~~~ page Fixes: 74fb98311c4e ("ALSA: trident: Drop shadow TLB pointer table") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/pci/trident/trident_memory.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c index f831ec02702f..4ad3855101c9 100644 --- a/sound/pci/trident/trident_memory.c +++ b/sound/pci/trident/trident_memory.c @@ -82,7 +82,7 @@ static inline void set_tlb_bus(struct snd_trident *trident, int page, { int i; page *= UNIT_PAGES; - for (i = 0; i < UNIT_PAGES; i++, pagetr++) { + for (i = 0; i < UNIT_PAGES; i++, page++) { __set_tlb_bus(trident, page, addr); addr += SNDRV_TRIDENT_PAGE_SIZE; } -- 2.17.1

2 1

Question about upmix plugin
by GitHub issues - opened 11 Jun '21

11 Jun '21

alsa-project/alsa-plugins issue #25 was opened from krumpfwylg: Hi, I've noticed the [latest commits](https://github.com/alsa-project/alsa-plugins/pull/2) done to the upmix plugin, but there's something I don't understand. Line 24 of pcm_upmix.c is _#define UPMIX_PCM_FORMAT SND_PCM_FORMAT_S16_ Does that mean the upmix plugin is still only capable of using S16_LE format ? Or does the UPMIX_PCM_FORMAT value can get changed to S32_LE by tweaking it in asoundrc file ? Issue URL : https://github.com/alsa-project/alsa-plugins/issues/25 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

Config parsing of pcm_rate broken since 1.1.7 (at least for a52 pcm plugin)
by GitHub issues - opened 11 Jun '21

11 Jun '21

alsa-project/alsa-lib issue #154 was opened from quequotion: As noted [in the mailing list](https://mailman.alsa-project.org/pipermail/alsa-devel/2019-December/1… back in 2019, config parsing (for alsa-plugins a52 pcm at least) has been broken since alsa-lib 1.1.7. As far as I know this was never bisected to a single commit in either alsa-lib or alsa-plugins. Looking at the mailing list discussion, it occurs to me that it may have given the impression that the config parser was broken when using [a patch that ports alsa-plugins to libswresample](https://git.harting.dev/anonfunc/alsa-plugins/commit/9cdbbb9874757b6f8fda7fb4ac2e3fc59da65946). Config parsing is broken *regardless* of that patch. @IdleGandalf worked around this by applying a patch to alsa-lib, in [src/pcm/pcm_rate.c](https://git.harting.dev/anonfunc/alsa-lib/commit/646bc80a63b0220300678a38e0c26a9a13af6fc0), but never got a response if this was an appropriate method to resolve the issue. Should this be fixed in alsa-lib or in alsa-plugins? Note: above links are to mirrors of the original patches, also in a repository owned by @IdleGandalf as far as I can tell. The links in the mailing list discussion are now 404. Issue URL : https://github.com/alsa-project/alsa-lib/issues/154 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

SNDRV_PCM_IOCTL_READI_FRAMES returns EIO
by Muni Sekhar 11 Jun '21

11 Jun '21

Hi all, I am using a USB sound card(snd-usb-audio), and it fails while doing the audio recording "arecord: pcm_read:2032: read error: Input/output error". I noticed recording(arecord) audio data on a USB sound card fails at SNDRV_PCM_IOCTL_READI_FRAMES is giving EIO. Can someone please clarify when the kernel returns EIO for Ioctl SNDRV_PCM_IOCTL_READI_FRAMES? and is there a workaround for this? Output of the alsamixer is attached. -- Thanks, Sekhar

2 5

[PATCH 0/3] ALSA: bebob: remove workarounds for suppressed quirks
by Takashi Sakamoto 11 Jun '21

11 Jun '21

Hi, As a result to apply the commit 1bd1b3be8655 ("ALSA: bebob: perform sequence replay for media clock recovery"), some quirks seem to be suppressed. This patchset removes workarounds for the suppressed quirks and includes code cleanup. Takashi Sakamoto (3): ALSA: bebob: dismiss sleep after breaking connections ALSA: bebob: delete workaround for protocol version 3 ALSA: bebob: code refactoring for model-dependent quirks sound/firewire/bebob/bebob.c | 41 ++++++++++++++++++++--------- sound/firewire/bebob/bebob.h | 10 ++++--- sound/firewire/bebob/bebob_stream.c | 32 ++++++---------------- 3 files changed, 42 insertions(+), 41 deletions(-) -- 2.27.0

2 4

Add support for automatic attributes in Topology2.0
by GitHub pull_request - opened 11 Jun '21

11 Jun '21

alsa-project/alsa-utils pull request #97 was opened from ranj063: Add support for automatic attributes in Topology2.0 and also add support for processing objects that are simply used to update the parent object's private data. Request URL : https://github.com/alsa-project/alsa-utils/pull/97 Patch URL : https://github.com/alsa-project/alsa-utils/pull/97.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

ALSA lib pcm_dmix.c:1063:(snd_pcm_dmix_open) unable to create IPC semaphore
by GitHub issues - edited 11 Jun '21

11 Jun '21

alsa-project/alsa-lib issue #153 was edited from Cean1024: alsa info: alsa-config-1.0 alsa-lib-1.1.5 alsa-plugins-1.1.5 alsa-utils-1.1.5 asound.conf: ``` pcm.dmixer { type dmix ipc_key 1024 slave { pcm "hw:0,0" period_time 0 period_size 1024 buffer_size 4096 rate 48000 format "S32_LE" channels 2 } bindings { 0 0 1 1 } } ``` test cmd: ``` arecord -f dat | aplay -f dat -D plug:dmixer ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/153 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ALSA lib pcm_dmix.c:1063:(snd_pcm_dmix_open) unable to create IPC semaphore
by GitHub issues - edited 11 Jun '21

11 Jun '21

alsa-project/alsa-lib issue #153 was edited from Cean1024: alsa info: alsa-config-1.0 alsa-lib-1.1.5 alsa-plugins-1.1.5 alsa-utils-1.1.5 asound.conf: pcm.dmixer { type dmix ipc_key 1024 slave { pcm "hw:0,0" period_time 0 period_size 1024 buffer_size 4096 rate 48000 format "S32_LE" channels 2 } bindings { 0 0 1 1 } } test cmd: arecord -f dat | aplay -f dat -D plug:dmixer Issue URL : https://github.com/alsa-project/alsa-lib/issues/153 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2 0/3] ALSA: pcm:firewire: allow to operate for period elapse event in process context
by Takashi Sakamoto 11 Jun '21

11 Jun '21

Hi, This patchset is revised version of my previous one: * https://lore.kernel.org/alsa-devel/20210609012244.24296-1-o-takashi@sakamoc… All of drivers in ALSA firewire stack have two chances to process isochronous packets of any isochronous context; in software IRQ context for 1394 OHCI, and in process context of ALSA PCM application. In the process context, callbacks of .pointer and .ack are utilized. The callbacks are done by ALSA PCM core under acquiring lock of PCM substream, In design of ALSA PCM core, call of snd_pcm_period_elapsed() is used for drivers to awaken user processes from waiting for available frames. The function voluntarily acquires lock of PCM substream, therefore it is not called in the process context since it causes dead lock. As a workaround to avoid the dead lock, all of drivers in ALSA firewire stack use workqueue to delegate the call. This patchset is my attempt for the issue. A variant of 'snd_pcm_period_elapsed()' without lock acquisition is going to be added, named 'snd_pcm_period_elapsed_under_stream_lock()'. The call is available in callbacks of .pointer and .ack of snd_pcm_ops structure. Changes from v1: * fix context section of kernel API documentation Takashi Sakamoto (3): ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream ALSA: firewire-lib: operate for period elapse event in process context ALSA: firewire-lib: obsolete workqueue for period update include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 71 +++++++++++++++++++++++++++-------- sound/firewire/amdtp-stream.c | 46 +++++++---------------- sound/firewire/amdtp-stream.h | 1 - 4 files changed, 71 insertions(+), 48 deletions(-) -- 2.27.0

2 16

ALSA lib pcm_dmix.c:1063:(snd_pcm_dmix_open) unable to create IPC semaphore
by GitHub issues - opened 11 Jun '21

11 Jun '21

alsa-project/alsa-lib issue #153 was opened from Cean1024: Issue URL : https://github.com/alsa-project/alsa-lib/issues/153 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Re: [PATCH] ASoC:fsl_spdif:Remove superfluous error message around platform_get_irq()
by Shengjiu Wang 11 Jun '21

11 Jun '21

On Thu, Jun 10, 2021 at 12:02 PM Zhongjun Tan <hbut_tan(a)163.com> wrote: > > From: Tan Zhongjun <tanzhongjun(a)yulong.com> > > The platform_get_irq() prints error message telling that interrupt is > missing, hence there is no need to duplicated that message. > > Signed-off-by: Tan Zhongjun <tanzhongjun(a)yulong.com> Acked-by: Shengjiu Wang <shengjiu.wang(a)gmail.com> > --- > sound/soc/fsl/fsl_spdif.c | 4 +--- > 1 file changed, 1 insertion(+), 3 deletions(-) > > diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c > index 2a76714eb8e6..29cefd459241 100644 > --- a/sound/soc/fsl/fsl_spdif.c > +++ b/sound/soc/fsl/fsl_spdif.c > @@ -1368,10 +1368,8 @@ static int fsl_spdif_probe(struct platform_device *pdev) > > for (i = 0; i < spdif_priv->soc->interrupts; i++) { > irq = platform_get_irq(pdev, i); > - if (irq < 0) { > - dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); > + if (irq < 0) > return irq; > - } > > ret = devm_request_irq(&pdev->dev, irq, spdif_isr, 0, > dev_name(&pdev->dev), spdif_priv); > -- > 2.17.1 >

1 0

Re: [PATCH] ASoC:fsl_easrc:Remove superfluous error message around platform_get_irq()
by Shengjiu Wang 11 Jun '21

11 Jun '21

On Thu, Jun 10, 2021 at 8:52 PM Zhongjun Tan <hbut_tan(a)163.com> wrote: > > From: Tan Zhongjun <tanzhongjun(a)yulong.com> > > Clean up the check for irq.dev_err is superfluous as platform_get_irq() > already prints an error.Remove curly braces to confirm to styling > requirements. > > Signed-off-by: Tan Zhongjun <tanzhongjun(a)yulong.com> Acked-by: Shengjiu Wang <shengjiu.wang(a)gmail.com> > --- > sound/soc/fsl/fsl_easrc.c | 4 +--- > 1 file changed, 1 insertion(+), 3 deletions(-) > > diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c > index b1765c7d3bcd..25747433916e 100644 > --- a/sound/soc/fsl/fsl_easrc.c > +++ b/sound/soc/fsl/fsl_easrc.c > @@ -1901,10 +1901,8 @@ static int fsl_easrc_probe(struct platform_device *pdev) > } > > irq = platform_get_irq(pdev, 0); > - if (irq < 0) { > - dev_err(dev, "no irq for node %pOF\n", np); > + if (irq < 0) > return irq; > - } > > ret = devm_request_irq(&pdev->dev, irq, fsl_easrc_isr, 0, > dev_name(dev), easrc); > -- > 2.17.1 > >

1 0

[PATCH] ALSA: seq: Fix race of snd_seq_timer_open()
by Takashi Iwai 10 Jun '21

10 Jun '21

The timer instance per queue is exclusive, and snd_seq_timer_open() should have managed the concurrent accesses. It looks as if it's checking the already existing timer instance at the beginning, but it's not right, because there is no protection, hence any later concurrent call of snd_seq_timer_open() may override the timer instance easily. This may result in UAF, as the leftover timer instance can keep running while the queue itself gets closed, as spotted by syzkaller recently. For avoiding the race, add a proper check at the assignment of tmr->timeri again, and return -EBUSY if it's been already registered. Reported-by: syzbot+ddc1260a83ed1cbf6fb5(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000dce34f05c42f110c@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_timer.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index 1645e4142e30..9863be6fd43e 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -297,8 +297,16 @@ int snd_seq_timer_open(struct snd_seq_queue *q) return err; } spin_lock_irq(&tmr->lock); - tmr->timeri = t; + if (tmr->timeri) + err = -EBUSY; + else + tmr->timeri = t; spin_unlock_irq(&tmr->lock); + if (err < 0) { + snd_timer_close(t); + snd_timer_instance_free(t); + return err; + } return 0; } -- 2.26.2

1 0

pulseaudio can't load configuration for snd_hda_intel
by GitHub issues - edited 10 Jun '21

10 Jun '21

alsa-project/alsa-ucm-conf issue #99 was edited from hzhuang1: In v1.2.2, the content of HDA-Intel.conf is in below. v1.2.2 is widely used, such as in ubuntu 20.04. ``` Syntax 3 Define.Use "" # a non-empty string to use UCM configuration for HDA devices Define.AcpCardId "$${CardIdByName:acp}" ``` With this content, pulseaudio can't load configuration from UCM. Then I tried to edit the content. Change it to ``` Define.Use "2" ``` or ``` Define.Use "3" ``` Pulseaudio worked with "options snd-hda-intel probe_mask=1" on DELL laptop. My question is what's the real meaning of **Define.Use** at here. Why the default value of **Define.Use** is empty string? Even in latest code, the **Define.Use** is also an empty string. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/99 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

pulseaudio can't load configuration for snd_hda_intel
by GitHub issues - opened 10 Jun '21

10 Jun '21

alsa-project/alsa-ucm-conf issue #99 was opened from hzhuang1: In v1.2.2, the content of HDA-Intel.conf is in below. v1.2.2 is widely used, such as in ubuntu 20.04. ``` Syntax 3 Define.Use "" # a non-empty string to use UCM configuration for HDA devices Define.AcpCardId "$${CardIdByName:acp}" ``` With this content, pulseaudio can't load configuration from UCM. Then I tried to edit the content. Change it to ``` Define.Use "2" ``` or ``` Define.Use "3" ``` Pulseaudio worked with "options snd-hda-intel probe_mask=1" on DELL laptop. My question is what's the real meaning of **Define.Use** at here. Why the default value of **Define.Use** is empty string? Even in v1.2.4, the **Define.Use** is also an empty string. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/99 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Logitech C922x microphone intermittently working
by GitHub issues - opened 10 Jun '21

10 Jun '21

alsa-project/alsa-lib issue #152 was opened from sambo57u: I have a Logitech C922X (detected as C922) USB webcam. Running latest updates on Fedora 34. The microphone of this device only works about 1 out of 5 reboots. When not working if I install another microphone, activate that one, the unplug and plug C922 back and switch to it, it starts to work. Some kind of a timing issue that is very annoying. Don't have this problem with Logitech C270. The camera always works fine. Issue URL : https://github.com/alsa-project/alsa-lib/issues/152 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Undeclarated constant
by GitHub issues - opened 10 Jun '21

10 Jun '21

alsa-project/alsa-plugins issue #24 was opened from unknown2045: Hi, I am trying to install the alsa-plugins, when I do the make command I get the following errors: ![Captura de pantalla de 2021-06-10 13-52-37](https://user-images.githubusercontent.com/85682900/121520376-229e… How can I solve it? Thanks in advance! Issue URL : https://github.com/alsa-project/alsa-plugins/issues/24 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[sound:for-next 164/168] include/sound/memalloc.h:63:17: error: use of undeclared identifier 'PAGE_SIZE'
by kernel test robot 10 Jun '21

10 Jun '21

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 83fbcaed24d797528de00c024674fd58e8f1634f commit: 37af81c5998f4b0f23fb452cffa4b8a1c00ce95b [164/168] ALSA: core: Abstract memory alloc helpers config: riscv-randconfig-r002-20210610 (attached as .config) compiler: clang version 13.0.0 (https://github.com/llvm/llvm-project d2012d965d60c3258b3a69d024491698f8aec386) reproduce (this is a W=1 build): wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # install riscv cross compiling tool for clang build # apt-get install binutils-riscv64-linux-gnu # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags sound for-next git checkout 37af81c5998f4b0f23fb452cffa4b8a1c00ce95b # save the attached .config to linux build tree COMPILER_INSTALL_PATH=$HOME/0day COMPILER=clang make.cross ARCH=riscv If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): In file included from sound/core/pcm_iec958.c:8: In file included from include/sound/pcm.h:12: >> include/sound/memalloc.h:63:17: error: use of undeclared identifier 'PAGE_SIZE' return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; ^ >> include/sound/memalloc.h:63:35: error: use of undeclared identifier 'PAGE_SHIFT' return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; ^ 2 errors generated. vim +/PAGE_SIZE +63 include/sound/memalloc.h ^1da177e4c3f41 Linus Torvalds 2005-04-16 57 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 58 /* 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 59 * return the pages matching with the given byte size 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 60 */ 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 61 static inline unsigned int snd_sgbuf_aligned_pages(size_t size) 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 62 { 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 @63 return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 64 } 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 65 :::::: The code at line 63 was first introduced by commit :::::: 4cae99d9b5305ab8cccc839fccceb81ec9e5abda ALSA: memalloc: declare snd_sgbuf_aligned_pages() unconditionally :::::: TO: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

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[PATCH] ALSA: core: Fix build error due to missing PAGE_SIZE
by Takashi Iwai 10 Jun '21

10 Jun '21

The recent refactoring of memalloc stuff removed the inclusion of asm/page.h for simplicity, but it turned out this caused a compile error due the lack of PAGE_SIZE definition on some architectures. Do a partial revert for recovering from that. Fixes: 37af81c5998f ("ALSA: core: Abstract memory alloc helpers") Reported-by: kernel test robot <lkp(a)intel.com> Link: https://lore.kernel.org/r/202106101858.PfXMMuAa-lkp@intel.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/memalloc.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 1918c60f1f35..44d87775b352 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -9,8 +9,9 @@ #ifndef __SOUND_MEMALLOC_H #define __SOUND_MEMALLOC_H +#include <asm/page.h> + struct device; -struct page; struct vm_area_struct; /* -- 2.26.2

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[sound:for-next 164/167] include/sound/memalloc.h:63:17: error: 'PAGE_SIZE' undeclared
by kernel test robot 10 Jun '21

10 Jun '21

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 7fdabab8f56239466275adea5d2f0a6fff27527b commit: 37af81c5998f4b0f23fb452cffa4b8a1c00ce95b [164/167] ALSA: core: Abstract memory alloc helpers config: h8300-randconfig-r006-20210610 (attached as .config) compiler: h8300-linux-gcc (GCC) 9.3.0 reproduce (this is a W=1 build): wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags sound for-next git checkout 37af81c5998f4b0f23fb452cffa4b8a1c00ce95b # save the attached .config to linux build tree COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-9.3.0 make.cross ARCH=h8300 If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): In file included from include/sound/pcm.h:12, from sound/core/pcm_iec958.c:8: include/sound/memalloc.h: In function 'snd_sgbuf_aligned_pages': >> include/sound/memalloc.h:63:17: error: 'PAGE_SIZE' undeclared (first use in this function) 63 | return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; | ^~~~~~~~~ include/sound/memalloc.h:63:17: note: each undeclared identifier is reported only once for each function it appears in >> include/sound/memalloc.h:63:35: error: 'PAGE_SHIFT' undeclared (first use in this function) 63 | return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; | ^~~~~~~~~~ vim +/PAGE_SIZE +63 include/sound/memalloc.h ^1da177e4c3f41 Linus Torvalds 2005-04-16 57 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 58 /* 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 59 * return the pages matching with the given byte size 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 60 */ 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 61 static inline unsigned int snd_sgbuf_aligned_pages(size_t size) 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 62 { 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 @63 return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 64 } 4cae99d9b5305a Pierre-Louis Bossart 2018-07-25 65 :::::: The code at line 63 was first introduced by commit :::::: 4cae99d9b5305ab8cccc839fccceb81ec9e5abda ALSA: memalloc: declare snd_sgbuf_aligned_pages() unconditionally :::::: TO: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

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[PATCH 0/1 v2] usb-audio: Add support for Denon DN-X1600
by Damien Zammit 10 Jun '21

10 Jun '21

Hi Takashi, You are correct, the patch still works without the hunk referring to the PITCH control. I have removed the hunk and was tested with this patch and it still functioned as expected. Regards, Damien Z

2 2

[PATCH] Docs: hda: Fix spelling mistakes
by huangjianghui 10 Jun '21

10 Jun '21

correct Reatek to Realtek Signed-off-by: huangjianghui <huangjianghui(a)uniontech.com> --- Documentation/sound/hd-audio/controls.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/hd-audio/controls.rst b/Documentation/sound/hd-audio/controls.rst index f2ebc4f79b44..dbe6483f4ff4 100644 --- a/Documentation/sound/hd-audio/controls.rst +++ b/Documentation/sound/hd-audio/controls.rst @@ -102,7 +102,7 @@ Conexant codecs --------------- Auto-Mute Mode - See Reatek codecs. + See Realtek codecs. Analog codecs -- 2.20.1

2 1

[PATCH] sound/arm: Remove unnecessary variables
by zuoqilin1＠163.com 10 Jun '21

10 Jun '21

From: zuoqilin <zuoqilin(a)yulong.com> There is no need to define the variable "ret" to receive. Signed-off-by: zuoqilin <zuoqilin(a)yulong.com> --- sound/arm/pxa2xx-ac97.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 6322e639..a67e668 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -47,9 +47,7 @@ static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97, static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { - int __always_unused ret; - - ret = pxa2xx_ac97_write(ac97->num, reg, val); + pxa2xx_ac97_write(ac97->num, reg, val); } static const struct snd_ac97_bus_ops pxa2xx_ac97_ops = { -- 1.9.1

2 1

[PATCH v3 0/3] ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream
by Takashi Sakamoto 10 Jun '21

10 Jun '21

Hi, This patchset is revised version of my previous one: * https://lore.kernel.org/alsa-devel/20210609143145.146680-1-o-takashi@sakamo… All of drivers in ALSA firewire stack have two chances to process isochronous packets of any isochronous context; in software IRQ context for 1394 OHCI, and in process context of ALSA PCM application. In the process context, callbacks of .pointer and .ack are utilized. The callbacks are done by ALSA PCM core under acquiring lock of PCM substream, In design of ALSA PCM core, call of snd_pcm_period_elapsed() is used for drivers to awaken user processes from waiting for available frames. The function voluntarily acquires lock of PCM substream, therefore it is not called in the process context since it causes dead lock. As a workaround to avoid the dead lock, all of drivers in ALSA firewire stack use workqueue to delegate the call. This patchset is my attempt for the issue. A variant of 'snd_pcm_period_elapsed()' without lock acquisition is going to be added, named 'snd_pcm_period_elapsed_under_stream_lock()'. The call is available in callbacks of .pointer and .ack of snd_pcm_ops structure. Changes from v2: * delete context section from kernel API documentation Takashi Sakamoto (3): ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream ALSA: firewire-lib: operate for period elapse event in process context ALSA: firewire-lib: obsolete workqueue for period update include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 64 +++++++++++++++++++++++++++-------- sound/firewire/amdtp-stream.c | 46 ++++++++----------------- sound/firewire/amdtp-stream.h | 1 - 4 files changed, 64 insertions(+), 48 deletions(-) -- 2.27.0

2 4

[Bug] Duplication in Array Element not Handled in Parsing conf
by GitHub issues - opened 10 Jun '21

10 Jun '21

alsa-project/alsa-lib issue #151 was opened from aiChaoSONG: When parsing array element in conf files, an auto-increase number is used as config id(`snd_config_t->id`), and array element as the config value(`snd_config_t->u.string/integer/integer64/real`), duplication is not handle in `parse_array_defs/parse_array_def/parse_value` functions. Suppose we have a conf: ``` array [ "a" "b" "a" ] ``` the parsed result will be: ``` parent | ---- snd_config_t {id: "0", u.string: "a"} | ---- snd_config_t {id: "1", u.string: "b"} | ---- snd_config_t {id: "2", u.string: "a"} ``` The duplication in "a" may be desired. But please consider this condition: suppose we have four conf files A, B, C and D, where A is the conf above, let B include A, C include A, D include B and C, so D include A twice indirectly. File A will be processed twice when parsing D, because duplication is not handled, here comes the problem, the parsed result of A in D is: ``` parent | ---- snd_config_t {id: "0", u.string: "a"} | ---- snd_config_t {id: "1", u.string: "b"} | ---- snd_config_t {id: "2", u.string: "a"} | ---- snd_config_t {id: "3", u.string: "a"} | ---- snd_config_t {id: "4", u.string: "b"} | ---- snd_config_t {id: "5", u.string: "a"} ``` This is obviously wrong. I think handle duplication in parse_value is not the solution, because element duplication in an array is wanted. The key problem here is that we parse the same conf more than once. Issue URL : https://github.com/alsa-project/alsa-lib/issues/151 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[bug report] ASoC: rsnd: add null CLOCKIN support
by Dan Carpenter 10 Jun '21

10 Jun '21

Hello Kuninori Morimoto, The patch d6956a7dde6f: "ASoC: rsnd: add null CLOCKIN support" from May 24, 2021, leads to the following static checker warning: sound/soc/sh/rcar/adg.c:435 rsnd_adg_get_clkin() warn: 'clk' isn't an ERR_PTR sound/soc/sh/rcar/adg.c 423 static int rsnd_adg_get_clkin(struct rsnd_priv *priv) 424 { 425 struct rsnd_adg *adg = priv->adg; 426 struct device *dev = rsnd_priv_to_dev(priv); 427 struct clk *clk; 428 int i; 429 430 for (i = 0; i < CLKMAX; i++) { 431 clk = devm_clk_get(dev, clk_name[i]); If devm_clk_get() returns NULL that's not an error. 432 433 if (IS_ERR(clk)) 434 clk = rsnd_adg_null_clk_get(priv); But I think if rsnd_adg_null_clk_get() returns an error then it is an error. 435 if (IS_ERR(clk)) This is impossible. 436 goto err; So I think it should be: if (IS_ERR(clk)) { clk = rsnd_adg_null_clk_get(priv); if (!clk) goto err; } 437 438 adg->clk[i] = clk; 439 } 440 441 return 0; 442 443 err: 444 dev_err(dev, "adg clock IN get failed\n"); 445 446 rsnd_adg_null_clk_clean(priv); 447 448 return -EIO; 449 } regards, dan carpenter

2 3

[PATCH v3 0/3] ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream
by Takashi Sakamoto 10 Jun '21

10 Jun '21

Hi, This patchset is revised version of my previous one: * https://lore.kernel.org/alsa-devel/20210609143145.146680-1-o-takashi@sakamo… All of drivers in ALSA firewire stack have two chances to process isochronous packets of any isochronous context; in software IRQ context for 1394 OHCI, and in process context of ALSA PCM application. In the process context, callbacks of .pointer and .ack are utilized. The callbacks are done by ALSA PCM core under acquiring lock of PCM substream, In design of ALSA PCM core, call of snd_pcm_period_elapsed() is used for drivers to awaken user processes from waiting for available frames. The function voluntarily acquires lock of PCM substream, therefore it is not called in the process context since it causes dead lock. As a workaround to avoid the dead lock, all of drivers in ALSA firewire stack use workqueue to delegate the call. This patchset is my attempt for the issue. A variant of 'snd_pcm_period_elapsed()' without lock acquisition is going to be added, named 'snd_pcm_period_elapsed_under_stream_lock()'. The call is available in callbacks of .pointer and .ack of snd_pcm_ops structure. Changes from v2: * delete context section from kernel API documentation Takashi Sakamoto (3): ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream ALSA: firewire-lib: operate for period elapse event in process context ALSA: firewire-lib: obsolete workqueue for period update include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 64 +++++++++++++++++++++++++++-------- sound/firewire/amdtp-stream.c | 46 ++++++++----------------- sound/firewire/amdtp-stream.h | 1 - 4 files changed, 64 insertions(+), 48 deletions(-) -- 2.27.0

1 1

aplay change to use 16-bit samples by default is not backwards compatible and is not documented
by GitHub issues - opened 10 Jun '21

10 Jun '21

alsa-project/alsa-utils issue #96 was opened from nwoeanhinnogaehr: The change introduced in https://github.com/alsa-project/alsa-utils/commit/0c5948e98a6a8535c89b7bcab… breaks a lot of scripts and other programs I have which invoke `aplay` and expect the default sample format to be U8. While I think it's somewhat reasonable to make this change given that U8 is a rather low quality sample format, it's worth acknowledging that aplay has had this behavior for a very long time and many people probably don't expect it to change. Given that the commit log seems to only talk about the case of `arecord`, I wonder if we could reach a compromise and keep the format as U8 for `aplay` and switch it for `arecord`, since users are unlikely to complain about poor sound quality when *playing raw audio* (surely they would take the time to get the sample format set up correctly before attempting to play raw audio). Also, the man page has not been updated to reflect this change and still says that U8 is the default. Issue URL : https://github.com/alsa-project/alsa-utils/issues/96 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH v2 0/8] ASoC: tidyup snd_soc_of_parse_daifmt()
by Kuninori Morimoto 10 Jun '21

10 Jun '21

Hi Mark These are v2 of parsing for daifmt. I want to add new audio-graph-card2 sound card driver, and this is last part of necessary soc-core cleanup for it. Current some drivers are using DT, and then, snd_soc_of_parse_daifmt() parses daifmt, but bitclock/frame provider parsing part is one of headache, because we are assuming below both cases. A) node { bitclock-master; frame-master; ... }; B) link { bitclock-master = <&xxx>; frame-master = <&xxx>; ... }; The original was style A), and style B) was added later. snd_soc_of_parse_daifmt() parses A) style as original style, and user need to update to B) style for clock_provider part if needed. To handle it more flexibile, this patch-set adds new functions which separates snd_soc_of_parse_daifmt() helper function. snd_soc_daifmt_parse_format() : format part snd_soc_daifmt_parse_clock_provider_as_flag() : clock part for style A) snd_soc_daifmt_parse_clock_provider_as_phandl() : clock part for style B) snd_soc_daifmt_parse_clock_provider_as_bitmap() : clock part use with _from_bitmap v1 -> v2 - tidyup parse_clock_provider functions to _as_flag/phandle/bitmap() - don't exchange code style on each drivers. Link: https://lore.kernel.org/r/875yypdxlm.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (8): ASoC: soc-core: add snd_soc_daifmt_clock_provider_from_bitmap() ASoC: soc-core: add snd_soc_daifmt_clock_provider_fliped() ASoC: soc-core: add snd_soc_daifmt_parse_format/clock_provider() ASoC: atmel: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: fsl: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: meson: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: simple-card-utils: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: soc-core: remove snd_soc_of_parse_daifmt() include/sound/soc.h | 21 ++++-- sound/soc/atmel/mikroe-proto.c | 16 +++-- sound/soc/fsl/fsl-asoc-card.c | 14 ++-- sound/soc/generic/simple-card-utils.c | 19 ++--- sound/soc/meson/meson-card-utils.c | 13 ++-- sound/soc/soc-core.c | 99 +++++++++++++++++---------- 6 files changed, 110 insertions(+), 72 deletions(-) -- 2.25.1

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[PATCH 0/5] ALSA: More abstracition of memory alloc helpers
by Takashi Iwai 09 Jun '21

09 Jun '21

Hi, this is a patch set for the code cleanup and refactoring around the ALSA core memory allocation helpers. It begins with a cleanup of the non-used code, then followed by the refacotoring with the new ops, finally moved the memory-specific mmap handling code from PCM to the core memalloc. Takashi === Takashi Iwai (5): ALSA: trident: Drop shadow TLB pointer table ALSA: core: Drop snd_sgbuf_get_ptr() ALSA: core: Abstract memory alloc helpers ALSA: core: Move mmap handler into memalloc ops ALSA: core: Add continuous and vmalloc mmap ops include/sound/memalloc.h | 83 +----- include/sound/pcm.h | 19 -- sound/core/memalloc.c | 444 +++++++++++++++++++---------- sound/core/memalloc_local.h | 19 ++ sound/core/pcm_local.h | 5 - sound/core/pcm_memory.c | 21 -- sound/core/pcm_native.c | 39 +-- sound/core/sgbuf.c | 90 ++++-- sound/pci/trident/trident.h | 1 - sound/pci/trident/trident_main.c | 11 +- sound/pci/trident/trident_memory.c | 53 ++-- 11 files changed, 405 insertions(+), 380 deletions(-) create mode 100644 sound/core/memalloc_local.h -- 2.26.2

1 5

Current version not working with Focusrite Scarlett 18i20
by GitHub issues - opened 09 Jun '21

09 Jun '21

alsa-project/alsa-lib issue #150 was opened from TheBitFighter: Hello, I hope I am in the right repository for this. My arch system updated alsa-card-profiles and alsa-utils today which caused my interface, a Focusrite Scarlett 18i20 1st gen to stop working. The card was still showing up in pulse and showed output, however no sound was actually played. Input was still working fine. It sounded like the output stage of the interface had been disabled, and the problem could only be fixed by booting Windows and setting new routing settings in Scarlett MixControl. The interface had exhibited some resetting and popping behavior for a few versions now (when stopping playback, changing application and playing something else there), but continued to work (given a few instances where sample rate played mismatched sample rate sent which caused weird audio slowdowns). It only stopped working today. What I did initially was reinstall the older versions of these two packages, which did not fix the problem. I now downgraded alsa-libs (1.2.5-2 -> 1.2.4-2), alsa-card-profiles (1:0.3.29-2 -> 1:0.3.28-1), lib32-alsa-lib (1.2.5-2 -> 1.2.4-1), alsa-topology-conf (1.2.5-1 -> 1.2.4-1), and alsa-ucm-conf (1.2.5-1 -> 1.2.4-1) which fixed the problem. (alsa-utils is at 1.2.5-2) `cat /etc/asound/version` now reports `Advanced Linux Sound Architecture Driver Version k5.12.9-arch1-1.` This also fixed the popping/resetting issues I had before. Thanks for the help! If you need any further information I am happy to help. Issue URL : https://github.com/alsa-project/alsa-lib/issues/150 Repository URL: https://github.com/alsa-project/alsa-lib

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opensuse tumbleweed no sound after alsa lib update (tumbleweed 2010605)
by GitHub issues - opened 09 Jun '21

09 Jun '21

alsa-project/alsa-lib issue #149 was opened from pawanyadav: sound is not working alsa (never installed pipewire and pulseaudio) asound.conf defaults.pcm.card 1 defaults.pcm.device 1 aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Generic [HD-Audio Generic], device 0: ALC887-VD Analog [ALC887-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Generic [HD-Audio Generic], device 1: ALC887-VD Digital [ALC887-VD Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 cat /proc/asound/cards 0 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfcf40000 irq 67 1 [Generic ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xfce00000 irq 69 speaker-test 1.2.4 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise ALSA lib confmisc.c:855:(parse_card) cannot find card '1dmix:1' ALSA lib conf.c:5091:(_snd_config_evaluate) function snd_func_card_id returned error: No such device ALSA lib confmisc.c:422:(snd_func_concat) error evaluating strings ALSA lib conf.c:5091:(_snd_config_evaluate) function snd_func_concat returned error: No such device ALSA lib confmisc.c:1334:(snd_func_refer) error evaluating name ALSA lib conf.c:5091:(_snd_config_evaluate) function snd_func_refer returned error: No such device ALSA lib conf.c:5579:(snd_config_expand) Evaluate error: No such device ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM dmix:1dmix:1 Playback open error: -19,No such device Issue URL : https://github.com/alsa-project/alsa-lib/issues/149 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: qcom: Add four speaker support on MI2S secondary
by Srinivasa Rao Mandadapu 09 Jun '21

09 Jun '21

Add four speaker support on MI2S secondary block by using I2S SD1 line on gpio52 pin, and add channel map control support in the lpass-cpu audio driver. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes Since V1: -- removed set_channel_map/get_channel_map implementation as default kcontrols added in pcm_new API. sound/soc/qcom/lpass-cpu.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/qcom/lpass-sc7180.c | 1 + sound/soc/qcom/lpass.h | 2 ++ 3 files changed, 35 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c62d2612e8f5..15d69e311b38 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -29,6 +29,14 @@ #define LPASS_CPU_I2S_SD0_1_2_MASK GENMASK(2, 0) #define LPASS_CPU_I2S_SD0_1_2_3_MASK GENMASK(3, 0) +/* + * Channel maps for Quad channel playbacks on MI2S Secondary + */ +static struct snd_pcm_chmap_elem lpass_quad_chmaps = { + .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_RL, + SNDRV_CHMAP_FR, SNDRV_CHMAP_RR } +}; static int lpass_cpu_init_i2sctl_bitfields(struct device *dev, struct lpaif_i2sctl *i2sctl, struct regmap *map) { @@ -324,6 +332,25 @@ const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int ret; + struct snd_soc_dai_driver *drv = dai->driver; + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + + if (drvdata->mi2s_playback_sd_mode[dai->id] == LPAIF_I2SCTL_MODE_QUAD01) { + ret = snd_pcm_add_chmap_ctls(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, + &lpass_quad_chmaps, drv->playback.channels_max, 0, + NULL); + if (ret < 0) + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(lpass_cpu_pcm_new); + int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); @@ -856,6 +883,11 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) PTR_ERR(drvdata->mi2s_bit_clk[dai_id])); return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]); } + if (drvdata->mi2s_playback_sd_mode[dai_id] == + LPAIF_I2SCTL_MODE_QUAD01) { + variant->dai_driver[dai_id].playback.channels_min = 4; + variant->dai_driver[dai_id].playback.channels_max = 4; + } } /* Allocation for i2sctl regmap fields */ diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 8c168d3c589e..77a556b27cf0 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -58,6 +58,7 @@ static struct snd_soc_dai_driver sc7180_lpass_cpu_dai_driver[] = { }, .probe = &asoc_qcom_lpass_cpu_dai_probe, .ops = &asoc_qcom_lpass_cpu_dai_ops, + .pcm_new = lpass_cpu_pcm_new, }, { .id = LPASS_DP_RX, .name = "Hdmi", diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..623ddccdafff 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -259,5 +259,7 @@ void asoc_qcom_lpass_cpu_platform_shutdown(struct platform_device *pdev); int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev); int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai); extern const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops; +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai); #endif /* __LPASS_H__ */ -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 2

[PATCH] usb-audio: Add support for Denon DN-X1600
by Damien Zammit 09 Jun '21

09 Jun '21

This patch applies clean to tiwai/sound.git repo on hash 82b954a96c1550bae2cb1f1f4f084425ab237cd7 and was tested on this version. This provides support for Denon DN-X1600 hardware mixer. The device itself supports 44100, 48000 and 96000 (Hz) sample rates, but switching rates via software is currently not working. Therefore, this patch hardcodes the sample rate to 48000Hz which enables all 8 channels to function correctly when the correct sample rate is selected on the hardware itself. MIDI also tested and works. Signed-off-by: Damien Zammit <damien(a)zamaudio.com> Tested-by: xalmoxis(a)gmail.com --- sound/usb/clock.c | 7 +++++ sound/usb/pcm.c | 4 +++ sound/usb/quirks-table.h | 70 ++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 81 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 0afae839d..52de52288 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -534,6 +534,13 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, * rate. */ clock = snd_usb_clock_find_source(chip, fmt, false); + + /* Denon DN-X1600 hardcoded + * Sample rate seems to be set on the hardware itself + */ + if (chip->usb_id == USB_ID(0x154e, 0x500e)) + return 0; + if (clock < 0) return clock; } diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c66831ee1..6913a5b02 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -197,6 +197,10 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, if (!(fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL)) return 0; + /* Denon DN-X1600 doesn't properly handle this, bail out too */ + if (chip->usb_id == USB_ID(0x154e, 0x500e)) + return 0; + usb_audio_dbg(chip, "enable PITCH for EP 0x%x\n", fmt->endpoint); switch (fmt->protocol) { diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index bdba37d0f..19bb499c1 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3044,6 +3044,76 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +/* Denon DN-X1600 */ +{ + USB_AUDIO_DEVICE(0x154e, 0x500e), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Denon", + .product_name = "DN-X1600", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]){ + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0x0, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ADAPTIVE, + .maxpacksize = 0x138, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0x0, + .endpoint = 0x85, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ADAPTIVE, + .maxpacksize = 0x138, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 4, + .type = QUIRK_MIDI_STANDARD_INTERFACE, + }, + { + .ifnum = -1 + } + } + } +}, + /* Microsoft XboxLive Headset/Xbox Communicator */ { USB_DEVICE(0x045e, 0x0283), -- 2.13.1

2 1

[PATCH 0/3] ALSA: pcm:firewire: allow to operate for period elapse event in process context
by Takashi Sakamoto 09 Jun '21

09 Jun '21

Hi, This patchset comes from my former RFC: [RFC][PATCH 0/3] ALSA: pcm/firewire: allow to queue period elapse event in process context * https://lore.kernel.org/alsa-devel/20210606091838.80812-1-o-takashi@sakamoc… All of drivers in ALSA firewire stack have two chances to process isochronous packets of any isochronous context; in software IRQ context for 1394 OHCI, and in process context of ALSA PCM application. In the process context, callbacks of .pointer and .ack are utilized. The callbacks are done by ALSA PCM core under acquiring lock of PCM substream, In design of ALSA PCM core, call of snd_pcm_period_elapsed() is used for drivers to awaken user processes from waiting for available frames. The function voluntarily acquires lock of PCM substream, therefore it is not called in the process context since it causes dead lock. As a workaround to avoid the dead lock, all of drivers in ALSA firewire stack use workqueue to delegate the call. This patchset is my attempt for the issue. A variant of 'snd_pcm_period_elapsed()' without lock acquisition is going to be added, named 'snd_pcm_period_elapsed_under_stream_lock()'. The call is available in callbacks of .pointer and .ack of snd_pcm_ops structure. Changes from RFC: * dismiss inlining * rename function name * improve function comments I tested the patchset with ALSA OXFW driver and ftrace, covering axfer and jackd cases since pulseaudio and pipewire are programmed with timer-based scheduling model and ALSA runtime expects drivers not to call snd_pcm_period_elapsed(). The configuration of tracer and filters is: ``` $ sudo trace-cmd record -p function_graph \ -l :mod:snd_firewire_lib \ -l fw_iso_context_queue:mod:firewire_core \ -l snd_pcm_ioctl:mod:snd_pcm \ -l snd_pcm_period_elapsed*:mod:snd_pcm ``` The runtime of axfer without '-M' option can often call .pointer and .ack. Below is a sample about .pointer case via SNDRV_PCM_IOCT_HWSYNC. The new snd_pcm_period_elapsed_under_stream_lock() should be called. ``` 3929.769359: funcgraph_entry: | snd_pcm_ioctl() { 3929.769360: funcgraph_entry: | amdtp_domain_stream_pcm_pointer() { 3929.769361: funcgraph_entry: | irq_target_callback() { 3929.769361: funcgraph_entry: | process_rx_packets() { 3929.769362: funcgraph_entry: | process_ctx_payloads() { 3929.769363: funcgraph_entry: 0.391 us | process_it_ctx_payloads(); 3929.769363: funcgraph_entry: | snd_pcm_period_elapsed_under_stream_lock() { 3929.769364: funcgraph_entry: 0.411 us | amdtp_domain_stream_pcm_pointer(); 3929.769365: funcgraph_exit: 1.723 us | } 3929.769365: funcgraph_exit: 3.106 us | } 3929.769366: funcgraph_entry: | queue_packet() { 3929.769366: funcgraph_entry: 0.561 us | fw_iso_context_queue(); 3929.769367: funcgraph_exit: 1.122 us | } 3929.769367: funcgraph_exit: 5.731 us | } 3929.769367: funcgraph_entry: | process_ctxs_in_domain() { 3929.769368: funcgraph_entry: | process_tx_packets() { 3929.769369: funcgraph_entry: | process_ctx_payloads() { 3929.769369: funcgraph_entry: 0.321 us | process_ir_ctx_payloads(); 3929.769369: funcgraph_exit: 0.962 us | } 3929.769370: funcgraph_entry: 0.491 us | fw_iso_context_queue(); 3929.769371: funcgraph_exit: 2.364 us | } 3929.769371: funcgraph_exit: 3.427 us | } 3929.769371: funcgraph_exit: 10.038 us | } 3929.769372: funcgraph_exit: 11.271 us | } 3929.769372: funcgraph_exit: 13.606 us | } ``` The runtime of jackd heavily relies on period wakeup scheduled in invocation of interrupt handler. Below is a sample about software IRQ case of 1394 OHCI. The snd_pcm_period_elapsed() should be called. ``` 5318.980502: funcgraph_entry: | irq_target_callback() { 5318.980503: funcgraph_entry: | process_rx_packets() { 5318.980503: funcgraph_entry: | process_ctx_payloads() { 5318.980503: funcgraph_entry: 0.531 us | process_it_ctx_payloads(); 5318.980505: funcgraph_entry: | snd_pcm_period_elapsed() { 5318.980505: funcgraph_entry: | snd_pcm_period_elapsed_under_stream_lock() { 5318.980505: funcgraph_entry: 0.180 us | amdtp_domain_stream_pcm_pointer(); 5318.980508: funcgraph_exit: 3.106 us | } 5318.980508: funcgraph_exit: 3.497 us | } 5318.980508: funcgraph_exit: 4.990 us | } 5318.980508: funcgraph_entry: | queue_packet() { 5318.980509: funcgraph_entry: 0.350 us | fw_iso_context_queue(); 5318.980509: funcgraph_exit: 0.702 us | } 5318.980509: funcgraph_entry: | queue_packet() { 5318.980509: funcgraph_entry: 0.291 us | fw_iso_context_queue(); 5318.980510: funcgraph_exit: 0.651 us | } 5318.980510: funcgraph_entry: | queue_packet() { 5318.980510: funcgraph_entry: 0.280 us | fw_iso_context_queue(); 5318.980511: funcgraph_exit: 0.641 us | } 5318.980511: funcgraph_exit: 8.416 us | } 5318.980511: funcgraph_entry: | process_ctxs_in_domain() { 5318.980512: funcgraph_entry: | process_tx_packets() { 5318.980512: funcgraph_entry: | process_ctx_payloads() { 5318.980512: funcgraph_entry: 0.531 us | process_ir_ctx_payloads(); 5318.980513: funcgraph_entry: | snd_pcm_period_elapsed() { 5318.980513: funcgraph_entry: | snd_pcm_period_elapsed_under_stream_lock() { 5318.980513: funcgraph_entry: 0.170 us | amdtp_domain_stream_pcm_pointer(); 5318.980514: funcgraph_exit: 0.842 us | } 5318.980514: funcgraph_exit: 1.242 us | } 5318.980514: funcgraph_exit: 2.335 us | } 5318.980515: funcgraph_entry: 0.301 us | fw_iso_context_queue(); 5318.980515: funcgraph_entry: 0.291 us | fw_iso_context_queue(); 5318.980516: funcgraph_entry: 0.290 us | fw_iso_context_queue(); 5318.980516: funcgraph_exit: 4.198 us | } 5318.980516: funcgraph_exit: 5.119 us | } 5318.980516: funcgraph_exit: 14.077 us | } ``` ALSA OXFW driver works well to select appropriate kernel API by distinguishing running context. I also tested for error cases that the sequence of tx packets is invalid or packet queueing fails. In both cases, the error doesn't bring system corruption. Takashi Sakamoto (3): ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream ALSA: firewire-lib: operate for period elapse event in process context ALSA: firewire-lib: obsolete workqueue for period update include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 68 +++++++++++++++++++++++++++-------- sound/firewire/amdtp-stream.c | 46 ++++++++---------------- sound/firewire/amdtp-stream.h | 1 - 4 files changed, 68 insertions(+), 48 deletions(-) -- 2.27.0

2 5

[PATCH] Asoc: qcom: Add four speaker support on MI2S secondary
by Srinivasa Rao Mandadapu 09 Jun '21

09 Jun '21

Add four speaker support on MI2S secondary block by using I2S SD1 line on gpio52 pin, and add channel map control support in the lpass-cpu audio driver. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-cpu.c | 106 +++++++++++++++++++++++++++++++++- sound/soc/qcom/lpass-sc7180.c | 1 + sound/soc/qcom/lpass.h | 5 ++ 3 files changed, 111 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c62d2612e8f5..88a038a107a7 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -28,6 +28,19 @@ #define LPASS_CPU_I2S_SD2_3_MASK GENMASK(3, 2) #define LPASS_CPU_I2S_SD0_1_2_MASK GENMASK(2, 0) #define LPASS_CPU_I2S_SD0_1_2_3_MASK GENMASK(3, 0) +#define LPASS_CHMAP_IDX_UNKNOWN -1 + +/* + * Channel maps for multi-channel playbacks on MI2S Secondary, up to 4 n_ch + */ +static struct snd_pcm_chmap_elem lpass_chmaps[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { } +}; static int lpass_cpu_init_i2sctl_bitfields(struct device *dev, struct lpaif_i2sctl *i2sctl, struct regmap *map) @@ -86,6 +99,9 @@ static int lpass_cpu_daiops_startup(struct snd_pcm_substream *substream, clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); return ret; } + if (drvdata->chmap_info && drvdata->chmap_info->max_channels == 4) + drvdata->chmap_info->chmap = lpass_chmaps; + return 0; } @@ -96,6 +112,7 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); + drvdata->chmap_idx = LPASS_CHMAP_IDX_UNKNOWN; } static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, @@ -224,9 +241,14 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, ret); return ret; } - if (channels >= 2) + if (channels >= 2) { ret = regmap_fields_write(i2sctl->spkmono, id, LPAIF_I2SCTL_SPKMONO_STEREO); + if (channels == 4) + drvdata->chmap_idx = 1; + else + drvdata->chmap_idx = 0; + } else ret = regmap_fields_write(i2sctl->spkmono, id, LPAIF_I2SCTL_SPKMONO_MONO); @@ -324,6 +346,84 @@ const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); +static int lpass_cpu_chmap_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned const char *map; + unsigned int i; + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct lpass_data *drvdata = info->private_data; + + if (info->max_channels == 2) + map = &lpass_chmaps[0].map[0]; + else if (info->max_channels == 4) + map = &lpass_chmaps[1].map[0]; + + for (i = 0; i < info->max_channels; i++) { + if (drvdata->chmap_idx == LPASS_CHMAP_IDX_UNKNOWN) + ucontrol->value.integer.value[i] = 0; + else + ucontrol->value.integer.value[i] = map[i]; + } + + return 0; +} + +static int lpass_cpu_chmap_ctl_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned int i; + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct lpass_data *drvdata = info->private_data; + unsigned char *map; + + if (info->max_channels == 2) + map = &lpass_chmaps[0].map[0]; + else if (info->max_channels == 4) + map = &lpass_chmaps[1].map[0]; + + for (i = 0; i < info->max_channels; i++) { + if (drvdata->chmap_idx == LPASS_CHMAP_IDX_UNKNOWN) + map[i] = 0; + else + map[i] = ucontrol->value.integer.value[i]; + } + + return 0; +} + +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int ret; + + struct snd_soc_dai_driver *drv = dai->driver; + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + + ret = snd_pcm_add_chmap_ctls(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_alt_chmaps, drv->playback.channels_max, 0, + &drvdata->chmap_info); + if (ret < 0) + return ret; + + + /* + * override handlers + */ + drvdata->chmap_info->private_data = drvdata; + drvdata->chmap_info->kctl->get = lpass_cpu_chmap_ctl_get; + drvdata->chmap_info->kctl->put = lpass_cpu_chmap_ctl_set; + + /* + * default chmap supported is stereo + */ + drvdata->chmap_info->chmap = lpass_chmaps; + drvdata->chmap_idx = LPASS_CHMAP_IDX_UNKNOWN; + + return 0; +} +EXPORT_SYMBOL_GPL(lpass_cpu_pcm_new); + int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); @@ -856,6 +956,10 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) PTR_ERR(drvdata->mi2s_bit_clk[dai_id])); return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]); } + if (drvdata->mi2s_playback_sd_mode[dai_id] == LPAIF_I2SCTL_MODE_QUAD01) { + variant->dai_driver[dai_id].playback.channels_min = 4; + variant->dai_driver[dai_id].playback.channels_max = 4; + } } /* Allocation for i2sctl regmap fields */ diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 8c168d3c589e..77a556b27cf0 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -58,6 +58,7 @@ static struct snd_soc_dai_driver sc7180_lpass_cpu_dai_driver[] = { }, .probe = &asoc_qcom_lpass_cpu_dai_probe, .ops = &asoc_qcom_lpass_cpu_dai_ops, + .pcm_new = lpass_cpu_pcm_new, }, { .id = LPASS_DP_RX, .name = "Hdmi", diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..fb82591d550c 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -76,6 +76,9 @@ struct lpass_data { /* regmap backed by the low-power audio interface (LPAIF) registers */ struct regmap *lpaif_map; struct regmap *hdmiif_map; + struct snd_pcm_chmap *chmap_info; + + int chmap_idx; /* interrupts from the low-power audio interface (LPAIF) */ int lpaif_irq; @@ -259,5 +262,7 @@ void asoc_qcom_lpass_cpu_platform_shutdown(struct platform_device *pdev); int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev); int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai); extern const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops; +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai); #endif /* __LPASS_H__ */ -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 2

[PATCH 0/9] ASoC: tidyup snd_soc_of_parse_daifmt()
by Kuninori Morimoto 09 Jun '21

09 Jun '21

Hi Mark I want to add new audio-graph-card2 sound card driver, and this is last part of necessary soc-core cleanup for it. Current some drivers are using DT, and Then, snd_soc_of_parse_daifmt() parses daifmt, but bitclock/frame provider parsing part is one of headache, because we are assuming below both cases. A) node { bitclock-master; frame-master; ... }; B) link { bitclock-master = <&xxx>; frame-master = <&xxx>; ... }; The original was style A), and style B) was added later. snd_soc_of_parse_daifmt() parses A) style as original style, and user need to update to B) style for clock_provider part if needed. In such case, user need to re-parse it, like below. daifmt = snd_soc_of_parse_daifmt(..., &bitclkmaster, &framemaster); daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; if (codec == bitclkmaster) daifmt |= (codec == framemaster) ? SND_SOC_DAIFMT_CBM_CFM : SND_SOC_DAIFMT_CBM_CFS; else daifmt |= (codec == framemaster) ? SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS; This patch-set adds new functions, and handle these more simply. Unfortunately, there are too many use-case, do it by 1 function was implessible. style A) bit_frame = snd_soc_daifmt_parse_clock_provider(); daifmt = snd_soc_daifmt_parse_format(...) | /* format part */ snd_soc_daifmt_clock_provider_pickup(bit_frame); /* clock part */ style B) snd_soc_daifmt_parse_clock_provider(..., &bit, &frame); daifmt = snd_soc_daifmt_parse_format(...) | /* format part */ snd_soc_daifmt_clock_provider_pickup( /* clock part */ ((codec == bit) << 4) + (codec == frame)); Kuninori Morimoto (9): ASoC: soc-core: don't use discriminatory terms on snd_soc_runtime_get_dai_fmt() ASoC: soc-core: add snd_soc_daifmt_clock_provider_pickup() ASoC: soc-core: add snd_soc_daifmt_clock_provider_fliped() ASoC: soc-core: add snd_soc_daifmt_parse_format/clock_provider() ASoC: atmel: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: fsl: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: meson: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: simple-card-utils: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: soc-core: remove snd_soc_of_parse_daifmt() include/sound/soc.h | 13 +++- sound/soc/atmel/mikroe-proto.c | 18 ++--- sound/soc/fsl/fsl-asoc-card.c | 16 +--- sound/soc/generic/simple-card-utils.c | 19 ++--- sound/soc/meson/meson-card-utils.c | 15 ++-- sound/soc/soc-core.c | 103 ++++++++++++++++---------- 6 files changed, 99 insertions(+), 85 deletions(-) -- 2.25.1

3 13

[bug report] media: sound/usb: Use Media Controller API to share media resources
by Dan Carpenter 08 Jun '21

08 Jun '21

Hello Shuah Khan, The patch 66354f18fe5f: "media: sound/usb: Use Media Controller API to share media resources" from Apr 1, 2019, leads to the following static checker warning: sound/usb/media.c:287 snd_media_device_create() warn: 'mdev' can also be NULL sound/usb/media.c 270 271 mdev = media_device_usb_allocate(usbdev, KBUILD_MODNAME, THIS_MODULE); ^^^^ If CONFIG_MEDIA_CONTROLLER is disabled then "mdev" is NULL. 272 if (IS_ERR(mdev)) 273 return -ENOMEM; 274 275 /* save media device - avoid lookups */ 276 chip->media_dev = mdev; 277 278 snd_mixer_init: 279 /* Create media entities for mixer and control dev */ 280 ret = snd_media_mixer_init(chip); 281 /* media_device might be registered, print error and continue */ 282 if (ret) 283 dev_err(&usbdev->dev, 284 "Couldn't create media mixer entities. Error: %d\n", 285 ret); 286 287 if (!media_devnode_is_registered(mdev->devnode)) { ^^^^^^^^^^^^^ dereferenced without checking here 288 /* dont'register if snd_media_mixer_init() failed */ 289 if (ret) 290 goto create_fail; 291 292 /* register media_device */ 293 ret = media_device_register(mdev); 294 create_fail: 295 if (ret) { 296 snd_media_mixer_delete(chip); 297 media_device_delete(mdev, KBUILD_MODNAME, THIS_MODULE); 298 /* clear saved media_dev */ 299 chip->media_dev = NULL; 300 dev_err(&usbdev->dev, 301 "Couldn't register media device. Error: %d\n", 302 ret); 303 return ret; 304 } 305 } 306 307 return ret; 308 } regards, dan carpenter

2 3

alsamixer: use background color instead of COLOR_BLACK
by GitHub pull_request - opened 08 Jun '21

08 Jun '21

alsa-project/alsa-utils pull request #77 was opened from r-burns: Hi! I have a cosmetic "improvement" I'd like to share. Scare quotes because this is possibly opinionated, but I'd still like to know what you think. I think on typical terminals, COLOR_BLACK is the same or quite similar to the background color, so this does not make much of a difference. But on "solarized" color schemes and similar, the COLOR_BLACK may be quite different from the background color, so in my opinion it looks odd to have the alsamixer panel stand out this way. For example, I'm using the "Nord" color scheme on a pitch black #000000 background, and here's how the mixer looks with/without this patch (blank terminal, htop, and ncmpcpp also provided for comparison): ![image](https://user-images.githubusercontent.com/52847440/107131140-dbf46500-6888-11eb-938a-9a3258bf7184.png) As you can see, the current alsamixer in the center looks a bit strange and sticks out against my chosen background color and the other TUIs I'm using. Htop and ncmpcpp agree on using the builtin `-1` background color for their own backgrounds, but alsamixer uses COLOR_BLACK which stands out from the actual background. The alsamixer on the bottom right is using this patch, and blends in seamlessly. Again, this is just my opinion! Feel free to close if this is wrong or superfluous. Nonetheless, I thought I'd share this patch, in case anyone else feels the same way. Request URL : https://github.com/alsa-project/alsa-utils/pull/77 Patch URL : https://github.com/alsa-project/alsa-utils/pull/77.patch Repository URL: https://github.com/alsa-project/alsa-utils

2 1

[PATCH 00/15] ASoC: fixes for SoundWire codec drivers
by Pierre-Louis Bossart 08 Jun '21

08 Jun '21

The SOF CI exposed a set of issues with suspend/resume, error handling, register access and mixer values. These fixes were indentified with a tag so that they can be applied by linux-stable and distributions. Thanks to the Realtek and Maxim teams for their help. Bard Liao (3): ASoC: rt5682-sdw: set regcache_cache_only false before reading RT5682_DEVICE_ID ASoC: rt711-sdca-sdw: add readable for SDW_SDCA_CTL() registers ASoC: rt711-sdca: handle mbq_regmap in rt711_sdca_io_init Jack Yu (1): ASoC: rt715-sdca: fix clock stop prepare timeout issue Oder Chiou (1): ASoC: rt5682: Fix a problem with error handling in the io init function of the soundwire Pierre-Louis Bossart (10): ASoC: max98373-sdw: add missing memory allocation check ASoC: max98373-sdw: use first_hw_init flag on resume ASoC: rt1308-sdw: use first_hw_init flag on resume ASoC: rt1316-sdw: use first_hw_init flag on resume ASoC: rt5682-sdw: use first_hw_init flag on resume ASoC: rt700-sdw: use first_hw_init flag on resume ASoC: rt711-sdca-sdw: use first_hw_init flag on resume ASoC: rt711-sdw: use first_hw_init flag on resume ASoC: rt715-sdca-sdw: use first_hw_init flag on resume ASoC: rt715-sdw: use first_hw_init flag on resume sound/soc/codecs/max98373-sdw.c | 14 ++++++++------ sound/soc/codecs/max98373.h | 2 +- sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt1316-sdw.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 19 +++++++++++-------- sound/soc/codecs/rt700-sdw.c | 2 +- sound/soc/codecs/rt711-sdca-sdw.c | 12 +++++++++++- sound/soc/codecs/rt711-sdca.c | 4 ++++ sound/soc/codecs/rt711-sdw.c | 2 +- sound/soc/codecs/rt715-sdca-sdw.c | 3 ++- sound/soc/codecs/rt715-sdca-sdw.h | 1 + sound/soc/codecs/rt715-sdca.c | 9 ++++++--- sound/soc/codecs/rt715-sdca.h | 3 ++- sound/soc/codecs/rt715-sdw.c | 2 +- 14 files changed, 51 insertions(+), 26 deletions(-) -- 2.25.1

2 16

[PATCH] ASoC: mediatek: mtk-btcvsd: Fix an error handling path in 'mtk_btcvsd_snd_probe()'
by Christophe JAILLET 08 Jun '21

08 Jun '21

If an error occurs after a successful 'of_iomap()' call, it must be undone by a corresponding 'iounmap()' call, as already done in the remove function. While at it, remove the useless initialization of 'ret' at the beginning of the function. Fixes: 4bd8597dc36c ("ASoC: mediatek: add btcvsd driver") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- This is NOT compile tested. Even if CONFIG_SND_SOC_MTK_BTCVSD is correctly set in my .config file, I've not been able to have this file compiled. --- sound/soc/mediatek/common/mtk-btcvsd.c | 24 ++++++++++++++++++------ 1 file changed, 18 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index f85b5ea180ec..d884bb7c0fc7 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -1281,7 +1281,7 @@ static const struct snd_soc_component_driver mtk_btcvsd_snd_platform = { static int mtk_btcvsd_snd_probe(struct platform_device *pdev) { - int ret = 0; + int ret; int irq_id; u32 offset[5] = {0, 0, 0, 0, 0}; struct mtk_btcvsd_snd *btcvsd; @@ -1337,7 +1337,8 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) btcvsd->bt_sram_bank2_base = of_iomap(dev->of_node, 1); if (!btcvsd->bt_sram_bank2_base) { dev_err(dev, "iomap bt_sram_bank2_base fail\n"); - return -EIO; + ret = -EIO; + goto unmap_pkv_err; } btcvsd->infra = syscon_regmap_lookup_by_phandle(dev->of_node, @@ -1345,7 +1346,8 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) if (IS_ERR(btcvsd->infra)) { dev_err(dev, "cannot find infra controller: %ld\n", PTR_ERR(btcvsd->infra)); - return PTR_ERR(btcvsd->infra); + ret = PTR_ERR(btcvsd->infra); + goto unmap_bank2_err; } /* get offset */ @@ -1354,7 +1356,7 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) ARRAY_SIZE(offset)); if (ret) { dev_warn(dev, "%s(), get offset fail, ret %d\n", __func__, ret); - return ret; + goto unmap_bank2_err; } btcvsd->infra_misc_offset = offset[0]; btcvsd->conn_bt_cvsd_mask = offset[1]; @@ -1373,8 +1375,18 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) mtk_btcvsd_snd_set_state(btcvsd, btcvsd->tx, BT_SCO_STATE_IDLE); mtk_btcvsd_snd_set_state(btcvsd, btcvsd->rx, BT_SCO_STATE_IDLE); - return devm_snd_soc_register_component(dev, &mtk_btcvsd_snd_platform, - NULL, 0); + ret = devm_snd_soc_register_component(dev, &mtk_btcvsd_snd_platform, + NULL, 0); + if (ret) + goto unmap_bank2_err; + + return 0; + +unmap_bank2_err: + iounmap(btcvsd->bt_sram_bank2_base); +unmap_pkv_err: + iounmap(btcvsd->bt_pkv_base); + return ret; } static int mtk_btcvsd_snd_remove(struct platform_device *pdev) -- 2.30.2

2 1

[PATCH] ASoC: remove unneeded semicolons in wcd934x.c
by Wan Jiabing 08 Jun '21

08 Jun '21

Fix following coccicheck warning: ./sound/soc/codecs/wcd934x.c:5136:2-3: Unneeded semicolon ./sound/soc/codecs/wcd934x.c:2466:2-3: Unneeded semicolon ./sound/soc/codecs/wcd934x.c:2527:2-3: Unneeded semicolon Signed-off-by: Wan Jiabing <wanjiabing(a)vivo.com> --- sound/soc/codecs/wcd934x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 16fd1ab62609..c496b359f2f4 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -2463,7 +2463,7 @@ static int wcd934x_micbias_control(struct snd_soc_component *component, dev_err(component->dev, "%s: Invalid micbias number: %d\n", __func__, micb_num); return -EINVAL; - }; + } mutex_lock(&wcd934x->micb_lock); switch (req) { @@ -2524,7 +2524,7 @@ static int wcd934x_micbias_control(struct snd_soc_component *component, wcd_mbhc_event_notify(wcd934x->mbhc, WCD_EVENT_POST_DAPM_MICBIAS_2_OFF); break; - }; + } mutex_unlock(&wcd934x->micb_lock); @@ -5133,7 +5133,7 @@ static int wcd934x_codec_enable_micbias(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMD: wcd934x_micbias_control(component, micb_num, MICB_DISABLE, true); break; - }; + } return 0; } -- 2.20.1

2 1

[GIT PULL] ASoC fixes for v5.13-rc4
by Mark Brown 08 Jun '21

08 Jun '21

The following changes since commit af2702549d68519ac78228e915d9b2c199056787: ASoC: qcom: lpass-cpu: Use optional clk APIs (2021-05-21 13:12:29 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.13-rc4 for you to fetch changes up to c8a4556d98510ca05bad8d02265a4918b03a8c0b: ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin (2021-06-07 15:54:08 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.13 A collection of fixes and device ID updates that have come up in the past few -rcs, none of which stand out particularly. ---------------------------------------------------------------- Colin Ian King (1): ASoC: topology: Fix spelling mistake "vesion" -> "version" Jack Yu (1): ASoC: rt5659: Fix the lost powers for the HDA header Jerome Brunet (1): ASoC: meson: gx-card: fix sound-dai dt schema Kai Vehmanen (1): ASoC: SOF: reset enabled_cores state at suspend Kefeng Wang (1): ASoC: core: Fix Null-point-dereference in fmt_single_name() Mark Pearson (2): ASoC: AMD Renoir - add DMI entry for Lenovo 2020 AMD platforms ASoC: AMD Renoir: Remove fix for DMI entry on Lenovo 2020 platforms Nicolas Cavallari (1): ASoC: fsl-asoc-card: Set .owner attribute when registering card. Oder Chiou (1): ASoC: rt5682: Fix the fast discharge for headset unplugging in soundwire mode Richard Weinberger (1): ASoC: tas2562: Fix TDM_CFG0_SAMPRATE values Srinivasa Rao Mandadapu (1): ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin .../bindings/sound/amlogic,gx-sound-card.yaml | 4 +- sound/soc/codecs/rt5659.c | 26 +++++-- sound/soc/codecs/rt5682-sdw.c | 3 +- sound/soc/codecs/tas2562.h | 14 ++-- sound/soc/fsl/fsl-asoc-card.c | 1 + sound/soc/qcom/lpass-cpu.c | 79 ++++++++++++++++++++++ sound/soc/qcom/lpass.h | 4 ++ sound/soc/soc-core.c | 2 + sound/soc/soc-topology.c | 6 +- sound/soc/sof/pm.c | 1 + 10 files changed, 122 insertions(+), 18 deletions(-)

2 1

[GIT PULL] HDMI codec improvements
by Maxime Ripard 08 Jun '21

08 Jun '21

Hi, Here's a PR for the changes to hdmi-codec that need to be shared between drm and ASoC. Thanks! Maxime The following changes since commit 6efb943b8616ec53a5e444193dccf1af9ad627b5: Linux 5.13-rc1 (2021-05-09 14:17:44 -0700) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/mripard/linux.git tags/asoc-hdmi-codec-improvements for you to fetch changes up to aee87e9c8fc0efbe933f0a8371990a0754ac65fd: ASoC: hdmi-codec: Add a prepare hook (2021-06-08 17:06:00 +0200) ---------------------------------------------------------------- Improvements to the hdmi-codec driver and ALSA infrastructure around it to support the HDMI Channel Mapping and IEC958 controls ---------------------------------------------------------------- Maxime Ripard (5): ALSA: doc: Clarify IEC958 controls iface ALSA: iec958: Split status creation and fill ASoC: hdmi-codec: Rework to support more controls ASoC: hdmi-codec: Add iec958 controls ASoC: hdmi-codec: Add a prepare hook .../sound/kernel-api/writing-an-alsa-driver.rst | 13 +- include/sound/hdmi-codec.h | 12 +- include/sound/pcm_iec958.h | 8 + sound/core/pcm_iec958.c | 176 +++++++++++++---- sound/soc/codecs/hdmi-codec.c | 219 +++++++++++++++++---- 5 files changed, 337 insertions(+), 91 deletions(-)

1 0

[PATCH 00/66] ALSA: Fix assignment in if condition
by Takashi Iwai 08 Jun '21

08 Jun '21

Hi, here is a largish patch set mostly for cleanup / a slight refactoring of the coding styles with assignment in if condition, typically such as: if ((err = xxx()) < 0) return err; This was a popular style in the old ages, but nowadays it's seen rather fairly harmful as it makes harder to read and easily hides bugs. Since we've hit quite a few issues (although most of them are trivial), I decided to fix those lines. While conversions, I spotted a couple of real bugs. They are old driver codes and only in certain error paths, so most likely no one ever hits it. Takashi === Takashi Iwai (66): ALSA: sb: Fix assignment in if condition ALSA: sb: Minor coding style fixes ALSA: sb: Fix potential double-free of CSP mixer elements ALSA: gus: Fix assignment in if condition ALSA: ad1816a: Fix assignment in if condition ALSA: wavefront: Fix assignment in if condition ALSA: cs423x: Fix assignment in if condition ALSA: opti9xx: Fix assignment in if condition ALSA: opl3sa2: Fix assignment in if condition ALSA: es18xx: Fix assignment in if condition ALSA: cmi8330: Fix assignment in if condition ALSA: als100: Fix assignment in if condition ALSA: azt2320: Fix assignment in if condition ALSA: isa: Fix assignment in if condition ALSA: ad1889: Fix assignment in if condition ALSA: ak4531: Fix assignment in if condition ALSA: als300: Fix assignment in if condition ALSA: als4000: Fix assignment in if condition ALSA: atiixp: Fix assignment in if condition ALSA: azt3328: Fix assignment in if condition ALSA: bt87x: Fix assignment in if condition ALSA: cmipci: Fix assignment in if condition ALSA: cs4281: Fix assignment in if condition ALSA: ens137x: Fix assignment in if condition ALSA: es1938: Fix assignment in if condition ALSA: es1968: Fix assignment in if condition ALSA: fm801: Fix assignment in if condition ALSA: intel8x0: Fix assignment in if condition ALSA: maestro3: Fix assignment in if condition ALSA: rme32: Fix assignment in if condition ALSA: rme96: Fix assignment in if condition ALSA: sonicvibes: Fix assignment in if condition ALSA: via82xx: Fix assignment in if condition ALSA: ac97: Fix assignment in if condition ALSA: au88x0: Fix assignment in if condition ALSA: ca0106: Fix assignment in if condition ALSA: cs46xx: Fix assignment in if condition ALSA: cs5535audio: Fix assignment in if condition ALSA: echoaudio: Fix assignment in if condition ALSA: emu10k1: Fix assignment in if condition ALSA: emu10k1x: Fix assignment in if condition ALSA: ice1712: Fix assignment in if condition ALSA: korg1212: Fix assignment in if condition ALSA: mixart: Fix assignment in if condition ALSA: nm256: Fix assignment in if condition ALSA: pcxhr: Fix assignment in if condition ALSA: riptide: Fix assignment in if condition ALSA: hdsp: Fix assignment in if condition ALSA: rme9652: Fix assignment in if condition ALSA: trident: Fix assignment in if condition ALSA: vx222: Fix assignment in if condition ALSA: ymfpci: Fix assignment in if condition ALSA: core: Fix assignment in if condition ALSA: pcm: Fix assignment in if condition ALSA: oss: Fix assignment in if condition ALSA: seq: Fix assignment in if condition ALSA: pcmcia: Fix assignment in if condition ALSA: sparc: Fix assignment in if condition ALSA: mpu401: Fix assignment in if condition ALSA: vx: Fix assignment in if condition ALSA: opl3: Fix assignment in if condition ALSA: serial: Fix assignment in if condition ALSA: synth: Fix assignment in if condition ALSA: poewrmac: Fix assignment in if condition ALSA: i2c: Fix assignment in if condition ALSA: parisc: Fix assignment in if condition sound/core/hwdep.c | 6 +- sound/core/info_oss.c | 3 +- sound/core/init.c | 12 +- sound/core/oss/mixer_oss.c | 45 +++-- sound/core/oss/pcm_oss.c | 70 +++++--- sound/core/oss/pcm_plugin.c | 26 +-- sound/core/pcm_compat.c | 6 +- sound/core/pcm_misc.c | 12 +- sound/core/pcm_native.c | 6 +- sound/core/seq/oss/seq_oss.c | 26 +-- sound/core/seq/oss/seq_oss_init.c | 9 +- sound/core/seq/oss/seq_oss_midi.c | 33 ++-- sound/core/seq/oss/seq_oss_rw.c | 3 +- sound/core/seq/oss/seq_oss_synth.c | 9 +- sound/core/seq/oss/seq_oss_writeq.c | 3 +- sound/core/seq/seq_clientmgr.c | 51 ++++-- sound/core/seq/seq_dummy.c | 3 +- sound/core/seq/seq_fifo.c | 3 +- sound/core/seq/seq_memory.c | 6 +- sound/core/seq/seq_midi.c | 27 +-- sound/core/seq/seq_queue.c | 21 ++- sound/core/seq/seq_virmidi.c | 9 +- sound/core/sound.c | 3 +- sound/core/sound_oss.c | 3 +- sound/drivers/mpu401/mpu401.c | 9 +- sound/drivers/mpu401/mpu401_uart.c | 19 +- sound/drivers/mtpav.c | 15 +- sound/drivers/mts64.c | 15 +- sound/drivers/opl3/opl3_lib.c | 42 +++-- sound/drivers/opl3/opl3_oss.c | 6 +- sound/drivers/opl3/opl3_seq.c | 9 +- sound/drivers/portman2x4.c | 15 +- sound/drivers/serial-u16550.c | 27 +-- sound/drivers/vx/vx_core.c | 60 ++++--- sound/drivers/vx/vx_hwdep.c | 12 +- sound/drivers/vx/vx_mixer.c | 39 +++-- sound/drivers/vx/vx_pcm.c | 32 ++-- sound/i2c/cs8427.c | 24 ++- sound/i2c/other/ak4114.c | 3 +- sound/i2c/other/ak4117.c | 3 +- sound/i2c/tea6330t.c | 24 ++- sound/isa/ad1816a/ad1816a.c | 23 ++- sound/isa/ad1816a/ad1816a_lib.c | 27 ++- sound/isa/als100.c | 18 +- sound/isa/azt2320.c | 21 ++- sound/isa/cmi8330.c | 37 ++-- sound/isa/cs423x/cs4236.c | 21 ++- sound/isa/cs423x/cs4236_lib.c | 12 +- sound/isa/es1688/es1688_lib.c | 3 +- sound/isa/es18xx.c | 51 ++++-- sound/isa/gus/gus_main.c | 18 +- sound/isa/gus/gus_mem.c | 3 +- sound/isa/gus/gus_mixer.c | 6 +- sound/isa/gus/gus_pcm.c | 23 ++- sound/isa/gus/gus_uart.c | 3 +- sound/isa/gus/gusclassic.c | 6 +- sound/isa/gus/gusextreme.c | 6 +- sound/isa/gus/gusmax.c | 36 ++-- sound/isa/gus/interwave.c | 90 ++++++---- sound/isa/opl3sa2.c | 68 +++++--- sound/isa/opti9xx/miro.c | 27 ++- sound/isa/opti9xx/opti92x-ad1848.c | 27 ++- sound/isa/sb/emu8000.c | 21 ++- sound/isa/sb/emu8000_patch.c | 3 +- sound/isa/sb/emu8000_pcm.c | 6 +- sound/isa/sb/sb16.c | 65 ++++--- sound/isa/sb/sb16_csp.c | 29 ++-- sound/isa/sb/sb16_main.c | 6 +- sound/isa/sb/sb8.c | 38 ++-- sound/isa/sb/sb8_main.c | 3 +- sound/isa/sb/sb8_midi.c | 3 +- sound/isa/sb/sb_common.c | 9 +- sound/isa/sb/sb_mixer.c | 55 +++--- sound/isa/wavefront/wavefront.c | 6 +- sound/isa/wavefront/wavefront_midi.c | 20 ++- sound/isa/wavefront/wavefront_synth.c | 56 +++--- sound/isa/wss/wss_lib.c | 9 +- sound/parisc/harmony.c | 7 +- sound/pci/ac97/ac97_codec.c | 205 ++++++++++++++-------- sound/pci/ac97/ac97_patch.c | 137 ++++++++++----- sound/pci/ad1889.c | 15 +- sound/pci/ak4531_codec.c | 9 +- sound/pci/als300.c | 22 ++- sound/pci/als4000.c | 55 +++--- sound/pci/atiixp.c | 33 ++-- sound/pci/atiixp_modem.c | 40 +++-- sound/pci/au88x0/au88x0.c | 77 +++++---- sound/pci/au88x0/au88x0_a3d.c | 28 +-- sound/pci/au88x0/au88x0_core.c | 47 ++--- sound/pci/au88x0/au88x0_eq.c | 20 ++- sound/pci/au88x0/au88x0_mixer.c | 3 +- sound/pci/au88x0/au88x0_mpu401.c | 14 +- sound/pci/au88x0/au88x0_pcm.c | 15 +- sound/pci/azt3328.c | 3 +- sound/pci/bt87x.c | 3 +- sound/pci/ca0106/ca0106_main.c | 18 +- sound/pci/ca0106/ca_midi.c | 3 +- sound/pci/cmipci.c | 83 ++++++--- sound/pci/cs4281.c | 54 ++++-- sound/pci/cs46xx/cs46xx.c | 31 ++-- sound/pci/cs46xx/cs46xx_lib.c | 51 ++++-- sound/pci/cs46xx/dsp_spos.c | 3 +- sound/pci/cs5535audio/cs5535audio.c | 28 +-- sound/pci/cs5535audio/cs5535audio_pcm.c | 10 +- sound/pci/echoaudio/darla20_dsp.c | 6 +- sound/pci/echoaudio/darla24_dsp.c | 6 +- sound/pci/echoaudio/echo3g_dsp.c | 3 +- sound/pci/echoaudio/echoaudio.c | 220 ++++++++++++++---------- sound/pci/echoaudio/echoaudio_dsp.c | 12 +- sound/pci/echoaudio/echoaudio_gml.c | 3 +- sound/pci/echoaudio/gina20_dsp.c | 6 +- sound/pci/echoaudio/gina24_dsp.c | 6 +- sound/pci/echoaudio/indigo_dsp.c | 6 +- sound/pci/echoaudio/indigodj_dsp.c | 6 +- sound/pci/echoaudio/indigoio_dsp.c | 6 +- sound/pci/echoaudio/layla20_dsp.c | 6 +- sound/pci/echoaudio/layla24_dsp.c | 9 +- sound/pci/echoaudio/mia_dsp.c | 6 +- sound/pci/echoaudio/midi.c | 4 +- sound/pci/echoaudio/mona_dsp.c | 6 +- sound/pci/emu10k1/emu10k1.c | 42 +++-- sound/pci/emu10k1/emu10k1_callback.c | 3 +- sound/pci/emu10k1/emu10k1x.c | 85 +++++---- sound/pci/emu10k1/emufx.c | 13 +- sound/pci/emu10k1/emumixer.c | 78 ++++++--- sound/pci/emu10k1/emumpu401.c | 12 +- sound/pci/emu10k1/emupcm.c | 24 ++- sound/pci/emu10k1/memory.c | 21 ++- sound/pci/emu10k1/p16v.c | 13 +- sound/pci/emu10k1/timer.c | 3 +- sound/pci/ens1370.c | 45 +++-- sound/pci/es1938.c | 42 +++-- sound/pci/es1968.c | 75 ++++---- sound/pci/fm801.c | 63 ++++--- sound/pci/ice1712/delta.c | 6 +- sound/pci/ice1712/ews.c | 24 ++- sound/pci/intel8x0.c | 46 +++-- sound/pci/intel8x0m.c | 36 ++-- sound/pci/korg1212/korg1212.c | 24 ++- sound/pci/maestro3.c | 21 ++- sound/pci/mixart/mixart.c | 29 ++-- sound/pci/mixart/mixart_hwdep.c | 17 +- sound/pci/mixart/mixart_mixer.c | 33 ++-- sound/pci/nm256/nm256.c | 27 ++- sound/pci/pcxhr/pcxhr.c | 22 ++- sound/pci/pcxhr/pcxhr_hwdep.c | 9 +- sound/pci/riptide/riptide.c | 88 ++++++---- sound/pci/rme32.c | 76 +++++--- sound/pci/rme96.c | 148 +++++++++------- sound/pci/rme9652/hdsp.c | 121 ++++++++----- sound/pci/rme9652/rme9652.c | 98 ++++++----- sound/pci/sonicvibes.c | 67 +++++--- sound/pci/trident/trident.c | 41 +++-- sound/pci/trident/trident_main.c | 84 ++++++--- sound/pci/via82xx.c | 78 ++++++--- sound/pci/via82xx_modem.c | 52 ++++-- sound/pci/vx222/vx222.c | 18 +- sound/pci/vx222/vx222_ops.c | 12 +- sound/pci/ymfpci/ymfpci.c | 71 ++++---- sound/pci/ymfpci/ymfpci_main.c | 72 +++++--- sound/pcmcia/pdaudiocf/pdaudiocf.c | 3 +- sound/pcmcia/vx/vxp_mixer.c | 6 +- sound/pcmcia/vx/vxp_ops.c | 6 +- sound/pcmcia/vx/vxpocket.c | 3 +- sound/ppc/beep.c | 5 +- sound/ppc/daca.c | 24 ++- sound/ppc/keywest.c | 6 +- sound/ppc/pmac.c | 6 +- sound/ppc/powermac.c | 21 ++- sound/ppc/tumbler.c | 72 +++++--- sound/sparc/amd7930.c | 14 +- sound/sparc/cs4231.c | 10 +- sound/sparc/dbri.c | 11 +- sound/synth/emux/emux.c | 3 +- sound/synth/emux/emux_effect.c | 13 +- sound/synth/emux/emux_hwdep.c | 6 +- sound/synth/emux/soundfont.c | 40 +++-- 177 files changed, 3186 insertions(+), 1831 deletions(-) -- 2.26.2

1 66

[PATCH 0/2] ASoC: Intel: boards: use software node API
by Pierre-Louis Bossart 08 Jun '21

08 Jun '21

This is an update on an earlier contribution from Heikki Krogerus, with mainly changes to avoid the use of devm_ routines for Baytrail machine drivers. Heikki Krogerus (2): ASoC: Intel: boards: use software node API in SoundWire machines ASoC: Intel: boards: use software node API in Atom boards sound/soc/intel/boards/bytcht_es8316.c | 25 +++++++++-- sound/soc/intel/boards/bytcr_rt5640.c | 42 +++++++++++++++--- sound/soc/intel/boards/bytcr_rt5651.c | 47 +++++++++++++++++---- sound/soc/intel/boards/sof_sdw_rt711.c | 20 ++++++--- sound/soc/intel/boards/sof_sdw_rt711_sdca.c | 20 ++++++--- 5 files changed, 127 insertions(+), 27 deletions(-) -- 2.25.1

3 7

[PATCH v3] ASoC: qcom: Add four speaker support on MI2S secondary
by Srinivasa Rao Mandadapu 08 Jun '21

08 Jun '21

Add four speaker support on MI2S secondary block by using I2S SD1 line on gpio52 pin, and add channel map control support in the lpass-cpu audio driver. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-cpu.c | 40 ++++++++++++++++++++++++++++++++++- sound/soc/qcom/lpass-sc7180.c | 1 + sound/soc/qcom/lpass.h | 8 ++++++- 3 files changed, 47 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c62d2612e8f5..d03cc359983a 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -324,10 +324,43 @@ const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int ret; + struct snd_soc_dai_driver *drv = dai->driver; + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + + ret = snd_pcm_add_chmap_ctls(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_alt_chmaps, drv->playback.channels_max, 0, + &drvdata->chmap_info); + if (ret < 0) + return ret; + + return 0; +} +EXPORT_SYMBOL_GPL(lpass_cpu_pcm_new); + int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); - int ret; + struct snd_soc_dai_driver *drv = dai->driver; + int ret, i; + + drvdata->rx_chmap.channels = drv->playback.channels_max; + drvdata->tx_chmap.channels = drv->playback.channels_max; + + if (drv->playback.channels_max == 2) + drvdata->chmap_idx = 1; + else if (drv->playback.channels_max == 4) + drvdata->chmap_idx = 2; + + for (i = 0; i < drv->playback.channels_max; i++) { + drvdata->tx_chmap.map[i] = + snd_pcm_alt_chmaps[drvdata->chmap_idx].map[i]; + drvdata->rx_chmap.map[i] = + snd_pcm_alt_chmaps[drvdata->chmap_idx].map[i]; + } /* ensure audio hardware is disabled */ ret = regmap_write(drvdata->lpaif_map, @@ -856,6 +889,11 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) PTR_ERR(drvdata->mi2s_bit_clk[dai_id])); return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]); } + if (drvdata->mi2s_playback_sd_mode[dai_id] == + LPAIF_I2SCTL_MODE_QUAD01) { + variant->dai_driver[dai_id].playback.channels_min = 4; + variant->dai_driver[dai_id].playback.channels_max = 4; + } } /* Allocation for i2sctl regmap fields */ diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 8c168d3c589e..77a556b27cf0 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -58,6 +58,7 @@ static struct snd_soc_dai_driver sc7180_lpass_cpu_dai_driver[] = { }, .probe = &asoc_qcom_lpass_cpu_dai_probe, .ops = &asoc_qcom_lpass_cpu_dai_ops, + .pcm_new = lpass_cpu_pcm_new, }, { .id = LPASS_DP_RX, .name = "Hdmi", diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..db239ea52946 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -110,6 +110,11 @@ struct lpass_data { struct lpass_hdmitx_dmactl *hdmi_tx_dmactl[LPASS_MAX_HDMI_DMA_CHANNELS]; struct lpass_dp_metadata_ctl *meta_ctl; struct lpass_sstream_ctl *sstream_ctl; + /* Channel map information */ + struct snd_pcm_chmap *chmap_info; + struct snd_pcm_chmap_elem rx_chmap; + struct snd_pcm_chmap_elem tx_chmap; + int chmap_idx; }; /* Vairant data per each SOC */ @@ -259,5 +264,6 @@ void asoc_qcom_lpass_cpu_platform_shutdown(struct platform_device *pdev); int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev); int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai); extern const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops; - +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai); #endif /* __LPASS_H__ */ -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

1 0

[PATCH V2 0/2] Read all configuration at init time
by Geoffrey D. Bennett 08 Jun '21

08 Jun '21

Version two of patches to add support for reading the mixer volumes and mux configuration from the hardware when the driver is initialising. Changes since v1: - fix missing endian conversion spotted by ktest bot Geoffrey D. Bennett (2): ALSA: usb-audio: scarlett2: Read mixer volumes at init time ALSA: usb-audio: scarlett2: Read mux at init time sound/usb/mixer_scarlett_gen2.c | 227 ++++++++++++++++++++++---------- 1 file changed, 161 insertions(+), 66 deletions(-) -- 2.31.1

2 3

[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs for HP ZBook Power G8
by Jeremy Szu 08 Jun '21

08 Jun '21

The HP ZBook Power G8 using ALC236 codec which using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dcd9bceace68..f2fa6953b8b7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8335,6 +8335,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), + SND_PCI_QUIRK(0x103c, 0x888d, "HP ZBook Power 15.6 inch G8 Mobile Workstation PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), -- 2.31.1

2 1

[PATCH] ALSA: hda/realtek: headphone and mic don't work on an Acer laptop
by Hui Wang 08 Jun '21

08 Jun '21

There are 2 issues on this machine, the 1st one is mic's plug/unplug can't be detected, that is because the mic is set to manual detecting mode, need to apply ALC255_FIXUP_XIAOMI_HEADSET_MIC to set it to auto detecting mode. The other one is headphone's plug/unplug can't be detected by pulseaudio, that is because the pulseaudio will use ucm2/sof-hda-dsp on this machine, and the ucm2 only handle 'Headphone Jack', but on this machine the headphone's pincfg sets the location to Front, then the alsa mixer name is "Front Headphone Jack" instead of "Headphone Jack", so override the pincfg to change location to Left. BugLink: http://bugs.launchpad.net/bugs/1930188 Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dcd9bceace68..be4a26f6bb96 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6568,6 +6568,7 @@ enum { ALC285_FIXUP_HP_SPECTRE_X360, ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, ALC623_FIXUP_LENOVO_THINKSTATION_P340, + ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -8146,6 +8147,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC283_FIXUP_HEADSET_MIC, }, + [ALC255_FIXUP_ACER_HEADPHONE_AND_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x21, 0x03211030 }, /* Change the Headphone location to Left */ + { } + }, + .chained = true, + .chain_id = ALC255_FIXUP_XIAOMI_HEADSET_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8182,6 +8192,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X), @@ -8740,6 +8751,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, {.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"}, {.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"}, + {.id = ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, .name = "alc255-acer-headphone-and-mic"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.25.1

2 1

[PATCH][next][V2] ASoC: rsnd: check for zero node count
by Colin King 08 Jun '21

08 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> Most callers of_get_child_count() check that "nr" is non-zero so it causes a static checker warning when we don't do that here. This does not cause a problem or a crash, but having zero SSUIes does not make sense either so let's add a check. Addresses-Coverity: ("Unchecked return value") Fixes: c413983eb66a ("ASoC: rsnd: adjust disabled module") Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- V2: Rephrase commit message based on Dan Carpenter's suggestion. Thanks Dan! --- sound/soc/sh/rcar/ssiu.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 5682c74bb7ff..0d8f97633dd2 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -515,6 +515,9 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv) else nr = priv->ssi_nr; + if (!nr) + return -EINVAL; + ssiu = devm_kcalloc(dev, nr, sizeof(*ssiu), GFP_KERNEL); if (!ssiu) return -ENOMEM; -- 2.31.1

3 2

added an option to aseqnet to optionally set the midi process name
by GitHub pull_request - opened 07 Jun '21

07 Jun '21

alsa-project/alsa-utils pull request #95 was opened from BrainDamage: This option allows to run multiple instances of aseqnet without having to double check the assigned port number, since each one can get spawned with a unique name. The patch is pretty minimal and clean, thopefully here shouldn't be troubles merging it. Request URL : https://github.com/alsa-project/alsa-utils/pull/95 Patch URL : https://github.com/alsa-project/alsa-utils/pull/95.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 0/4] Support CS42L42 on JSL platform
by Brent Lu 07 Jun '21

07 Jun '21

This series consists four patches. Patch 1 adds dai sequence support for cml/jsl/tgl platforms which is different from the sequence on glk platform. Patch 2 adds max98360a support to the maxim-common module. Patch 3 adds driver data for jsl_cs4242_mx98360a which supports cs42l42 and max98360a running on jsl boards. Patch 4 refactor the sof_rt5682 to use the max98360 code in the maxim-common module. Brent Lu (4): ASoC: Intel: sof_cs42l42: support JSL DAI link sequence ASoC: Intel: maxim-common: support max98360a ASoC: intel: sof_cs42l42: add support for jsl_cs4242_mx98360a ASoC: Intel: sof_rt5682: code refactor for max98360a sound/soc/intel/boards/sof_cs42l42.c | 340 ++++++++++++------ sound/soc/intel/boards/sof_maxim_common.c | 17 +- sound/soc/intel/boards/sof_maxim_common.h | 4 +- sound/soc/intel/boards/sof_rt5682.c | 52 +-- .../intel/common/soc-acpi-intel-jsl-match.c | 8 + 5 files changed, 254 insertions(+), 167 deletions(-) -- 2.17.1

3 12

[PATCH 0/2] ALSA: usb-audio: scarlett2: Read all configuration at init time
by Geoffrey D. Bennett 07 Jun '21

07 Jun '21

These two patches add support for reading the mixer volumes and mux configuration from the hardware when the driver is initialising. Previously the ALSA volume controls were initialised to zero and the mux configuration set to a fixed default instead of being initialised to match the hardware state. The ALSA controls for the Scarlett Gen 2 interfaces should now always be in sync with the hardware. Thanks to Vladimir Sadovnikov for figuring out how to do this. Takashi, if these pass your review, I believe that they are appropriate for: #Cc: stable(a)vger.kernel.org Geoffrey D. Bennett (2): ALSA: usb-audio: scarlett2: Read mixer volumes at init time ALSA: usb-audio: scarlett2: Read mux at init time sound/usb/mixer_scarlett_gen2.c | 227 ++++++++++++++++++++++---------- 1 file changed, 161 insertions(+), 66 deletions(-) -- 2.31.1

4 7

[PATCH][V2][next] ASoC: rk817: remove redundant assignment to pointer node, add missing of_node_put
by Colin King 07 Jun '21

07 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> The pointer node is being initialized with a value that is never read and it is being updated later with a new value. The initialization is redundant and can be removed. The function is missing a of_node_put on node, fix this by adding the call before returning. Addresses-Coverity: ("Unused value") Fixes: 0d6a04da9b25 ("ASoC: Add Rockchip rk817 audio CODEC support") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- V2: Add missing of_node_put call, kudos to Dan Carpenter for spotting this issue. --- sound/soc/codecs/rk817_codec.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index fd3a5ba034a9..a44d3cad1119 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -456,7 +456,7 @@ static const struct snd_soc_component_driver soc_codec_dev_rk817 = { static void rk817_codec_parse_dt_property(struct device *dev, struct rk817_codec_priv *rk817) { - struct device_node *node = dev->parent->of_node; + struct device_node *node; node = of_get_child_by_name(dev->parent->of_node, "codec"); if (!node) { @@ -466,6 +466,8 @@ static void rk817_codec_parse_dt_property(struct device *dev, rk817->mic_in_differential = of_property_read_bool(node, "rockchip,mic-in-differential"); + + of_node_put(node); } static int rk817_platform_probe(struct platform_device *pdev) -- 2.31.1

3 2

[v7] ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
by Judy Hsiao 07 Jun '21

07 Jun '21

From: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> This patch fixes PoP noise of around 15ms observed during audio capture begin. Enables BCLK and LRCLK in snd_soc_dai_ops prepare call for introducing some delay before capture start. Co-developed-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> (am from https://patchwork.kernel.org/patch/12276369/) (also found at https://lore.kernel.org/r/20210524142114.18676-1-srivasam@codeaurora.org) --- Changes Since V6: -- Removed clk_disable OSR clock in lpass_cpu_daiops_prepare error case as failure of prepare will result in calling shutdown which should take care of this. Changes Since V5: -- Fixed nit. -- Updated `mi2s_was_prepared[dai->driver->id] = true;` after checking for errors. Changes Since V4: -- Replaced the __clk_is_enabled(BCLK) check by the self maintained. mi2s_was_prepared bool state. -- Removed unrelated changes. -- Refined comments. Changes Since V3: -- Checked BCLK is off before enabling it in lpass_cpu_daiops_prepare as lpass_cpu_daiops_prepare can be called multiple times -- Checked BCLK is on before disabling it in lpass_cpu_daiops_shutdown to fix the WARN. It is because BCLK may not be enabled if lpass_cpu_daiops_prepare is not called before lpass_cpu_daiops_shutdown -- Added more comments Changes Since V2: -- Updated comments as per linux style -- Removed unrelated changes Changes Since V1: -- Enabled BCLK and LRCLK in dai ops prepare API instead of startup API -- Added comments sound/soc/qcom/lpass-cpu.c | 79 ++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/lpass.h | 4 ++ 2 files changed, 83 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index af8cb64924a0..647423a6cb57 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -93,8 +93,30 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + /* + * Ensure LRCLK is disabled even in device node validation. + * Will not impact if disabled in lpass_cpu_daiops_trigger() + * suspend. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); + else + regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE); + + /* + * BCLK may not be enabled if lpass_cpu_daiops_prepare is called before + * lpass_cpu_daiops_shutdown. It's paired with the clk_enable in + * lpass_cpu_daiops_prepare. + */ + if (drvdata->mi2s_was_prepared[dai->driver->id]) { + drvdata->mi2s_was_prepared[dai->driver->id] = false; + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); + } + clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } @@ -275,6 +297,18 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * Ensure lpass BCLK/LRCLK is enabled during + * device resume as lpass_cpu_daiops_prepare() is not called + * after the device resumes. We don't check mi2s_was_prepared before + * enable/disable BCLK in trigger events because: + * 1. These trigger events are paired, so the BCLK + * enable_count is balanced. + * 2. the BCLK can be shared (ex: headset and headset mic), + * we need to increase the enable_count so that we don't + * turn off the shared BCLK while other devices are using + * it. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); @@ -296,6 +330,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* + * To ensure lpass BCLK/LRCLK is disabled during + * device suspend. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); @@ -315,12 +353,53 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, return ret; } +static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret; + + /* + * Ensure lpass BCLK/LRCLK is enabled bit before playback/capture + * data flow starts. This allows other codec to have some delay before + * the data flow. + * (ex: to drop start up pop noise before capture starts). + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); + else + ret = regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_ENABLE); + + if (ret) { + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + return ret; + } + + /* + * Check mi2s_was_prepared before enabling BCLK as lpass_cpu_daiops_prepare can + * be called multiple times. It's paired with the clk_disable in + * lpass_cpu_daiops_shutdown. + */ + if (!drvdata->mi2s_was_prepared[dai->driver->id]) { + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + return ret; + } + drvdata->mi2s_was_prepared[dai->driver->id] = true; + } + return 0; +} + const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, .trigger = lpass_cpu_daiops_trigger, + .prepare = lpass_cpu_daiops_prepare, }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..7f72214404ba 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -67,6 +67,10 @@ struct lpass_data { /* MI2S SD lines to use for playback/capture */ unsigned int mi2s_playback_sd_mode[LPASS_MAX_MI2S_PORTS]; unsigned int mi2s_capture_sd_mode[LPASS_MAX_MI2S_PORTS]; + + /* The state of MI2S prepare dai_ops was called */ + bool mi2s_was_prepared[LPASS_MAX_MI2S_PORTS]; + int hdmi_port_enable; /* low-power audio interface (LPAIF) registers */ -- 2.32.0.rc1.229.g3e70b5a671-goog

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[PATCH v3 0/7] ASoC: adds new .auto_selectable_formats support
by Kuninori Morimoto 07 Jun '21

07 Jun '21

Hi Mark These are v3 of "ASoC: adds new .get_fmt support", but renamed Subject. This is a little bit challenging patch-set. The idea/code is almost same as v1 / v2. v3 has "priority" support. We need to set dai_link->dai_fmt to select CPU/Codec settings, and it is selected by Sound Card Driver, today. Because of it, Sound Card user need to know both CPU / Codec available dai_fmt, and needs to select it. For example simple-card / audio-graph case, it is selected by "format" and "bitclock/frame-master/inversion" on DT. But, it can be automatically selected if both CPU and Codec drivers indicate it to ALSA SoC Framework, somehow. By this patch, dai_fmt can be automatically selected from each driver if both CPU / Codec driver had .auto_selectable_formats. Automatically selectable *field* is depends on each drivers. For example, some driver want to select format "automatically", but want to select other fields "manually", because of complex limitation. Or other example, in case of both CPU and Codec are possible to be clock provider, but the quality was different. In these case, user need/want to *manually* select each fields from Sound Card driver. It uses Sound Card specified fields preferentially, and try to select non-specific fields from CPU and Codec driver settings if driver had .auto_selectable_formats. In other words, we can select all dai_fmt via Sound Card driver same as before. Select dai_fmt 100% automatically is very difficult and will be very complex, but select automatically some fields only is very easy, I guess. This patch-set is based on such assumption. v1 -> v2 - Add more detail explanation on git-log, code, comment. - Possible to be Clock/Frame provider is depends on driver's situation. v2 -> v3 - has priority - tidyup function explanation for snd_soc_dai_get_fmt() - Each driver don't try to have SND_SOC_DAIFMT_CBx_CFx to avoid confusion Link: https://lore.kernel.org/r/871rb3hypy.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871racbx0w.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (7): ASoC: soc-core: move snd_soc_runtime_set_dai_fmt() to upside ASoC: soc-core: add snd_soc_runtime_get_dai_fmt() ASoC: ak4613: add .auto_selectable_formats support ASoC: pcm3168a: add .auto_selectable_formats support ASoC: rsnd: add .auto_selectable_formats support ASoC: fsi: add .auto_selectable_formats support ASoC: hdmi-codec: add .auto_selectable_formats support include/sound/soc-dai.h | 55 +++++++ sound/soc/codecs/ak4613.c | 11 ++ sound/soc/codecs/hdmi-codec.c | 21 +++ sound/soc/codecs/pcm3168a.c | 26 +++ sound/soc/sh/fsi.c | 15 ++ sound/soc/sh/rcar/core.c | 31 +++- sound/soc/soc-core.c | 288 ++++++++++++++++++++++++++-------- sound/soc/soc-dai.c | 63 ++++++++ sound/soc/soc-utils.c | 29 ++++ 9 files changed, 475 insertions(+), 64 deletions(-) -- 2.25.1

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[PATCH] ASoC: rk817: Remove unneeded semicolon
by Jiapeng Chong 07 Jun '21

07 Jun '21

Fix the following coccicheck warnings: ./sound/soc/codecs/rk817_codec.c:49:2-3: Unneeded semicolon. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/codecs/rk817_codec.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 17e672b..e95fe75 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -46,7 +46,8 @@ static int rk817_init(struct snd_soc_component *component) if (rk817->mic_in_differential) { snd_soc_component_update_bits(component, RK817_CODEC_AMIC_CFG0, MIC_DIFF_MASK, MIC_DIFF_EN); - }; + } + return 0; } -- 1.8.3.1

2 2

Re: [PATCH 0/2] ALSA: usb-audio: scarlett2: Read all configuration at init time
by Takashi Iwai 07 Jun '21

07 Jun '21

On Mon, 07 Jun 2021 17:00:10 +0200, Vladimir Sadovnikov wrote: > > Hello! > > I would like to say some words from my side. > > The Scarlett device (especially 18i20) is pretty complicated device > and holds a lot of settings in it's internal > configuration area (hardware and software). > > So this is not the only patch which will configure the driver in proper way. > Since the device stores it's internal state (and that's good for power > safety and mobility), ideally, we should get > the almost fully compatible mixer settings with the original Focusrite > Control Software. > > The huge amount of job I've already done i my fork of Geoffrey's driver: > https://github.com/sadko4u/focusrite-scarlett-backports/blob/master/prod-dr… > > So we're planning to work on integrating our changes into the common > patch sets and will submit changes here. Sure, I don't mean against the patches, this looks like an acceptable approach. So don't worry, I'd take the patches once when the fixed version is submitted. However, from the system design POV, all those configurations should be a software issue, and ideally we shouldn't rely on the hardware preset state which has been done *somehow* -- it may allow malfunction easily. One thing I've learned over years is that you can never trust hardware :) Takashi > > Best, > Vladimir > > 07.06.2021 10:23, Takashi Iwai пишет: > > On Sun, 06 Jun 2021 16:16:44 +0200, > > Geoffrey D. Bennett wrote: > >> These two patches add support for reading the mixer volumes and mux > >> configuration from the hardware when the driver is initialising. > >> > >> Previously the ALSA volume controls were initialised to zero and the > >> mux configuration set to a fixed default instead of being initialised > >> to match the hardware state. > >> > >> The ALSA controls for the Scarlett Gen 2 interfaces should now always > >> be in sync with the hardware. Thanks to Vladimir Sadovnikov for > >> figuring out how to do this. > >> > >> Takashi, if these pass your review, I believe that they are > >> appropriate for: > >> #Cc: stable(a)vger.kernel.org > > Well, in general, having a proper fixed value for the initial mixer > > value is the right thing, which is a part of the driver's role. > > Though, in snd-usb-audio, we don't set up the initial values just > > because of laziness; since the topology in USB audio is variable per > > device and often hard to parse correctly, it's difficult to determine > > the suitable initial values, hence we leave untouched. So, in that > > sense, setting the zero isn't wrong, rather safer, per se. > > > > However, Scarlett 2 seems to want to be different; it has already some > > initialization code to read the existing configs. So this change > > sounds more or less acceptable. But it's questionable whether it's > > really for stable as a "fix". > > > > In anyway, please fix the bug ktest bot spotted, the missing endian > > conversions and resubmit. > > > > > > thanks, > > > > Takashi > >

2 1

alsa-lib 1.2.5(-2 on arch) only one of two connected hdmi/displayport outputs visible at once. My mic is completely MIA
by GitHub issues - opened 07 Jun '21

07 Jun '21

alsa-project/alsa-lib issue #148 was opened from fkroener: Since the recent update to 1.2.5-2 on archlinux, only one of my two usb-c/displayport/hdmi devices show up in pavucontrol and kde mixer. I am able to switch between hdmi0 and hdmi1, by changing the Profile in pavucontrol of HD-Audio Generic. The microphone doesn't show up at all now. ![KDE Mixer](https://user-images.githubusercontent.com/28690007/121057421-8c36938… This doesn't appear related to #142 or #143, since I have revision 2 from arch installed, which should include the fixes. This is a Lenovo T14 AMD. ``` pactl list cards Card #0 Name: alsa_card.pci-0000_07_00.1 Driver: module-alsa-card.c Owner Module: 6 Properties: alsa.card = "0" alsa.card_name = "HD-Audio Generic" alsa.long_card_name = "HD-Audio Generic at 0xfd3c8000 irq 91" alsa.driver_name = "snd_hda_intel" device.bus_path = "pci-0000:07:00.1" sysfs.path = "/devices/pci0000:00/0000:00:08.1/0000:07:00.1/sound/card0" device.bus = "pci" device.vendor.id = "1002" device.vendor.name = "Advanced Micro Devices, Inc. [AMD/ATI]" device.product.id = "1637" device.string = "0" device.description = "HD-Audio Generic" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" Profiles: output:hdmi-stereo: Digital Stereo (HDMI) Output (sinks: 1, sources: 0, priority: 5900, available: yes) output:hdmi-stereo-extra1: Digital Stereo (HDMI 2) Output (sinks: 1, sources: 0, priority: 5700, available: yes) output:hdmi-stereo-extra2: Digital Stereo (HDMI 3) Output (sinks: 1, sources: 0, priority: 5700, available: no) output:hdmi-surround-extra2: Digital Surround 5.1 (HDMI 3) Output (sinks: 1, sources: 0, priority: 600, available: no) output:hdmi-surround71-extra2: Digital Surround 7.1 (HDMI 3) Output (sinks: 1, sources: 0, priority: 600, available: no) off: Off (sinks: 0, sources: 0, priority: 0, available: yes) Active Profile: output:hdmi-stereo Ports: hdmi-output-0: HDMI / DisplayPort (type: HDMI, priority: 5900, latency offset: 0 usec, available) Properties: device.icon_name = "video-display" device.product.name = "P27h-20" Part of profile(s): output:hdmi-stereo hdmi-output-1: HDMI / DisplayPort 2 (type: HDMI, priority: 5800, latency offset: 0 usec, available) Properties: device.icon_name = "video-display" device.product.name = "P27h-20" Part of profile(s): output:hdmi-stereo-extra1 hdmi-output-2: HDMI / DisplayPort 3 (type: HDMI, priority: 5700, latency offset: 0 usec, not available) Properties: device.icon_name = "video-display" Part of profile(s): output:hdmi-stereo-extra2, output:hdmi-surround-extra2, output:hdmi-surround71-extra2 Card #1 Name: alsa_card.pci-0000_07_00.6 Driver: module-alsa-card.c Owner Module: 8 Properties: alsa.card = "1" alsa.card_name = "HD-Audio Generic" alsa.long_card_name = "HD-Audio Generic at 0xfd3c0000 irq 92" alsa.driver_name = "snd_hda_intel" device.bus_path = "pci-0000:07:00.6" sysfs.path = "/devices/pci0000:00/0000:00:08.1/0000:07:00.6/sound/card1" device.bus = "pci" device.vendor.id = "1022" device.vendor.name = "Advanced Micro Devices, Inc. [AMD]" device.product.id = "15e3" device.product.name = "Family 17h (Models 10h-1fh) HD Audio Controller" device.string = "1" device.description = "Family 17h (Models 10h-1fh) HD Audio Controller" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" Profiles: input:analog-stereo: Analog Stereo Input (sinks: 0, sources: 1, priority: 65, available: no) output:analog-stereo: Analog Stereo Output (sinks: 1, sources: 0, priority: 6500, available: yes) output:analog-stereo+input:analog-stereo: Analog Stereo Duplex (sinks: 1, sources: 1, priority: 6565, available: no) off: Off (sinks: 0, sources: 0, priority: 0, available: yes) Active Profile: output:analog-stereo Ports: analog-input-mic: Microphone (type: Mic, priority: 8700, latency offset: 0 usec, not available) Properties: device.icon_name = "audio-input-microphone" Part of profile(s): input:analog-stereo, output:analog-stereo+input:analog-stereo analog-output-speaker: Speakers (type: Speaker, priority: 10000, latency offset: 0 usec, availability unknown) Properties: device.icon_name = "audio-speakers" Part of profile(s): output:analog-stereo, output:analog-stereo+input:analog-stereo analog-output-headphones: Headphones (type: Headphones, priority: 9900, latency offset: 0 usec, not available) Properties: device.icon_name = "audio-headphones" Part of profile(s): output:analog-stereo, output:analog-stereo+input:analog-stereo ``` If you need any more information, I'm gladly at your disposal. Thanks! Issue URL : https://github.com/alsa-project/alsa-lib/issues/148 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH AUTOSEL 5.12 27/49] ALSA: hda: Add AlderLake-M PCI ID
by Sasha Levin 07 Jun '21

07 Jun '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 4ad7935df6a566225c3d51900bde8f2f0f8b6de3 ] Add HD Audio PCI ID for Intel AlderLake-M. Add rules to snd_intel_dsp_find_config() to choose SOF driver for ADL-M systems with PCH-DMIC or Soundwire codecs, and legacy driver for the rest. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210528185123.48332-1-kai.vehmanen@linux.intel.c… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/intel-dsp-config.c | 4 ++++ sound/pci/hda/hda_intel.c | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index ab5ff7867eb9..d8be146793ee 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -331,6 +331,10 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x51c8, }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x51cc, + }, #endif }; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 79ade335c8a0..470753b36c8a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2485,6 +2485,9 @@ static const struct pci_device_id azx_ids[] = { /* Alderlake-P */ { PCI_DEVICE(0x8086, 0x51c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Alderlake-M */ + { PCI_DEVICE(0x8086, 0x51cc), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, -- 2.30.2

1 0

[PATCH 0/9] ALSA: firewire: cease from delayed card registration
by Takashi Sakamoto 07 Jun '21

07 Jun '21

Hi, Since Linux kernel v4.7, delayed card registration was introduced to all drivers in ALSA firewire stack. Nowadays it brings less benefit than code complication. This patchset ceases from it. Takashi Sakamoto (9): ALSA: bebob: cease from delayed card registration ALSA: fireworks: cease from delayed card registration ALSA: oxfw: cease from delayed card registration ALSA: dice: cease from delayed card registration ALSA: firewire-digi00x: cease from delayed card registration ALSA: firewire-tascam: cease from delayed card registration ALSA: firewire-motu: cease from delayed card registration ALSA: fireface: cease from delayed card registration ALSA: firewire-lib: delete unused kernel API sound/firewire/bebob/bebob.c | 159 ++++++++++----------------- sound/firewire/bebob/bebob.h | 4 - sound/firewire/dice/dice.c | 138 ++++++++--------------- sound/firewire/dice/dice.h | 4 - sound/firewire/digi00x/digi00x.c | 101 +++++------------ sound/firewire/digi00x/digi00x.h | 3 - sound/firewire/fireface/ff.c | 90 +++++---------- sound/firewire/fireface/ff.h | 3 - sound/firewire/fireworks/fireworks.c | 107 ++++++------------ sound/firewire/fireworks/fireworks.h | 3 - sound/firewire/lib.c | 32 ------ sound/firewire/lib.h | 3 - sound/firewire/motu/motu.c | 84 +++++--------- sound/firewire/motu/motu.h | 3 - sound/firewire/oxfw/oxfw.c | 128 ++++++++------------- sound/firewire/oxfw/oxfw.h | 6 +- sound/firewire/tascam/tascam.c | 92 +++++----------- sound/firewire/tascam/tascam.h | 2 - 18 files changed, 296 insertions(+), 666 deletions(-) -- 2.27.0

2 10

[PATCH v2] ASoC: qcom: Fix for DMA interrupt clear reg overwriting
by Srinivasa Rao Mandadapu 07 Jun '21

07 Jun '21

The DMA interrupt clear register overwritten during simultaneous playback and capture in lpass platform interrupt handler. It's causing playback or capture stuck in similtaneous plaback on speaker and capture on dmic test. Update appropriate reg fields of corresponding channel instead of entire register write. Fixes: commit c5c8635a04711 ("ASoC: qcom: Add LPASS platform driver") Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-platform.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 0df9481ea4c6..f220a2739ac3 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -526,7 +526,7 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, return -EINVAL; } - ret = regmap_write(map, reg_irqclr, val_irqclr); + ret = regmap_update_bits(map, reg_irqclr, val_irqclr, val_irqclr); if (ret) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", ret); return ret; @@ -650,7 +650,7 @@ static irqreturn_t lpass_dma_interrupt_handler( struct lpass_variant *v = drvdata->variant; irqreturn_t ret = IRQ_NONE; int rv; - unsigned int reg = 0, val = 0; + unsigned int reg, val, val_clr, val_mask; struct regmap *map; unsigned int dai_id = cpu_dai->driver->id; @@ -676,8 +676,9 @@ static irqreturn_t lpass_dma_interrupt_handler( return -EINVAL; } if (interrupts & LPAIF_IRQ_PER(chan)) { - - rv = regmap_write(map, reg, LPAIF_IRQ_PER(chan) | val); + val_clr = LPAIF_IRQ_PER(chan) | val; + val_mask = LPAIF_IRQ_ALL(chan); + rv = regmap_update_bits(map, reg, val_mask, val_clr); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); @@ -688,7 +689,9 @@ static irqreturn_t lpass_dma_interrupt_handler( } if (interrupts & LPAIF_IRQ_XRUN(chan)) { - rv = regmap_write(map, reg, LPAIF_IRQ_XRUN(chan) | val); + val_clr = (LPAIF_IRQ_XRUN(chan) | val); + val_mask = LPAIF_IRQ_ALL(chan); + rv = regmap_update_bits(map, reg, val_mask, val_clr); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); @@ -700,7 +703,9 @@ static irqreturn_t lpass_dma_interrupt_handler( } if (interrupts & LPAIF_IRQ_ERR(chan)) { - rv = regmap_write(map, reg, LPAIF_IRQ_ERR(chan) | val); + val_clr = (LPAIF_IRQ_ERR(chan) | val); + val_mask = LPAIF_IRQ_ALL(chan); + rv = regmap_update_bits(map, reg, val_mask, val_clr); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 2

[PATCH] ASoC: Remove unneeded semicolon in rk817_codec.c
by Wan Jiabing 07 Jun '21

07 Jun '21

Fix following coccicheck warning: ./sound/soc/codecs/rk817_codec.c:49:2-3: Unneeded semicolon Signed-off-by: Wan Jiabing <wanjiabing(a)vivo.com> --- sound/soc/codecs/rk817_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index fd3a5ba034a9..414d21f22a05 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -46,7 +46,7 @@ static int rk817_init(struct snd_soc_component *component) if (rk817->mic_in_differential) { snd_soc_component_update_bits(component, RK817_CODEC_AMIC_CFG0, MIC_DIFF_MASK, MIC_DIFF_EN); - }; + } return 0; } -- 2.20.1

3 3

Possible ALSA bug: Cannot record via mic while PA is running
by Meghdad Moradi 07 Jun '21

07 Jun '21

Hello, Please take a look at the following thread that I created some time ago: [1]https://ubuntuforums.org/showthread.php?t=2462882&p=14041574#post140 41574 I have included most relevant and useful info in the first and last post. pa-info output: [2]https://gist.github.com/no149/5c239626048a860a882f8d412013a4e6 I'd provide further info, if it's needed. References 1. https://ubuntuforums.org/showthread.php?t=2462882&p=14041574#post14041574 2. https://gist.github.com/no149/5c239626048a860a882f8d412013a4e6

1 0

[RFC][PATCH 0/3] ALSA: pcm/firewire: allow to queue period elapse event in process context
by Takashi Sakamoto 07 Jun '21

07 Jun '21

Hi, All of drivers in ALSA firewire stack processes two chances to process isochronous packets in any isochronous context; in software IRQ context for 1394 OHCI, and in process context of ALSA PCM application. In the process context, callbacks of .pointer and .ack are utilized. The callbacks are done by ALSA PCM core under acquiring lock of PCM substream, In design of ALSA PCM core, call of snd_pcm_period_elapsed() is used for drivers to awaken user processes from waiting for available frames. The function voluntarily acquires lock of PCM substream, therefore it is not called in the process context since it causes dead lock. As a workaround to avoid the dead lock, all of drivers in ALSA firewire stack uses workqueue to delegate the call. This patchset is my attempt for the issue. A variant of 'snd_pcm_period_elapsed()' without lock acquisition is going to be added, named 'snd_pcm_period_elapsed_without_lock()'. This is used in callbacks of .pointer and .ack of snd_pcm_ops structure. The patchset is still under my test, but it looks to work well in my easy and rough test. Before posting for merge, I'd like to get your comment to the idea. When evaluating, please merge below two histories: * 64584f329352 (for-next) * 9981b20a5e36 (for-linus) Takashi Sakamoto (3): ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream ALSA: firewire-lib: queue event of period elapse in process context ALSA: firewire-lib: obsolete workqueue for period update include/sound/pcm.h | 53 ++++++++++++++++++++++++++++++++++- sound/core/pcm_lib.c | 25 +++-------------- sound/firewire/amdtp-stream.c | 46 +++++++++--------------------- sound/firewire/amdtp-stream.h | 1 - 4 files changed, 70 insertions(+), 55 deletions(-) -- 2.27.0

2 5

New lavc api
by GitHub pull_request - opened 06 Jun '21

06 Jun '21

alsa-project/alsa-plugins pull request #23 was opened from jamrial: This is untested (I don't have access to a system with alsa, but the change was straightforward enough that i could write it). It is unlikely that it will fix the silence issue described in #22, but it nonetheless is a change required in order to support future versions of ffmpeg. The silence is, as far as i could tell, related to the fact that starting with ffmpeg 4.4, the ac3_fixed encoder takes S32 planar as input, versus the S16 planar it used to. This plugin should look at the AVCodec sample_fmts[0] value and feed the encoder the appropriate data. Request URL : https://github.com/alsa-project/alsa-plugins/pull/23 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/23.patch Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH 0/8] ALSA: usb-audio: scarlett2: cleanup
by Geoffrey D. Bennett 06 Jun '21

06 Jun '21

Hi Takashi, Over the next few weeks I will have ready about 40 patches to clean up the Scarlett Gen 2 mixer driver a bit, add a few missing features, and add support for all Scarlett Gen 3 models. I was planning on submitting them in batches of 5-10 at a time; is that okay? Here are the first 8 patches which are ready to go; only small cleanups/renames/fixes. They are intended to be applied after these two patches that I submitted earlier: - ALSA: usb-audio: scarlett2: Read mixer volumes at init time - ALSA: usb-audio: scarlett2: Read mux at init time Thanks, Geoffrey. Geoffrey D. Bennett (8): ALSA: usb-audio: scarlett2: Remove incorrect S/PDIF comment ALSA: usb-audio: scarlett2: Coding style improvements ALSA: usb-audio: scarlett2: Remove unused/useless code ALSA: usb-audio: scarlett2: Remove interrupt debug message ALSA: usb-audio: scarlett2: Rename buttons/interrupts ALSA: usb-audio: scarlett2: Rename struct scarlett2_mixer_data ALSA: usb-audio: scarlett2: Add temp variable for consistency ALSA: usb-audio: scarlett2: Fix data_mutex lock sound/usb/mixer_scarlett_gen2.c | 229 ++++++++++++++------------------ 1 file changed, 102 insertions(+), 127 deletions(-) -- 2.31.1

1 8

a52: not compatible with recent avcodec
by GitHub issues - opened 06 Jun '21

06 Jun '21

alsa-project/alsa-plugins issue #22 was opened from manio: Hi! Current a52 plugin is not compatible with recent changes in the FFmpeg lib. The symptom is that using newest FFmpeg the a52 plugin produces silent in the speakers. During the weekend I was able to bisect the problem and shed some light on this, and here is what I found out: 1. The a52 plugin is working fine before commit: https://github.com/FFmpeg/FFmpeg/commit/93016f5d1d280f9cb7856883af287fa66af… 2. Above commit is wrapping avcodec_encode_audio2(), which a52 plugin is using, into new API internally inside the avcodec 3. All applications should move to new so called ["avcodec_send_frame() API"](https://ffmpeg.org/doxygen/3.4/group__lavc__encdec.html) 4. After the commit in point 1 the a52 output plugin is not working anymore I was asking the commit author @jamrial about the commit and he said that the main culprit could rather be the commit: https://github.com/FFmpeg/FFmpeg/commit/2d85e6e7230406e33e61d903b9c29ab562e… which is changing the S16 to S32 for ac3_fixed encoder while the a52 plugin _"is ignoring the kind of format the encoder takes as input (Starting from the aforementioned commit, ac3_fixed takes S32 planar audio), and unconditionally feeding it S16 planar audio."_ I was trying to change the a52 from S16 to S32 but I think some more work is needed on this... I was also try to change the do_encode() function from avcodec_encode_audio2 to avcodec_send_frame() with no luck... Issue URL : https://github.com/alsa-project/alsa-plugins/issues/22 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] ALSA: firewire-motu: add support for hybrid model of MOTU Ultralite mk3
by Takashi Sakamoto 06 Jun '21

06 Jun '21

This commit adds support for the hybrid model of MOTU Ultralite mk3 with alpha connector, which is already discontinued. The hardware specification of the model is the same as the one of FireWire-only model. $ cd linux-firewire-utils $ python3 src/crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 04101573 bus_info_length 4, crc_length 16, crc 5491 404 31333934 bus_name "1394" 408 20ff7000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256) 40c 0001f200 company_id 0001f2 | 410 000a059c device_id 00000a059c | EUI-64 0001f200000a059c root directory ----------------------------------------------------------------- 414 0004ef04 directory_length 4, crc 61188 418 030001f2 vendor 41c 0c0083c0 node capabilities per IEEE 1394 420 d1000002 --> unit directory at 428 424 8d000005 --> eui-64 leaf at 438 unit directory at 428 ----------------------------------------------------------------- 428 0003f00b directory_length 3, crc 61451 42c 120001f2 specifier id 430 13000030 version 434 17103800 model eui-64 leaf at 438 ----------------------------------------------------------------- 438 0002d89c leaf_length 2, crc 55452 43c 0001f200 company_id 0001f2 | 440 000a059c device_id 00000a059c | EUI-64 0001f200000a059c Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 2 ++ sound/firewire/motu/motu.c | 1 + 2 files changed, 3 insertions(+) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index def1f3d5ecf5..fc6858131b59 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -154,6 +154,8 @@ config SND_FIREWIRE_MOTU * 8pre * 828mk3 (FireWire only) * 828mk3 (Hybrid) + * Ultralite mk3 (FireWire only) + * Ultralite mk3 (Hybrid) * Audio Express * 4pre diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index a4929c1302dc..2a8a6ea2d3f1 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -190,6 +190,7 @@ static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x00000f, &snd_motu_spec_8pre), SND_MOTU_DEV_ENTRY(0x000015, &snd_motu_spec_828mk3), // FireWire only. SND_MOTU_DEV_ENTRY(0x000019, &snd_motu_spec_ultralite_mk3), // FireWire only. + SND_MOTU_DEV_ENTRY(0x000030, &snd_motu_spec_ultralite_mk3), // Hybrid. SND_MOTU_DEV_ENTRY(0x000035, &snd_motu_spec_828mk3), // Hybrid. SND_MOTU_DEV_ENTRY(0x000033, &snd_motu_spec_audio_express), SND_MOTU_DEV_ENTRY(0x000045, &snd_motu_spec_4pre), -- 2.27.0

2 1

[PATCH] ALSA: firewire-lib: remove useless operations for kernel preemption
by Takashi Sakamoto 06 Jun '21

06 Jun '21

In all of drivers of ALSA firewire stack, the callback of .pointer and .ack in snd_pcm_ops structure is done in acquired spin_lock of PCM substream, therefore already under disabled kernel preemption. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 945597ffacc2..1c566ddf0776 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -1751,13 +1751,8 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d, // Later, the process context will sometimes schedules software // IRQ context of the period_work. Then, no need to flush the // queue by the same reason as described in the above - if (current_work() != &s->period_work) { - // Queued packet should be processed without any kernel - // preemption to keep latency against bus cycle. - preempt_disable(); + if (current_work() != &s->period_work) fw_iso_context_flush_completions(irq_target->context); - preempt_enable(); - } } return READ_ONCE(s->pcm_buffer_pointer); @@ -1777,13 +1772,8 @@ int amdtp_domain_stream_pcm_ack(struct amdtp_domain *d, struct amdtp_stream *s) // Process isochronous packets for recent isochronous cycle to handle // queued PCM frames. - if (irq_target && amdtp_stream_running(irq_target)) { - // Queued packet should be processed without any kernel - // preemption to keep latency against bus cycle. - preempt_disable(); + if (irq_target && amdtp_stream_running(irq_target)) fw_iso_context_flush_completions(irq_target->context); - preempt_enable(); - } return 0; } -- 2.27.0

2 1

[PATCH] ALSA: firewire-lib: fix error codes for allocation failure
by Dan Carpenter 06 Jun '21

06 Jun '21

Return -ENOMEM if kcalloc() fails. Currently the code returns success. Fixes: f9e5ecdfc2c2 ("ALSA: firewire-lib: add replay target to cache sequence of packet") Fixes: 6f24bb8a157c ("ALSA: firewire-lib: pool sequence of packet in IT context independently") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/firewire/amdtp-stream.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 945597ffacc2..b37cec3cc579 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -1625,8 +1625,10 @@ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed, s->ctx_data.tx.cache.tail = 0; s->ctx_data.tx.cache.descs = kcalloc(s->ctx_data.tx.cache.size, sizeof(*s->ctx_data.tx.cache.descs), GFP_KERNEL); - if (!s->ctx_data.tx.cache.descs) + if (!s->ctx_data.tx.cache.descs) { + err = -ENOMEM; goto err_context; + } } } else { static const struct { @@ -1643,8 +1645,10 @@ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed, }; s->ctx_data.rx.seq.descs = kcalloc(queue_size, sizeof(*s->ctx_data.rx.seq.descs), GFP_KERNEL); - if (!s->ctx_data.rx.seq.descs) + if (!s->ctx_data.rx.seq.descs) { + err = -ENOMEM; goto err_context; + } s->ctx_data.rx.seq.size = queue_size; s->ctx_data.rx.seq.tail = 0; s->ctx_data.rx.seq.head = 0; -- 2.30.2

3 2

amixer and alsamixer unaware of controls for ThinkPad T14's 'acp' card
by GitHub issues - opened 06 Jun '21

06 Jun '21

alsa-project/alsa-lib issue #146 was opened from jwatt: I have the AMD Ryzen 7 Pro 4750U version of the [ThinkPad T14 gen 1 type 20UD](https://pcsupport.lenovo.com/gb/en/products/laptops-and-netbooks/thin… ([tech specs](https://www.lenovo.com/gb/en/laptops/thinkpad/t-series/ThinkPad-T14-…) with Fedora 34 installed. I am unable to control the built-in microphone using `amixer`/`alsamixer`. ``` $ cat /proc/asound/modules 0 snd_hda_intel 1 snd_hda_intel 2 snd_acp3x_rn ``` As best I can tell (I could be mistaken!), 0 is the HDMI port, 1 is the build-in speakers and headphones jack, and 2 is the Ryzen "Audio Co-Processor" (i.e. "acp" / "AMD TrueAudio") which is (or includes) the built-in microphone. To be specific, I only get microphone input if I select 'acp' as the input. For the 'acp' card, AlsaMixer v1.2.4 reports "This sound device does not have any controls." It appears support for the Ryzen 4000 mobile series Audio Co-Processor was added in kernel 5.8 (see the section "Add Renoir ACP driver" on the [5.8 release notes](https://kernelnewbies.org/Linux_5.8#Audio)). If the kernel supports it, it's unclear to me why the ALSA tools wouldn't allow me to control it. Is this currently expected? I'm happy to provide any other information that would be helpful - just let me know what you need. Issue URL : https://github.com/alsa-project/alsa-lib/issues/146 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

alsa-ucm-conf 1.2.5 broke front headphone output on pipewire-pulse
by GitHub issues - opened 06 Jun '21

06 Jun '21

alsa-project/alsa-ucm-conf issue #98 was opened from amezin: After upgrade to 1.2.5, front headphone output isn't working. In GNOME Settings, when I select front headphone output and click "Test", it shows a dialog with text "Click a speaker to test", but there are no buttons to click. `systemctl --user status pipewire` shows this: Jun 06 04:40:30 trx40 systemd[2732]: Started Multimedia Service. Jun 06 04:40:41 trx40 pipewire[44766]: '_ucm0005.hw:ALC1220VBDT,2': playback open failed: No such device Jun 06 04:40:41 trx40 pipewire[44766]: adapter 0x559741478670: can't get format: No such device Jun 06 04:40:41 trx40 pipewire[44766]: usage: node.name=<string> Jun 06 04:40:41 trx40 pipewire[44766]: '_ucm0005.hw:ALC1220VBDT,1': playback open failed: No such device Jun 06 04:40:41 trx40 pipewire[44766]: adapter 0x559741530560: can't get format: No such device Jun 06 04:40:41 trx40 pipewire[44766]: usage: node.name=<string> Downgrading alsa-ucm-conf to 1.2.4 fixes the issue Tried pipewire 0.3.29 and pipewire git master Though with pulseaudio 14.2 the output works Motherboard: ASUS ROG Strix TRX40-E With alsa-ucm-conf 1.2.4 - output working: alsa-info.sh output: [alsa-info-good.txt](https://github.com/alsa-project/alsa-ucm-conf/files/660… pw-dump output: [pw-dump-good.txt](https://github.com/alsa-project/alsa-ucm-conf/files/66033… With alsa-ucm-conf 1.2.5 - output not working: alsa-info.sh output: [alsa-info-bad.txt](https://github.com/alsa-project/alsa-ucm-conf/files/6603… pw-dump output: [pw-dump-bad.txt](https://github.com/alsa-project/alsa-ucm-conf/files/660336… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/98 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: firewire-lib: fix the context to call snd_pcm_stop_xrun()
by Takashi Sakamoto 05 Jun '21

05 Jun '21

In the workqueue to queue wake-up event, isochronous context is not processed, thus it's useless to check context for the workqueue to switch status of runtime for PCM substream to XRUN. On the other hand, in software IRQ context of 1394 OHCI, it's needed. This commit fixes the bug introduced when tasklet was replaced with workqueue. Cc: <stable(a)vger.kernel.org> Fixes: 2b3d2987d800 ("ALSA: firewire: Replace tasklet with work") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 945597ffacc2..19c343c53585 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -1032,7 +1032,7 @@ static void generate_pkt_descs(struct amdtp_stream *s, const __be32 *ctx_header, static inline void cancel_stream(struct amdtp_stream *s) { s->packet_index = -1; - if (current_work() == &s->period_work) + if (in_interrupt()) amdtp_stream_pcm_abort(s); WRITE_ONCE(s->pcm_buffer_pointer, SNDRV_PCM_POS_XRUN); } -- 2.27.0

2 1

[PATCH 1/3] ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP Elite Dragonfly G2
by Jeremy Szu 05 Jun '21

05 Jun '21

The HP Elite Dragonfly G2 using ALC285 codec which using 0x04 to control mute LED and 0x01 to control micmute LED. In the other hand, there is no output from right channel of speaker. Therefore, add a quirk to make it works. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 43e37145eb5d..9f65171a902d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8310,6 +8310,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360), SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO), + SND_PCI_QUIRK(0x103c, 0x8716, "HP Elite Dragonfly G2 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8730, "HP ProBook 445 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), -- 2.31.1

2 5

[PATCH] ALSA: hda/realtek: Change device names for quirks to barebone names
by Werner Sembach 05 Jun '21

05 Jun '21

Change the name string of several devices needing quirks to the Clevo-barebone ones. Also make the names follow the same pattern for multiple Clevo names referring to the same mainboard. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> --- sound/pci/hda/patch_realtek.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 552e2cb73291..39cf3667118c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8368,7 +8368,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x152d, 0x1082, "Quanta NL3", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x1558, 0x1323, "Clevo N130ZU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1558, 0x1325, "System76 Darter Pro (darp5)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x1325, "Clevo N15[01][CW]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1401, "Clevo L140[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1403, "Clevo N140CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1404, "Clevo N150CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), @@ -8403,11 +8403,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x8521, "Clevo NH77D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8535, "Clevo NH50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8536, "Clevo NH79D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1558, 0x8550, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1558, 0x8551, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), - SND_PCI_QUIRK(0x1558, 0x8561, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), - SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[5|7][0-9]RZ[Q]", ALC269_FIXUP_DMIC), + SND_PCI_QUIRK(0x1558, 0x8550, "Clevo NH[57][0-9][ER][ACDH]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8551, "Clevo NH[57][0-9][ER][ACDH]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8560, "Clevo NH[57][0-9][ER][ACDH]Q", ALC269_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1558, 0x8561, "Clevo NH[57][0-9][ER][ACDH]Q", ALC269_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[57][0-9]RZ[Q]", ALC269_FIXUP_DMIC), SND_PCI_QUIRK(0x1558, 0x8668, "Clevo NP50B[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8680, "Clevo NJ50LU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), @@ -8418,8 +8418,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x9600, "Clevo N960K[PR]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x961d, "Clevo N960S[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x971d, "Clevo N970T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL53RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1558, 0xa600, "Clevo NL5XNU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL5[03]RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0xa600, "Clevo NL50NU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb022, "Clevo NH77D[DC][QW]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.25.1

2 1

[PATCH] ASoC: qcom: Fix for DMA interrupt clear reg overwriting
by Srinivasa Rao Mandadapu 05 Jun '21

05 Jun '21

This patch fixes the DMA interrupt registers overwriting issue in lpass platform interrupt handler. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-platform.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 0df9481ea4c6..e02caa121fa4 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -526,7 +526,7 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, return -EINVAL; } - ret = regmap_write(map, reg_irqclr, val_irqclr); + ret = regmap_update_bits(map, reg_irqclr, val_irqclr, val_irqclr); if (ret) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", ret); return ret; @@ -650,7 +650,7 @@ static irqreturn_t lpass_dma_interrupt_handler( struct lpass_variant *v = drvdata->variant; irqreturn_t ret = IRQ_NONE; int rv; - unsigned int reg = 0, val = 0; + unsigned int reg = 0, val = 0, val_clr = 0, val_mask = 0; struct regmap *map; unsigned int dai_id = cpu_dai->driver->id; @@ -676,8 +676,9 @@ static irqreturn_t lpass_dma_interrupt_handler( return -EINVAL; } if (interrupts & LPAIF_IRQ_PER(chan)) { - - rv = regmap_write(map, reg, LPAIF_IRQ_PER(chan) | val); + val_clr = (LPAIF_IRQ_PER(chan) | val); + val_mask = LPAIF_IRQ_ALL(chan); + rv = regmap_update_bits(map, reg, val_mask, val_clr); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); @@ -688,7 +689,9 @@ static irqreturn_t lpass_dma_interrupt_handler( } if (interrupts & LPAIF_IRQ_XRUN(chan)) { - rv = regmap_write(map, reg, LPAIF_IRQ_XRUN(chan) | val); + val_clr = (LPAIF_IRQ_XRUN(chan) | val); + val_mask = LPAIF_IRQ_ALL(chan); + rv = regmap_update_bits(map, reg, val_mask, val_clr); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); @@ -700,7 +703,9 @@ static irqreturn_t lpass_dma_interrupt_handler( } if (interrupts & LPAIF_IRQ_ERR(chan)) { - rv = regmap_write(map, reg, LPAIF_IRQ_ERR(chan) | val); + val_clr = (LPAIF_IRQ_ERR(chan) | val); + val_mask = LPAIF_IRQ_ALL(chan); + rv = regmap_update_bits(map, reg, val_mask, val_clr); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 2

[PATCH] ASoC: rt5682: Fix the fast discharge for headset unplugging in soundwire mode
by Oder Chiou 04 Jun '21

04 Jun '21

Based on ("5a15cd7fce20b1fd4aece6a0240e2b58cd6a225d"), the setting also should be set in soundwire mode. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5682-sdw.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index fed80c8f994f..e78ba3b064c4 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -462,7 +462,8 @@ static int rt5682_io_init(struct device *dev, struct sdw_slave *slave) regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2, RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); - regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd042); + regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd142); + regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_5, 0x0700, 0x0600); regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_3, RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN); regmap_update_bits(rt5682->regmap, RT5682_SAR_IL_CMD_1, -- 2.31.1

2 2

[PATCH v2][next] ASoC: ti: davinci-mcasp: Fix fall-through warning for Clang
by Gustavo A. R. Silva 04 Jun '21

04 Jun '21

In preparation to enable -Wimplicit-fallthrough for Clang, fix a warning by explicitly adding a fallthrough; statement. Link: https://github.com/KSPP/linux/issues/115 Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- Changes in v2: - Use fallthrough; instead of break; - Update changelog text. Link: https://lore.kernel.org/linux-hardening/25941c37-6e38-88ae-3afe-4f5af44380d… sound/soc/ti/davinci-mcasp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index b94220306d1a..017a5a5e56cd 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2317,6 +2317,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) break; default: dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret); + fallthrough; case -EPROBE_DEFER: goto err; } -- 2.27.0

4 3

[PATCH] ASoC: snd-soc-lpass requires REGMAP_MMIO
by Bertrand Jacquin 04 Jun '21

04 Jun '21

With CONFIG_SND_SOC_LPASS_RX_MACRO=m and CONFIG_REGMAP_MMIO undefined, build fails with the following error make -f /var/tmp/portage/sys-kernel/stable-sources-5.12.8/work/linux-5.12.8-stable/scripts/Makefile.modpost sed 's/\.ko$/\.o/' modules.order | scripts/mod/modpost -E -o modules-only.symvers -i vmlinux.symvers -T - ERROR: modpost: "__devm_regmap_init_mmio_clk" [sound/soc/codecs/snd-soc-lpass-rx-macro.ko] undefined! This does also apply to other Qualcomm Macro LPASS all making call to devm_regmap_init_mmio() Signed-off-by: Bertrand Jacquin <bertrand(a)jacquin.bzh> --- sound/soc/codecs/Kconfig | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1c87b42606c9..f5614f47f44d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1824,18 +1824,22 @@ config SND_SOC_TPA6130A2 config SND_SOC_LPASS_WSA_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm WSA Macro in LPASS(Low Power Audio SubSystem)" config SND_SOC_LPASS_VA_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm VA Macro in LPASS(Low Power Audio SubSystem)" config SND_SOC_LPASS_RX_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm RX Macro in LPASS(Low Power Audio SubSystem)" config SND_SOC_LPASS_TX_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm TX Macro in LPASS(Low Power Audio SubSystem)" endmenu

2 1

[PATCH -next] ASoC: rk817: fix compile warning in rk817_probe()
by Yang Yingliang 04 Jun '21

04 Jun '21

Fix the compile warning by return ret: sound/soc/codecs/rk817_codec.c:418:6: warning: variable ‘ret’ set but not used [-Wunused-but-set-variable] 418 | int ret; | ^~~ Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/codecs/rk817_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 17e672b85ee5..56de5ec5f03b 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -430,7 +430,7 @@ static int rk817_probe(struct snd_soc_component *component) snd_soc_component_set_pll(component, 0, 0, 0, 0); - return 0; + return ret; } static void rk817_remove(struct snd_soc_component *component) -- 2.25.1

2 1

[v6] ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
by Judy Hsiao 04 Jun '21

04 Jun '21

From: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> This patch fixes PoP noise of around 15ms observed during audio capture begin. Enables BCLK and LRCLK in snd_soc_dai_ops prepare call for introducing some delay before capture start. Co-developed-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> (am from https://patchwork.kernel.org/patch/12276369/) (also found at https://lore.kernel.org/r/20210524142114.18676-1-srivasam@codeaurora.org) --- Changes Since V5: -- Fixed nit. -- Updated `mi2s_was_prepared[dai->driver->id] = true;` after checking for errors. Changes Since V4: -- Replaced the __clk_is_enabled(BCLK) check by the self maintained. mi2s_was_prepared bool state. -- Removed unrelated changes. -- Refined comments. Changes Since V3: -- Checked BCLK is off before enabling it in lpass_cpu_daiops_prepare as lpass_cpu_daiops_prepare can be called multiple times -- Checked BCLK is on before disabling it in lpass_cpu_daiops_shutdown to fix the WARN. It is because BCLK may not be enabled if lpass_cpu_daiops_prepare is not called before lpass_cpu_daiops_shutdown -- Added more comments Changes Since V2: -- Updated comments as per linux style -- Removed unrelated changes Changes Since V1: -- Enabled BCLK and LRCLK in dai ops prepare API instead of startup API -- Added comments sound/soc/qcom/lpass-cpu.c | 80 ++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/lpass.h | 4 ++ 2 files changed, 84 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index af8cb64924a0..0430cc797391 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -93,8 +93,30 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + /* + * Ensure LRCLK is disabled even in device node validation. + * Will not impact if disabled in lpass_cpu_daiops_trigger() + * suspend. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); + else + regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE); + + /* + * BCLK may not be enabled if lpass_cpu_daiops_prepare is called before + * lpass_cpu_daiops_shutdown. It's paired with the clk_enable in + * lpass_cpu_daiops_prepare. + */ + if (drvdata->mi2s_was_prepared[dai->driver->id]) { + drvdata->mi2s_was_prepared[dai->driver->id] = false; + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); + } + clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } @@ -275,6 +297,18 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * Ensure lpass BCLK/LRCLK is enabled during + * device resume as lpass_cpu_daiops_prepare() is not called + * after the device resumes. We don't check mi2s_was_prepared before + * enable/disable BCLK in trigger events because: + * 1. These trigger events are paired, so the BCLK + * enable_count is balanced. + * 2. the BCLK can be shared (ex: headset and headset mic), + * we need to increase the enable_count so that we don't + * turn off the shared BCLK while other devices are using + * it. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); @@ -296,6 +330,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* + * To ensure lpass BCLK/LRCLK is disabled during + * device suspend. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); @@ -315,12 +353,54 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, return ret; } +static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret; + + /* + * Ensure lpass BCLK/LRCLK is enabled bit before playback/capture + * data flow starts. This allows other codec to have some delay before + * the data flow. + * (ex: to drop start up pop noise before capture starts). + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); + else + ret = regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_ENABLE); + + if (ret) { + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + return ret; + } + + /* + * Check mi2s_was_prepared before enabling BCLK as lpass_cpu_daiops_prepare can + * be called multiple times. It's paired with the clk_disable in + * lpass_cpu_daiops_shutdown. + */ + if (!drvdata->mi2s_was_prepared[dai->driver->id]) { + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + clk_disable(drvdata->mi2s_osr_clk[id]); + return ret; + } + drvdata->mi2s_was_prepared[dai->driver->id] = true; + } + return 0; +} + const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, .trigger = lpass_cpu_daiops_trigger, + .prepare = lpass_cpu_daiops_prepare, }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..7f72214404ba 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -67,6 +67,10 @@ struct lpass_data { /* MI2S SD lines to use for playback/capture */ unsigned int mi2s_playback_sd_mode[LPASS_MAX_MI2S_PORTS]; unsigned int mi2s_capture_sd_mode[LPASS_MAX_MI2S_PORTS]; + + /* The state of MI2S prepare dai_ops was called */ + bool mi2s_was_prepared[LPASS_MAX_MI2S_PORTS]; + int hdmi_port_enable; /* low-power audio interface (LPAIF) registers */ -- 2.32.0.rc1.229.g3e70b5a671-goog

2 1

hda: improve the microphone device allocation
by GitHub pull_request - opened 04 Jun '21

04 Jun '21

alsa-project/alsa-ucm-conf pull request #97 was opened from perexg: This change use the local variables for the UCM microphone device names. Also, handle the no-dmic case for sof-hda-dsp hardware. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/97 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/97.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH][next] ASoC: ti: davinci-mcasp: Fix fall-through warning for Clang
by Gustavo A. R. Silva 04 Jun '21

04 Jun '21

In preparation to enable -Wimplicit-fallthrough for Clang, fix a warning by explicitly adding a goto statement instead of letting the code fall through to the next case. Link: https://github.com/KSPP/linux/issues/115 Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- JFYI: We had thousands of these sorts of warnings and now we are down to just 25 in linux-next. This is one of those last remaining warnings. sound/soc/ti/davinci-mcasp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index b94220306d1a..587967720135 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2317,6 +2317,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) break; default: dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret); + goto err; case -EPROBE_DEFER: goto err; } -- 2.27.0

4 3

[PATCH -next] ALSA: firewire-motu: fix error return code in snd_motu_stream_reserve_duplex()
by Yang Yingliang 04 Jun '21

04 Jun '21

Fix to return a negative error code from the error handling case instead of 0, as done elsewhere in this function. Fixes: e50dfac81f73 ("ALSA: firewire-motu: cache event ticks in source packet header per data block") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/firewire/motu/motu-stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/motu/motu-stream.c b/sound/firewire/motu/motu-stream.c index 43ff5be32b15..9e6ca39ebd7f 100644 --- a/sound/firewire/motu/motu-stream.c +++ b/sound/firewire/motu/motu-stream.c @@ -191,7 +191,7 @@ int snd_motu_stream_reserve_duplex(struct snd_motu *motu, unsigned int rate, if (!motu->cache.event_offsets) { fw_iso_resources_free(&motu->tx_resources); fw_iso_resources_free(&motu->rx_resources); - return err; + return -ENOMEM; } } -- 2.25.1

3 2

[v5] ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
by Judy Hsiao 04 Jun '21

04 Jun '21

From: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> This patch fixes PoP noise of around 15ms observed during audio capture begin. Enables BCLK and LRCLK in snd_soc_dai_ops prepare call for introducing some delay before capture start. Co-developed-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> (am from https://patchwork.kernel.org/patch/12276369/) (also found at https://lore.kernel.org/r/20210524142114.18676-1-srivasam@codeaurora.org) --- Changes Since V4: -- Replaced the __clk_is_enabled(BCLK) check by the self maintained mi2s_was_prepared bool state. -- Removed unrelated changes. -- Refined comments. Changes Since V3: -- Checked BCLK is off before enabling it in lpass_cpu_daiops_prepare as lpass_cpu_daiops_prepare can be called multiple times -- Checked BCLK is on before disabling it in lpass_cpu_daiops_shutdown to fix the WARN. It is because BCLK may not be enabled if lpass_cpu_daiops_prepare is not called before lpass_cpu_daiops_shutdown -- Added more comments Changes Since V2: -- Updated comments as per linux style -- Removed unrelated changes Changes Since V1: -- Enabled BCLK and LRCLK in dai ops prepare API instead of startup API -- Added comments sound/soc/qcom/lpass-cpu.c | 80 ++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/lpass.h | 4 ++ 2 files changed, 84 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index af8cb64924a0..3129e7cc25a4 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -93,8 +93,30 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + /* + * Ensure LRCLK is disabled even in device node validation. + * Will not impact if disabled in lpass_cpu_daiops_trigger() + * suspend. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); + else + regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE); + + /* + * BCLK may not be enabled if lpass_cpu_daiops_prepare is called before + * lpass_cpu_daiops_shutdown. It's paired with the clk_enable in + * lpass_cpu_daiops_prepare. + */ + if (drvdata->mi2s_was_prepared[dai->driver->id]) { + drvdata->mi2s_was_prepared[dai->driver->id] = false; + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); + } + clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } @@ -275,6 +297,18 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * Ensure lpass BCLK/LRCLK is enabled during + * device resume as lpass_cpu_daiops_prepare() is not called + * after the device resumes. We don't check mi2s_was_prepared before + * enable/disable BCLK in trigger events because: + * 1. These trigger events are paired, so the BCLK + * enable_count is balanced. + * 2. the BCLK can be shared (ex: headset and headset mic), + * we need to increase the enable_count so that we don't + * turn off the shared BCLK while other devices are using + * it. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); @@ -296,6 +330,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* + * To ensure lpass BCLK/LRCLK is disabled during + * device suspend. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); @@ -315,12 +353,54 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, return ret; } +static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret; + /* + * Ensure lpass BCLK/LRCLK is enabled bit before playback/capture + * data flow starts. This allows other codec to have some delay before + * the data flow. + * (ex: to drop start up pop noise before capture starts). + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); + else + ret = regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_ENABLE); + + if (ret) { + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + return ret; + } + + /* + * Check mi2s_was_prepared before enabling BCLK as lpass_cpu_daiops_prepare can + * be called multiple times. It's paired with the clk_disable in + * lpass_cpu_daiops_shutdown. + */ + if (!drvdata->mi2s_was_prepared[dai->driver->id]) { + drvdata->mi2s_was_prepared[dai->driver->id] = true; + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + clk_disable(drvdata->mi2s_osr_clk[id]); + drvdata->mi2s_was_prepared[dai->driver->id] = false; + return ret; + } + } + return 0; +} + const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, .trigger = lpass_cpu_daiops_trigger, + .prepare = lpass_cpu_daiops_prepare, }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..7f72214404ba 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -67,6 +67,10 @@ struct lpass_data { /* MI2S SD lines to use for playback/capture */ unsigned int mi2s_playback_sd_mode[LPASS_MAX_MI2S_PORTS]; unsigned int mi2s_capture_sd_mode[LPASS_MAX_MI2S_PORTS]; + + /* The state of MI2S prepare dai_ops was called */ + bool mi2s_was_prepared[LPASS_MAX_MI2S_PORTS]; + int hdmi_port_enable; /* low-power audio interface (LPAIF) registers */ -- 2.31.0

3 2

[PATCH] ASoC: snd-soc-lpass requires REGMAP_MMIO
by Bertrand Jacquin 04 Jun '21

04 Jun '21

With CONFIG_SND_SOC_LPASS_RX_MACRO=m and CONFIG_REGMAP_MMIO undefined, build fails with the following error make -f /var/tmp/portage/sys-kernel/stable-sources-5.12.8/work/linux-5.12.8-stable/scripts/Makefile.modpost sed 's/\.ko$/\.o/' modules.order | scripts/mod/modpost -E -o modules-only.symvers -i vmlinux.symvers -T - ERROR: modpost: "__devm_regmap_init_mmio_clk" [sound/soc/codecs/snd-soc-lpass-rx-macro.ko] undefined! This does also apply to other Qualcomm Macro LPASS all making call to devm_regmap_init_mmio() --- sound/soc/codecs/Kconfig | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1c87b42606c9..f5614f47f44d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1824,18 +1824,22 @@ config SND_SOC_TPA6130A2 config SND_SOC_LPASS_WSA_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm WSA Macro in LPASS(Low Power Audio SubSystem)" config SND_SOC_LPASS_VA_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm VA Macro in LPASS(Low Power Audio SubSystem)" config SND_SOC_LPASS_RX_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm RX Macro in LPASS(Low Power Audio SubSystem)" config SND_SOC_LPASS_TX_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm TX Macro in LPASS(Low Power Audio SubSystem)" endmenu

2 2

[PATCH] ASoC: img: Fix PM reference leak in img_i2s_in_probe()
by Yufen Yu 03 Jun '21

03 Jun '21

pm_runtime_get_sync will increment pm usage counter even it failed. Forgetting to putting operation will result in reference leak here. Fix it by replacing it with pm_runtime_resume_and_get to keep usage counter balanced. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yufen Yu <yuyufen(a)huawei.com> --- sound/soc/img/img-i2s-in.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index 0843235d73c9..fd3432a1d6ab 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -464,7 +464,7 @@ static int img_i2s_in_probe(struct platform_device *pdev) if (ret) goto err_pm_disable; } - ret = pm_runtime_get_sync(&pdev->dev); + ret = pm_runtime_resume_and_get(&pdev->dev); if (ret < 0) goto err_suspend; -- 2.25.4

2 1

[PATCH] ASoC: AMD Renoir: Remove fix for DMI entry on Lenovo 2020 platforms
by Mark Pearson 03 Jun '21

03 Jun '21

Unfortunately the previous patch to fix issues using the AMD ACP bridge has the side effect of breaking the dmic in other cases and needs to be reverted. Removing the changes while we revisit the fix and find something better. Apologies for the churn. Suggested-by: Gabriel Craciunescu <unix.or.die(a)gmail.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 35 ----------------------------- 1 file changed, 35 deletions(-) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index c9fb1c8fbf8c..19438da5dfa5 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -199,41 +199,6 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), } }, - { - /* Lenovo ThinkPad P14s Gen 1 (20Y1) */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20Y1"), - } - }, - { - /* Lenovo ThinkPad T14s Gen1 */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UH"), - } - }, - { - /* Lenovo ThinkPad T14s Gen1 Campus */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UJ"), - } - }, - { - /* Lenovo ThinkPad T14 Gen 1*/ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UD"), - } - }, - { - /* Lenovo ThinkPad X13 Gen 1*/ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UF"), - } - }, {} }; -- 2.31.1

2 1

[PATCH v2] ASoC: dt-bindings: Convert imx-audmux binding to json schema
by Oleksij Rempel 03 Jun '21

03 Jun '21

Convert the imx-audmux binding to DT schema format using json-schema Signed-off-by: Oleksij Rempel <o.rempel(a)pengutronix.de> Reviewed-by: Rob Herring <robh(a)kernel.org> --- changes v2: - fix yamllint warnings .../devicetree/bindings/sound/imx-audmux.txt | 28 ----- .../devicetree/bindings/sound/imx-audmux.yaml | 119 ++++++++++++++++++ 2 files changed, 119 insertions(+), 28 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/imx-audmux.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audmux.yaml diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.txt b/Documentation/devicetree/bindings/sound/imx-audmux.txt deleted file mode 100644 index 2db4dcbee1b9..000000000000 --- a/Documentation/devicetree/bindings/sound/imx-audmux.txt +++ /dev/null @@ -1,28 +0,0 @@ -Freescale Digital Audio Mux (AUDMUX) device - -Required properties: - - - compatible : "fsl,imx21-audmux" for AUDMUX version firstly used - on i.MX21, or "fsl,imx31-audmux" for the version - firstly used on i.MX31. - - - reg : Should contain AUDMUX registers location and length. - -An initial configuration can be setup using child nodes. - -Required properties of optional child nodes: - - - fsl,audmux-port : Integer of the audmux port that is configured by this - child node. - - - fsl,port-config : List of configuration options for the specific port. - For imx31-audmux and above, it is a list of tuples - <ptcr pdcr>. For imx21-audmux it is a list of pcr - values. - -Example: - -audmux@21d8000 { - compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; - reg = <0x021d8000 0x4000>; -}; diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.yaml b/Documentation/devicetree/bindings/sound/imx-audmux.yaml new file mode 100644 index 000000000000..dab45c310670 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audmux.yaml @@ -0,0 +1,119 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/imx-audmux.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Digital Audio Mux device + +maintainers: + - Oleksij Rempel <o.rempel(a)pengutronix.de> + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx27-audmux + - const: fsl,imx21-audmux + - items: + - enum: + - fsl,imx25-audmux + - fsl,imx35-audmux + - fsl,imx50-audmux + - fsl,imx51-audmux + - fsl,imx53-audmux + - fsl,imx6q-audmux + - fsl,imx6sl-audmux + - fsl,imx6sll-audmux + - fsl,imx6sx-audmux + - const: fsl,imx31-audmux + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: audmux + +patternProperties: + "^mux-[0-9a-z]*$": + type: object + properties: + fsl,audmux-port: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Integer of the audmux port that is configured by this child node + + fsl,port-config: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + List of configuration options for the specific port. + For imx31-audmux and above, it is a list of tuples ptcr pdcr. + For imx21-audmux it is a list of pcr values. + + required: + - fsl,audmux-port + - fsl,port-config + + additionalProperties: false + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + audmux@21d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; + }; + - | + audmux@10016000 { + compatible = "fsl,imx27-audmux", "fsl,imx21-audmux"; + reg = <0x10016000 0x1000>; + clocks = <&clks 1>; + clock-names = "audmux"; + + mux-ssi0 { + fsl,audmux-port = <0>; + fsl,port-config = <0xcb205000>; + }; + + mux-pins4 { + fsl,audmux-port = <2>; + fsl,port-config = <0x00001000>; + }; + }; + - | + #include <dt-bindings/sound/fsl-imx-audmux.h> + audmux@21d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_audmux>; + + mux-ssi1 { + fsl,audmux-port = <0>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN 0 + IMX_AUDMUX_V2_PTCR_TFSEL(2) 0 + IMX_AUDMUX_V2_PTCR_TCSEL(2) 0 + IMX_AUDMUX_V2_PTCR_TFSDIR 0 + IMX_AUDMUX_V2_PTCR_TCLKDIR IMX_AUDMUX_V2_PDCR_RXDSEL(2) + >; + }; + + mux-pins3 { + fsl,audmux-port = <2>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN IMX_AUDMUX_V2_PDCR_RXDSEL(0) + 0 IMX_AUDMUX_V2_PDCR_TXRXEN + >; + }; + }; -- 2.29.2

2 1

[PATCH] ASoC: tas2562: Fix TDM_CFG0_SAMPRATE values
by Richard Weinberger 03 Jun '21

03 Jun '21

TAS2562_TDM_CFG0_SAMPRATE_MASK starts at bit 1, not 0. So all values need to be left shifted by 1. Signed-off-by: Richard Weinberger <richard(a)nod.at> --- sound/soc/codecs/tas2562.h | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 81866aeb3fbf..55b2a1f52ca3 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -57,13 +57,13 @@ #define TAS2562_TDM_CFG0_RAMPRATE_MASK BIT(5) #define TAS2562_TDM_CFG0_RAMPRATE_44_1 BIT(5) #define TAS2562_TDM_CFG0_SAMPRATE_MASK GENMASK(3, 1) -#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ 0x0 -#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ 0x1 -#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ 0x2 -#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ 0x3 -#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ 0x4 -#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ 0x5 -#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ 0x6 +#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ (0x0 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ (0x1 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ (0x2 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ (0x3 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ (0x4 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ (0x5 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ (0x6 << 1) #define TAS2562_TDM_CFG2_RIGHT_JUSTIFY BIT(6) -- 2.26.2

2 1

[PATCH] ASoC: meson: gx-card: fix sound-dai dt schema
by Jerome Brunet 03 Jun '21

03 Jun '21

There is a fair amount of warnings when running 'make dtbs_check' with amlogic,gx-sound-card.yaml. Ex: arch/arm64/boot/dts/amlogic/meson-gxm-q200.dt.yaml: sound: dai-link-0:sound-dai:0:1: missing phandle tag in 0 arch/arm64/boot/dts/amlogic/meson-gxm-q200.dt.yaml: sound: dai-link-0:sound-dai:0:2: missing phandle tag in 0 arch/arm64/boot/dts/amlogic/meson-gxm-q200.dt.yaml: sound: dai-link-0:sound-dai:0: [66, 0, 0] is too long The reason is that the sound-dai phandle provided has cells, and in such case the schema should use 'phandle-array' instead of 'phandle'. Fixes: fd00366b8e41 ("ASoC: meson: gx: add sound card dt-binding documentation") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- .../devicetree/bindings/sound/amlogic,gx-sound-card.yaml | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml index db61f0731a20..2e35aeaa8781 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml @@ -57,7 +57,7 @@ patternProperties: rate sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle + $ref: /schemas/types.yaml#/definitions/phandle-array description: phandle of the CPU DAI patternProperties: @@ -71,7 +71,7 @@ patternProperties: properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle + $ref: /schemas/types.yaml#/definitions/phandle-array description: phandle of the codec DAI required: -- 2.32.0.rc0

2 1

[PATCH 0/5] ASoC: rsnd: tidyup adg and header
by Kuninori Morimoto 03 Jun '21

03 Jun '21

Hi Mark I noticed that adg can be more clean code. And rsnd.h header comment was not so good because patch has been randomly added. This patch tidyup these. Kuninori Morimoto (5): ASoC: rsnd: adg: supply __printf(x, y) formatting for dbg_msg() ASoC: rsnd: adg: tidyup rsnd_adg_get_clkin/out() parameter ASoC: rsnd: adg: use more simple method for null_clk ASoC: rsnd: adg: check return value for rsnd_adg_get_clkin/out() ASoC: rsnd: tidyup __rsnd_mod_xxx macro comments sound/soc/sh/rcar/adg.c | 139 +++++++++++++++++++++++++-------------- sound/soc/sh/rcar/rsnd.h | 21 ++---- 2 files changed, 97 insertions(+), 63 deletions(-) -- 2.25.1

2 6

[PATCH AUTOSEL 4.4 02/15] ASoC: sti-sas: add missing MODULE_DEVICE_TABLE
by Sasha Levin 03 Jun '21

03 Jun '21

From: Zou Wei <zou_wei(a)huawei.com> [ Upstream commit e072b2671606c77538d6a4dd5dda80b508cb4816 ] This patch adds missing MODULE_DEVICE_TABLE definition which generates correct modalias for automatic loading of this driver when it is built as an external module. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zou Wei <zou_wei(a)huawei.com> Link: https://lore.kernel.org/r/1620789145-14936-1-git-send-email-zou_wei@huawei.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/sti-sas.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index 160d61a66204..71a1fde5a7ef 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -542,6 +542,7 @@ static const struct of_device_id sti_sas_dev_match[] = { }, {}, }; +MODULE_DEVICE_TABLE(of, sti_sas_dev_match); static int sti_sas_driver_probe(struct platform_device *pdev) { -- 2.30.2

1 0

[PATCH AUTOSEL 4.9 02/17] ASoC: sti-sas: add missing MODULE_DEVICE_TABLE
by Sasha Levin 03 Jun '21

03 Jun '21

From: Zou Wei <zou_wei(a)huawei.com> [ Upstream commit e072b2671606c77538d6a4dd5dda80b508cb4816 ] This patch adds missing MODULE_DEVICE_TABLE definition which generates correct modalias for automatic loading of this driver when it is built as an external module. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zou Wei <zou_wei(a)huawei.com> Link: https://lore.kernel.org/r/1620789145-14936-1-git-send-email-zou_wei@huawei.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/sti-sas.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index d6e00c77edcd..7cf76661c3cc 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -542,6 +542,7 @@ static const struct of_device_id sti_sas_dev_match[] = { }, {}, }; +MODULE_DEVICE_TABLE(of, sti_sas_dev_match); static int sti_sas_driver_probe(struct platform_device *pdev) { -- 2.30.2

1 0

[PATCH AUTOSEL 4.14 03/18] ASoC: sti-sas: add missing MODULE_DEVICE_TABLE
by Sasha Levin 03 Jun '21

03 Jun '21

From: Zou Wei <zou_wei(a)huawei.com> [ Upstream commit e072b2671606c77538d6a4dd5dda80b508cb4816 ] This patch adds missing MODULE_DEVICE_TABLE definition which generates correct modalias for automatic loading of this driver when it is built as an external module. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zou Wei <zou_wei(a)huawei.com> Link: https://lore.kernel.org/r/1620789145-14936-1-git-send-email-zou_wei@huawei.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/sti-sas.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index 62c618765224..730dd453a744 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -407,6 +407,7 @@ static const struct of_device_id sti_sas_dev_match[] = { }, {}, }; +MODULE_DEVICE_TABLE(of, sti_sas_dev_match); static int sti_sas_driver_probe(struct platform_device *pdev) { -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 05/23] ASoC: sti-sas: add missing MODULE_DEVICE_TABLE
by Sasha Levin 03 Jun '21

03 Jun '21

From: Zou Wei <zou_wei(a)huawei.com> [ Upstream commit e072b2671606c77538d6a4dd5dda80b508cb4816 ] This patch adds missing MODULE_DEVICE_TABLE definition which generates correct modalias for automatic loading of this driver when it is built as an external module. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zou Wei <zou_wei(a)huawei.com> Link: https://lore.kernel.org/r/1620789145-14936-1-git-send-email-zou_wei@huawei.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/sti-sas.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index 7316c80b8179..27196126f710 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -411,6 +411,7 @@ static const struct of_device_id sti_sas_dev_match[] = { }, {}, }; +MODULE_DEVICE_TABLE(of, sti_sas_dev_match); static int sti_sas_driver_probe(struct platform_device *pdev) { -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 04/23] ASoC: Intel: bytcr_rt5640: Add quirk for the Lenovo Miix 3-830 tablet
by Sasha Levin 03 Jun '21

03 Jun '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit f0353e1f53f92f7b3da91e6669f5d58ee222ebe8 ] The Lenovo Miix 3-830 tablet has only 1 speaker, has an internal analog mic on IN1 and uses JD2 for jack-detect, add a quirk to automatically apply these settings on Lenovo Miix 3-830 tablets. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210508150146.28403-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 077529113ee7..186c0ee059da 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -643,6 +643,20 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_MONO_SPEAKER | BYT_RT5640_MCLK_EN), }, + { /* Lenovo Miix 3-830 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_PRODUCT_VERSION, "Lenovo MIIX 3-830"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Linx Linx7 tablet */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LINX"), -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 03/23] ASoC: Intel: bytcr_rt5640: Add quirk for the Glavey TM800A550L tablet
by Sasha Levin 03 Jun '21

03 Jun '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 28c268d3acdd4cbcd2ac320b85609e77f84e74a7 ] Add a quirk for the Glavey TM800A550L tablet, this BYTCR tablet has no CHAN package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which is the default for BYTCR devices. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210508150146.28403-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 4ebc023f1507..077529113ee7 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -565,6 +565,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Glavey TM800A550L */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Aptio CRB"), + /* Above strings are too generic, also match on BIOS version */ + DMI_MATCH(DMI_BIOS_VERSION, "ZY-8-BI-PX4S70VTR400-X423B-005-D"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 01/31] ASoC: max98088: fix ni clock divider calculation
by Sasha Levin 03 Jun '21

03 Jun '21

From: Marco Felsch <m.felsch(a)pengutronix.de> [ Upstream commit 6c9762a78c325107dc37d20ee21002b841679209 ] The ni1/ni2 ratio formula [1] uses the pclk which is the prescaled mclk. The max98088 datasheet [2] has no such formula but table-12 equals so we can assume that it is the same for both devices. While on it make use of DIV_ROUND_CLOSEST_ULL(). [1] https://datasheets.maximintegrated.com/en/ds/MAX98089.pdf; page 86 [2] https://datasheets.maximintegrated.com/en/ds/MAX98088.pdf; page 82 Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> Link: https://lore.kernel.org/r/20210423135402.32105-1-m.felsch@pengutronix.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98088.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index f031d2caa8b7..fa4cdbfd0b80 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -41,6 +41,7 @@ struct max98088_priv { enum max98088_type devtype; struct max98088_pdata *pdata; struct clk *mclk; + unsigned char mclk_prescaler; unsigned int sysclk; struct max98088_cdata dai[2]; int eq_textcnt; @@ -998,13 +999,16 @@ static int max98088_dai1_hw_params(struct snd_pcm_substream *substream, /* Configure NI when operating as master */ if (snd_soc_component_read32(component, M98088_REG_14_DAI1_FORMAT) & M98088_DAI_MAS) { + unsigned long pclk; + if (max98088->sysclk == 0) { dev_err(component->dev, "Invalid system clock frequency\n"); return -EINVAL; } ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) * (unsigned long long int)rate; - do_div(ni, (unsigned long long int)max98088->sysclk); + pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler); + ni = DIV_ROUND_CLOSEST_ULL(ni, pclk); snd_soc_component_write(component, M98088_REG_12_DAI1_CLKCFG_HI, (ni >> 8) & 0x7F); snd_soc_component_write(component, M98088_REG_13_DAI1_CLKCFG_LO, @@ -1065,13 +1069,16 @@ static int max98088_dai2_hw_params(struct snd_pcm_substream *substream, /* Configure NI when operating as master */ if (snd_soc_component_read32(component, M98088_REG_1C_DAI2_FORMAT) & M98088_DAI_MAS) { + unsigned long pclk; + if (max98088->sysclk == 0) { dev_err(component->dev, "Invalid system clock frequency\n"); return -EINVAL; } ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) * (unsigned long long int)rate; - do_div(ni, (unsigned long long int)max98088->sysclk); + pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler); + ni = DIV_ROUND_CLOSEST_ULL(ni, pclk); snd_soc_component_write(component, M98088_REG_1A_DAI2_CLKCFG_HI, (ni >> 8) & 0x7F); snd_soc_component_write(component, M98088_REG_1B_DAI2_CLKCFG_LO, @@ -1113,8 +1120,10 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai, */ if ((freq >= 10000000) && (freq < 20000000)) { snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x10); + max98088->mclk_prescaler = 1; } else if ((freq >= 20000000) && (freq < 30000000)) { snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x20); + max98088->mclk_prescaler = 2; } else { dev_err(component->dev, "Invalid master clock frequency\n"); return -EINVAL; -- 2.30.2

1 3

[PATCH AUTOSEL 5.10 01/39] ASoC: max98088: fix ni clock divider calculation
by Sasha Levin 03 Jun '21

03 Jun '21

From: Marco Felsch <m.felsch(a)pengutronix.de> [ Upstream commit 6c9762a78c325107dc37d20ee21002b841679209 ] The ni1/ni2 ratio formula [1] uses the pclk which is the prescaled mclk. The max98088 datasheet [2] has no such formula but table-12 equals so we can assume that it is the same for both devices. While on it make use of DIV_ROUND_CLOSEST_ULL(). [1] https://datasheets.maximintegrated.com/en/ds/MAX98089.pdf; page 86 [2] https://datasheets.maximintegrated.com/en/ds/MAX98088.pdf; page 82 Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> Link: https://lore.kernel.org/r/20210423135402.32105-1-m.felsch@pengutronix.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98088.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 4be24e7f51c8..f8e49e45ce33 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -41,6 +41,7 @@ struct max98088_priv { enum max98088_type devtype; struct max98088_pdata *pdata; struct clk *mclk; + unsigned char mclk_prescaler; unsigned int sysclk; struct max98088_cdata dai[2]; int eq_textcnt; @@ -998,13 +999,16 @@ static int max98088_dai1_hw_params(struct snd_pcm_substream *substream, /* Configure NI when operating as master */ if (snd_soc_component_read(component, M98088_REG_14_DAI1_FORMAT) & M98088_DAI_MAS) { + unsigned long pclk; + if (max98088->sysclk == 0) { dev_err(component->dev, "Invalid system clock frequency\n"); return -EINVAL; } ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) * (unsigned long long int)rate; - do_div(ni, (unsigned long long int)max98088->sysclk); + pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler); + ni = DIV_ROUND_CLOSEST_ULL(ni, pclk); snd_soc_component_write(component, M98088_REG_12_DAI1_CLKCFG_HI, (ni >> 8) & 0x7F); snd_soc_component_write(component, M98088_REG_13_DAI1_CLKCFG_LO, @@ -1065,13 +1069,16 @@ static int max98088_dai2_hw_params(struct snd_pcm_substream *substream, /* Configure NI when operating as master */ if (snd_soc_component_read(component, M98088_REG_1C_DAI2_FORMAT) & M98088_DAI_MAS) { + unsigned long pclk; + if (max98088->sysclk == 0) { dev_err(component->dev, "Invalid system clock frequency\n"); return -EINVAL; } ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) * (unsigned long long int)rate; - do_div(ni, (unsigned long long int)max98088->sysclk); + pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler); + ni = DIV_ROUND_CLOSEST_ULL(ni, pclk); snd_soc_component_write(component, M98088_REG_1A_DAI2_CLKCFG_HI, (ni >> 8) & 0x7F); snd_soc_component_write(component, M98088_REG_1B_DAI2_CLKCFG_LO, @@ -1113,8 +1120,10 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai, */ if ((freq >= 10000000) && (freq < 20000000)) { snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x10); + max98088->mclk_prescaler = 1; } else if ((freq >= 20000000) && (freq < 30000000)) { snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x20); + max98088->mclk_prescaler = 2; } else { dev_err(component->dev, "Invalid master clock frequency\n"); return -EINVAL; -- 2.30.2

1 4

[PATCH AUTOSEL 5.12 01/43] ASoC: max98088: fix ni clock divider calculation
by Sasha Levin 03 Jun '21

03 Jun '21

From: Marco Felsch <m.felsch(a)pengutronix.de> [ Upstream commit 6c9762a78c325107dc37d20ee21002b841679209 ] The ni1/ni2 ratio formula [1] uses the pclk which is the prescaled mclk. The max98088 datasheet [2] has no such formula but table-12 equals so we can assume that it is the same for both devices. While on it make use of DIV_ROUND_CLOSEST_ULL(). [1] https://datasheets.maximintegrated.com/en/ds/MAX98089.pdf; page 86 [2] https://datasheets.maximintegrated.com/en/ds/MAX98088.pdf; page 82 Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> Link: https://lore.kernel.org/r/20210423135402.32105-1-m.felsch@pengutronix.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98088.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 4be24e7f51c8..f8e49e45ce33 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -41,6 +41,7 @@ struct max98088_priv { enum max98088_type devtype; struct max98088_pdata *pdata; struct clk *mclk; + unsigned char mclk_prescaler; unsigned int sysclk; struct max98088_cdata dai[2]; int eq_textcnt; @@ -998,13 +999,16 @@ static int max98088_dai1_hw_params(struct snd_pcm_substream *substream, /* Configure NI when operating as master */ if (snd_soc_component_read(component, M98088_REG_14_DAI1_FORMAT) & M98088_DAI_MAS) { + unsigned long pclk; + if (max98088->sysclk == 0) { dev_err(component->dev, "Invalid system clock frequency\n"); return -EINVAL; } ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) * (unsigned long long int)rate; - do_div(ni, (unsigned long long int)max98088->sysclk); + pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler); + ni = DIV_ROUND_CLOSEST_ULL(ni, pclk); snd_soc_component_write(component, M98088_REG_12_DAI1_CLKCFG_HI, (ni >> 8) & 0x7F); snd_soc_component_write(component, M98088_REG_13_DAI1_CLKCFG_LO, @@ -1065,13 +1069,16 @@ static int max98088_dai2_hw_params(struct snd_pcm_substream *substream, /* Configure NI when operating as master */ if (snd_soc_component_read(component, M98088_REG_1C_DAI2_FORMAT) & M98088_DAI_MAS) { + unsigned long pclk; + if (max98088->sysclk == 0) { dev_err(component->dev, "Invalid system clock frequency\n"); return -EINVAL; } ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) * (unsigned long long int)rate; - do_div(ni, (unsigned long long int)max98088->sysclk); + pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler); + ni = DIV_ROUND_CLOSEST_ULL(ni, pclk); snd_soc_component_write(component, M98088_REG_1A_DAI2_CLKCFG_HI, (ni >> 8) & 0x7F); snd_soc_component_write(component, M98088_REG_1B_DAI2_CLKCFG_LO, @@ -1113,8 +1120,10 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai, */ if ((freq >= 10000000) && (freq < 20000000)) { snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x10); + max98088->mclk_prescaler = 1; } else if ((freq >= 20000000) && (freq < 30000000)) { snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x20); + max98088->mclk_prescaler = 2; } else { dev_err(component->dev, "Invalid master clock frequency\n"); return -EINVAL; -- 2.30.2

1 6

Allen&Heath Xone43C: USB communication
by Roman 03 Jun '21

03 Jun '21

Hello all, hello list, I am not sure whether this is the correct list, since it's not strictly an ALSA affair (fw to kernel-devel). Recent progress on driver support for Pioneer devices encouraged me to dissect USB communication with an Allen&Heath Xone43C, a mixer with USB interface. It does feature 16 channels (8 in, 8 out), but only 4 (2 in, 2 out) show up, 2 of which remain automagically silent. There is no linux driver yet, and maybe I can help? Addresses are: 0x80 Endpoint 0 (in) 0x00 Endpoint 0 (out) 0x81 Endpoint 1 (in) Interface 2.1 and Interface 2.2 (vendor spec) 0x02 Endpoint 2 (out) Interface 1.1 and Interface 1.2 (vendor spec) 0x85 Endpoint 5 (in) Interface 1.1 0x83 Endpoint 3 (in) Interface 3 MIDI 0x04 Endpoint 4 (out) Interface 3 MIDI ---------------------------------- I have a sniff here: https://drive.google.com/file/d/1UPXCN-r7jc9hfm1d-Hip8vckSs2RQNtC/view? usp=sharing --------------------------------- USB info: Bus 001 Device 006: ID 22f0:000f Allen&Heath Xone:43C Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x22f0 idProduct 0x000f bcdDevice 2.00 iManufacturer 1 Allen&Heath iProduct 2 Xone:43C iSerial 3 no serial number bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0126 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x0035 bInCollection 3 baInterfaceNr(0) 1 baInterfaceNr(1) 2 baInterfaceNr(2) 3 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0602 Digital Audio Interface bAssocTerminal 0 bSourceID 1 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0602 Digital Audio Interface bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 3 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0126 1x 294 bytes bInterval 4 bRefresh 0 bSynchAddress 133 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 4 bRefresh 5 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 bRefresh 0 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 4 bDelay 1 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0126 1x 294 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0000 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 bRefresh 0 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 0 MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x0041 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 2 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 3 bNrInputPins 1 baSourceID( 0) 2 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 4 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 3 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 4 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 1 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 0 bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0001 Self Powered --- Best wishes, Roman

2 1

[PATCH] ASoC: SOF: Intel: Remove duplicate include of shim.h
by Wan Jiabing 03 Jun '21

03 Jun '21

shim.h is included twice. As shim.h is not related to later header file, keep one which has comment and remove the former one. Signed-off-by: Wan Jiabing <wanjiabing(a)vivo.com> --- sound/soc/sof/intel/pci-tng.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/intel/pci-tng.c b/sound/soc/sof/intel/pci-tng.c index 4ee1da3..4bded66 100644 --- a/sound/soc/sof/intel/pci-tng.c +++ b/sound/soc/sof/intel/pci-tng.c @@ -15,7 +15,6 @@ #include <sound/sof.h> #include "../ops.h" #include "atom.h" -#include "shim.h" #include "../sof-pci-dev.h" #include "../sof-audio.h" -- 2.7.4

2 1

[PATCH v2 0/4] ASoC: codecs: wcd934x: add Headset and button detection support
by Srinivas Kandagatla 03 Jun '21

03 Jun '21

This patchset adds support to MBHC(Multi Button Headset Control) block found in Qualcomm WCD codecs. MBHC support headset type detection, both Mechanical and electrical insert/removal detection along with 8 buttons detection, Over current interrupts on HPHL/R, Impedance Measurements on HPHL/R. Eventhough MBHC block supports things like OverCurrent detection, Currently its reported as a kernel debug message. Should this be reported as an uevent to userspace? like the way USB reports? Any suggestions? First patch adds a common mbhc driver and the second one wcd934x specific driver changes along with sdm845 soundcard related changes. Common wcd-mbhc-v2 driver should be reusable across multiple codecs like WCD9335, WCD934x, WCD937x and WCD938x. Most of the work is derived from downstream Qualcomm kernels. Credits to various Qualcomm authors from Patrick Lai's team who have contributed to this code. This patchset thanks, srini Changes since v1: - updated buttons-vthreshold bindings from review comments for wcd938x Srinivas Kandagatla (4): ASoC: dt-bindings: wcd934x: add bindings for Headset Button detection ASoC: codecs: wcd: add multi button Headset detection support ASoC: codecs: wcd934x: add mbhc support ASoC: qcom: sdm845: add jack support for WCD934x .../bindings/sound/qcom,wcd934x.yaml | 30 + include/linux/mfd/wcd934x/registers.h | 57 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-mbhc-v2.c | 1468 +++++++++++++++++ sound/soc/codecs/wcd-mbhc-v2.h | 340 ++++ sound/soc/codecs/wcd934x.c | 884 +++++++++- sound/soc/qcom/sdm845.c | 8 + 8 files changed, 2778 insertions(+), 15 deletions(-) create mode 100644 sound/soc/codecs/wcd-mbhc-v2.c create mode 100644 sound/soc/codecs/wcd-mbhc-v2.h -- 2.21.0

3 7

[PATCH][next] ASoC: rsnd: check for zero node count
by Colin King 03 Jun '21

03 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> The call to rsnd_node_count can potentially return a zero node count so add a check for this corner case. (Note that the two other calls to rsnd_node_count in the kernel perform this check, so I think it justifies adding this). This avoids using a zero nr in a devm_kcalloc call. Addresses-Coverity: ("Unchecked return value") Fixes: c413983eb66a ("ASoC: rsnd: adjust disabled module") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sh/rcar/ssiu.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 5682c74bb7ff..0d8f97633dd2 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -515,6 +515,9 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv) else nr = priv->ssi_nr; + if (!nr) + return -EINVAL; + ssiu = devm_kcalloc(dev, nr, sizeof(*ssiu), GFP_KERNEL); if (!ssiu) return -ENOMEM; -- 2.31.1

4 3

[PATCH v12 0/4] ASoC: codecs: add rk817 support
by Chris Morgan 03 Jun '21

03 Jun '21

From: Chris Morgan <macromorgan(a)hotmail.com> This series is to add support for the Rockchip rk817 audio codec which is present in all Rockchip rk817 MFD chips. Changes in v12: - Updated codec cell in MFD driver to a single line. Changes in v11: - Stopped checking for presence of parent node in codec driver, as driver should never be called if parent device doesn't exist to call it. - Made codec subnode optional and stopped returning errors if it is missing. - Downgraded missing codec clock message from dev_err to dev_dbg. Users who choose to use the rk817 PMIC but do not choose to use the codec will not get an error message unless they have debugging enabled, whereas users who wish to use the codec but don't have the clock defined will not get an error message unless they have debug messages enabled. This seems like a fair compromise between mututally exclusive users. Note that errors with the clock or codec will still result in a dev_err. - Updated DT documentation to note that codec subnode is optional if no properties are to exist beneath it. Changes in v10: - Correct order of test/ack/signed-off to chronological order. - Removed ifdef from mfd driver keep cell from being added when the codec was not to be built. - Changed codec DT parsing messages from error to debug to prevent errors in dmesg log in the rare but valid case a user has the rk817 PMIC but does not use the codec. Changes in v9: - Add cover letter. - Remove documentation for interrupt parent per Rob Herring's request. - Remove unused MODULE_DEVICE_TABLE to fix a bug identified by kernel test robot. Changes in v8: - Added additional documentation for missing properties of #sound-dai-cells, interrupt-parent, and wakeup-source for mfd documentation. - Corrected order of elements descriptions in device tree documentation. - Changed name of "mic-in-differential" to "rockchip,mic-in-differential". - Changed name of sound card from "rockchip,rk817-codec" to "Analog". - Removed unused resets and reset-names from the i2s1_2ch node. Changes in v7: - Removed ifdef around register definitions for MFD. - Replaced codec documentation with updates to MFD documentation. - Reordered elements in example to comply with upstream rules. - Added binding update back for Odroid Go Advance as requested. - Submitting patches from gmail now. Changes in v6: - Included additional project maintainers for correct subsystems. - Removed unneeded compatible from DT documentation. - Removed binding update for Odroid Go Advance (will do in separate series). Changes in v5: - Move register definitions from rk817_codec.h to main rk808.h register definitions. - Add volatile register for codec bits. - Add default values for codec bits. - Removed of_compatible from mtd driver (not necessary). - Switched to using parent regmap instead of private regmap for codec. Changes in v4: - Created set_pll() call. - Created user visible gain control in mic. - Check for return value of clk_prepare_enable(). - Removed duplicate clk_prepare_enable(). - Split DT documentation to separate commit. Changes in v3: - Use DAPM macros to set audio path. - Updated devicetree binding (as every rk817 has this codec chip). - Changed documentation to yaml format. - Split MFD changes to separate commit. Changes in v2: - Fixed audio path registers to solve some bugs. Chris Morgan (4): mfd: Add Rockchip rk817 audio CODEC support ASoC: Add Rockchip rk817 audio CODEC support dt-bindings: Add Rockchip rk817 audio CODEC support arm64: dts: rockchip: add rk817 codec to Odroid Go .../devicetree/bindings/mfd/rk808.txt | 188 ++++++ .../boot/dts/rockchip/rk3326-odroid-go2.dts | 36 +- drivers/mfd/rk808.c | 81 +++ include/linux/mfd/rk808.h | 81 +++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk817_codec.c | 539 ++++++++++++++++++ 7 files changed, 931 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/rk817_codec.c -- 2.25.1

4 13

Speakers + Microphone Volume Reset
by GitHub issues - edited 03 Jun '21

03 Jun '21

alsa-project/alsa-lib issue #144 was edited from rosenbergd: Every time I start up, connect or disconnect headphones, my volume resets to 74%, I've also noticed the microphone is doing the same thing. I'm not sure if this is a pulseaudio bug, but here are the logs from `pulseaudio -vv` which seems to show alsa increasing the volume to 74% after initially setting it to 25% which is the correct volume that it should be at: ``` D: [pulseaudio] module-alsa-card.c: Jack 'Headphone Jack' is now plugged in D: [pulseaudio] device-port.c: Setting port analog-output-headphones to status yes D: [pulseaudio] module-switch-on-port-available.c: Trying to switch to port analog-output-headphones I: [pulseaudio] alsa-sink.c: Successfully enabled deferred volume. I: [pulseaudio] alsa-sink.c: Hardware volume ranges from -179.00 dB to 0.00 dB. I: [pulseaudio] alsa-sink.c: Fixing base volume to 0.00 dB I: [pulseaudio] alsa-sink.c: Using hardware volume control. Hardware dB scale supported. I: [pulseaudio] alsa-sink.c: Using hardware mute control. D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Activating path analog-output-headphones D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Path analog-output-headphones (Headphones), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-179, max_dB=0 D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Master', direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Headphone', direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Speaker', direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Bass Speaker', direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'PCM', direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'IEC958', direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Dock Headphone, alsa_name='Dock Headphone Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Dock Headphone Phantom, alsa_name='Dock Headphone Phantom Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone, alsa_name='Front Headphone Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone, alsa_name='Front Headphone Jack', index='1', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone Front, alsa_name='Front Headphone Front Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone Surround, alsa_name='Front Headphone Surround Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone Phantom, alsa_name='Front Headphone Phantom Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone, alsa_name='Headphone Jack', index='0', detection possible D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone Phantom, alsa_name='Headphone Phantom Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone Mic, alsa_name='Headphone Mic Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone - Output, alsa_name='Headphone - Output Jack', index='0', detection unavailable I: [pulseaudio] sink.c: Changed port of sink 0 "alsa_output.pci-0000_00_1f.3.analog-stereo" to analog-output-headphones I: [pulseaudio] module-device-restore.c: Restoring volume for sink alsa_output.pci-0000_00_1f.3.analog-stereo. D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 1369 / 2% / -100.80 dB, front-right: 1369 / 2% / -100.80 dB to front-left: 16380 / 25% / -36.13 dB, front-right: 16380 / 25% / -36.13 dB. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Requested volume: front-left: 16380 / 25% / -36.13 dB, front-right: 16380 / 25% / -36.13 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Got hardware volume: front-left: 16462 / 25% / -36.00 dB, front-right: 16462 / 25% / -36.00 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Calculated software volume: front-left: 65210 / 100% / -0.13 dB, front-right: 65210 / 100% / -0.13 dB (accurate-enough=yes) D: [alsa-sink-ALC892 Analog] sink.c: Volume going up to 16380 at 263091695 D: [alsa-sink-ALC892 Analog] sink.c: Volume change to 16380 at 263091695 was written 5 usec late I: [pulseaudio] module-device-restore.c: Restoring mute state for sink alsa_output.pci-0000_00_1f.3.analog-stereo. D: [pulseaudio] sink.c: The mute of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from yes to no. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] device-port.c: Setting port analog-output-speaker to status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-21 to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-21+input:analog-stereo to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-40 to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-40+input:analog-stereo to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 5578 / 9% / -64.20 dB, front-right: 5578 / 9% / -64.20 dB D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 16380 / 25% / -36.13 dB, front-right: 16380 / 25% / -36.13 dB to front-left: 5578 / 9% / -64.20 dB, front-right: 5578 / 9% / -64.20 dB. D: [pulseaudio] sink.c: The mute of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from no to yes. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 49714 / 76% / -7.20 dB, front-right: 49714 / 76% / -7.20 dB D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 5578 / 9% / -64.20 dB, front-right: 5578 / 9% / -64.20 dB to front-left: 49714 / 76% / -7.20 dB, front-right: 49714 / 76% / -7.20 dB. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] sink.c: The mute of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from yes to no. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 49714 / 76% / -7.20 dB, front-right: 49714 / 76% / -7.20 dB to front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB I: [pulseaudio] module-device-restore.c: Storing volume/mute for device+port sink:alsa_output.pci-0000_00_1f.3.analog-stereo:analog-output-headphones. I: [pulseaudio] module-device-restore.c: Synced. ``` I'm using Linux Mint 20.1 Cinnamon v4.8.6 Kernel `v5.4.0-74` (but this was happening on `-73` as well) Issue URL : https://github.com/alsa-project/alsa-lib/issues/144 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Speakers + Microphone Volume Reset
by GitHub issues - opened 03 Jun '21

03 Jun '21

alsa-project/alsa-lib issue #144 was opened from rosenbergd: Every time I start up, connect or disconnect headphones, my volume resets to 74%, I've also noticed the microphone is doing the same thing. I'm not sure if this is a pulseaudio bug, but here are the logs from `pulseaudio -vv` which seems to show alsa increasing the volume to 74% after initially setting it to 25% which is the correct volume that it should be at: ``` D: [pulseaudio] module-alsa-card.c: Jack 'Headphone Jack' is now plugged in D: [pulseaudio] device-port.c: Setting port analog-output-headphones to status yes D: [pulseaudio] module-switch-on-port-available.c: Trying to switch to port analog-output-headphones I: [pulseaudio] alsa-sink.c: Successfully enabled deferred volume. I: [pulseaudio] alsa-sink.c: Hardware volume ranges from -179.00 dB to 0.00 dB. I: [pulseaudio] alsa-sink.c: Fixing base volume to 0.00 dB I: [pulseaudio] alsa-sink.c: Using hardware volume control. Hardware dB scale supported. I: [pulseaudio] alsa-sink.c: Using hardware mute control. D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Activating path analog-output-headphones D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Path analog-output-headphones (Headphones), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-179, max_dB=0 D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Master', direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Headphone', direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Speaker', direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Bass Speaker', direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'PCM', direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'IEC958', direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Dock Headphone, alsa_name='Dock Headphone Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Dock Headphone Phantom, alsa_name='Dock Headphone Phantom Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone, alsa_name='Front Headphone Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone, alsa_name='Front Headphone Jack', index='1', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone Front, alsa_name='Front Headphone Front Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone Surround, alsa_name='Front Headphone Surround Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone Phantom, alsa_name='Front Headphone Phantom Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone, alsa_name='Headphone Jack', index='0', detection possible D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone Phantom, alsa_name='Headphone Phantom Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone Mic, alsa_name='Headphone Mic Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone - Output, alsa_name='Headphone - Output Jack', index='0', detection unavailable I: [pulseaudio] sink.c: Changed port of sink 0 "alsa_output.pci-0000_00_1f.3.analog-stereo" to analog-output-headphones I: [pulseaudio] module-device-restore.c: Restoring volume for sink alsa_output.pci-0000_00_1f.3.analog-stereo. D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 1369 / 2% / -100.80 dB, front-right: 1369 / 2% / -100.80 dB to front-left: 16380 / 25% / -36.13 dB, front-right: 16380 / 25% / -36.13 dB. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Requested volume: front-left: 16380 / 25% / -36.13 dB, front-right: 16380 / 25% / -36.13 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Got hardware volume: front-left: 16462 / 25% / -36.00 dB, front-right: 16462 / 25% / -36.00 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Calculated software volume: front-left: 65210 / 100% / -0.13 dB, front-right: 65210 / 100% / -0.13 dB (accurate-enough=yes) D: [alsa-sink-ALC892 Analog] sink.c: Volume going up to 16380 at 263091695 D: [alsa-sink-ALC892 Analog] sink.c: Volume change to 16380 at 263091695 was written 5 usec late I: [pulseaudio] module-device-restore.c: Restoring mute state for sink alsa_output.pci-0000_00_1f.3.analog-stereo. D: [pulseaudio] sink.c: The mute of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from yes to no. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] device-port.c: Setting port analog-output-speaker to status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-21 to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-21+input:analog-stereo to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-40 to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-40+input:analog-stereo to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 5578 / 9% / -64.20 dB, front-right: 5578 / 9% / -64.20 dB D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 16380 / 25% / -36.13 dB, front-right: 16380 / 25% / -36.13 dB to front-left: 5578 / 9% / -64.20 dB, front-right: 5578 / 9% / -64.20 dB. D: [pulseaudio] sink.c: The mute of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from no to yes. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 49714 / 76% / -7.20 dB, front-right: 49714 / 76% / -7.20 dB D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 5578 / 9% / -64.20 dB, front-right: 5578 / 9% / -64.20 dB to front-left: 49714 / 76% / -7.20 dB, front-right: 49714 / 76% / -7.20 dB. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] sink.c: The mute of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from yes to no. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 49714 / 76% / -7.20 dB, front-right: 49714 / 76% / -7.20 dB to front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB I: [pulseaudio] module-device-restore.c: Storing volume/mute for device+port sink:alsa_output.pci-0000_00_1f.3.analog-stereo:analog-output-headphones. I: [pulseaudio] module-device-restore.c: Synced. ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/144 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: fsl-asoc-card: change dev_err to dev_err_probe for defer probe
by Shengjiu Wang 03 Jun '21

03 Jun '21

Don't need to print error message for defer probe Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - use dev_err_probe instead of dev_dbg 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 4f55b316cf0f..2fa371ec6b1c 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -709,8 +709,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) of_node_put(framemaster); if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { - dev_err(&pdev->dev, "failed to find codec device\n"); ret = -EPROBE_DEFER; + dev_err_probe(&pdev->dev, ret, "failed to find codec device\n"); goto asrc_fail; } -- 2.27.0

1 0

trivial patch
by josef radinger 02 Jun '21

02 Jun '21

just fix some trivial typo in comments, sorry for the third try. now the direction between diffs is correct. diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 1855b805eba2..3c68df82c95c 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -373,7 +373,7 @@ static int q6adm_device_open(struct q6adm *adm, struct q6copp *copp, * @rate: rate at which copp is required. * @channel_mode: channel mode * @topology: adm topology id - * @perf_mode: performace mode. + * @perf_mode: performance mode. * @bit_width: audio sample bit width * @app_type: Application type. * @acdb_id: ACDB id @@ -452,7 +452,7 @@ EXPORT_SYMBOL_GPL(q6adm_get_copp_id); * @dev: Pointer to adm child device. * @path: playback or capture path. * @payload_map: map between session id and afe ports. - * @perf_mode: Performace mode. + * @perf_mode: Performance mode. * * Return: Will be an negative on error or a zero on success. */ yours josef

1 0

[PATCH] ASoC: AMD Renoir - add DMI entry for Lenovo 2020 AMD platforms
by Mark Pearson 02 Jun '21

02 Jun '21

More laptops identified where the AMD ACP bridge needs to be blocked or the microphone will not work when connected to HDMI. Use DMI to block the microphone PCM device for these platforms. Suggested-by: Gabriel Craciunescu <nix.or.die(a)gmail.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 35 +++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 19438da5dfa5..c9fb1c8fbf8c 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -199,6 +199,41 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), } }, + { + /* Lenovo ThinkPad P14s Gen 1 (20Y1) */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20Y1"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UH"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 Campus */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UJ"), + } + }, + { + /* Lenovo ThinkPad T14 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UD"), + } + }, + { + /* Lenovo ThinkPad X13 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UF"), + } + }, {} }; -- 2.31.1

1 1

[PATCH] ALSA: timer: Fix master timer notification
by Takashi Iwai 02 Jun '21

02 Jun '21

snd_timer_notify1() calls the notification to each slave for a master event, but it passes a wrong event number. It should be +10 offset, corresponding to SNDRV_TIMER_EVENT_MXXX, but it's incorrectly with +100 offset. Casually this was spotted by UBSAN check via syzkaller. Reported-by: syzbot+d102fa5b35335a7e544e(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000e5560e05c3bd1d63@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/timer.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 6898b1ac0d7f..92b7008fcdb8 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -520,9 +520,10 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event) return; if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE) return; + event += 10; /* convert to SNDRV_TIMER_EVENT_MXXX */ list_for_each_entry(ts, &ti->slave_active_head, active_list) if (ts->ccallback) - ts->ccallback(ts, event + 100, &tstamp, resolution); + ts->ccallback(ts, event, &tstamp, resolution); } /* start/continue a master timer */ -- 2.26.2

2 1

[PATCH] ASoC: AMD Renoir - add DMI entry for Lenovo 2020 AMD platforms
by Mark Pearson 02 Jun '21

02 Jun '21

More laptops identified where the AMD ACP bridge needs to be blocked or the microphone will not work when connected to HDMI. Use DMI to block the microphone PCM device for these platforms. Suggested-by: Gabriel Craciunescu <nix.or.die(a)gmail.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 35 +++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 19438da5dfa5..c9fb1c8fbf8c 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -199,6 +199,41 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), } }, + { + /* Lenovo ThinkPad P14s Gen 1 (20Y1) */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20Y1"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UH"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 Campus */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UJ"), + } + }, + { + /* Lenovo ThinkPad T14 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UD"), + } + }, + { + /* Lenovo ThinkPad X13 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UF"), + } + }, {} }; -- 2.31.1

2 3

[PATCH v1 1/4] ASoC: dt-bindings: nxp,tfa989x: Add tfa9897 support
by Vincent Knecht 02 Jun '21

02 Jun '21

Document TFA9897 bindings. Signed-off-by: Vincent Knecht <vincent.knecht(a)mailoo.org> --- Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml index 45db5776550c..46ddc1f3fc0c 100644 --- a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml +++ b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml @@ -13,6 +13,7 @@ properties: compatible: enum: - nxp,tfa9895 + - nxp,tfa9897 reg: maxItems: 1 -- 2.31.1

3 7

[PATCH -next] ASoC: Intel: Skylake: use DEVICE_ATTR_RO macro
by YueHaibing 02 Jun '21

02 Jun '21

Use DEVICE_ATTR_RO() helper instead of plain DEVICE_ATTR(), which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/intel/skylake/skl-nhlt.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 87c891c46291..64226072f0ee 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -149,8 +149,8 @@ int skl_nhlt_update_topology_bin(struct skl_dev *skl) return 0; } -static ssize_t skl_nhlt_platform_id_show(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t platform_id_show(struct device *dev, + struct device_attribute *attr, char *buf) { struct pci_dev *pci = to_pci_dev(dev); struct hdac_bus *bus = pci_get_drvdata(pci); @@ -166,7 +166,7 @@ static ssize_t skl_nhlt_platform_id_show(struct device *dev, return sprintf(buf, "%s\n", platform_id); } -static DEVICE_ATTR(platform_id, 0444, skl_nhlt_platform_id_show, NULL); +static DEVICE_ATTR_RO(platform_id); int skl_nhlt_create_sysfs(struct skl_dev *skl) { -- 2.17.1

3 2

[PATCH] ASoC: sigmadsp: Remove the repeated declaration
by Shaokun Zhang 02 Jun '21

02 Jun '21

Function 'sigmadsp_reset' is declared twice, so remove the repeated declaration. Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Shaokun Zhang <zhangshaokun(a)hisilicon.com> --- sound/soc/codecs/sigmadsp.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/sigmadsp.h b/sound/soc/codecs/sigmadsp.h index d63b8c366efb..2783eff633a1 100644 --- a/sound/soc/codecs/sigmadsp.h +++ b/sound/soc/codecs/sigmadsp.h @@ -44,7 +44,6 @@ struct sigmadsp { struct sigmadsp *devm_sigmadsp_init(struct device *dev, const struct sigmadsp_ops *ops, const char *firmware_name); -void sigmadsp_reset(struct sigmadsp *sigmadsp); int sigmadsp_restrict_params(struct sigmadsp *sigmadsp, struct snd_pcm_substream *substream); -- 2.7.4

2 1

[v4] ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
by Judy Hsiao 02 Jun '21

02 Jun '21

From: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> This patch fixes PoP noise of around 15ms observed during audio capture begin. Enables BCLK and LRCLK in snd_soc_dai_ops prepare call for introducing some delay before capture start and clock enable. Co-developed-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> (am from https://patchwork.kernel.org/patch/12276369/) (also found at https://lore.kernel.org/r/20210524142114.18676-1-srivasam@codeaurora.org) --- Chages Since V3: -- Check BCLK is off before enabling it in lpass_cpu_daiops_prepare as lpass_cpu_daiops_prepare can be called multiple times. -- Check BCLK is on before disabling it in lpass_cpu_daiops_shutdown to fix the WARN. It is because BCLK may not be enabled if lpass_cpu_daiops_prepare is not called before lpass_cpu_daiops_shutdown. -- Adds more comments. Changes Since V2: -- Updated comments as per linux style -- Removed unrelated changes. Changes Since V1: -- Enableed BCLK and LRCLK in dai ops prepare API instead of startup API -- Added comments sound/soc/qcom/lpass-cpu.c | 83 +++++++++++++++++++++++++++++++++++++- 1 file changed, 82 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index af8cb64924a0..b572d7874554 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -6,6 +6,7 @@ */ #include <linux/clk.h> +#include <linux/clk-provider.h> #include <linux/kernel.h> #include <linux/module.h> #include <linux/of.h> @@ -93,8 +94,28 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + /* + * To ensure LRCLK disabled even in device node validation + * Will not impact if disabled in lpass_cpu_daiops_trigger() + * suspend. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); + else + regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE); + + /* + * BCLK may not be enabled if lpass_cpu_daiops_prepare is called before + * lpass_cpu_daiops_shutdown. It's paired with the clk_enable in + * lpass_cpu_daiops_prepare. + */ + if (__clk_is_enabled(drvdata->mi2s_bit_clk[dai->driver->id])) + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); + clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } @@ -275,6 +296,18 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * To ensure lpass BCLK/LRCLK is enabled during + * device resume as lpass_cpu_daiops_prepare() is not called + * after the device resumes. We don't check BCLK status before + * enable/disable BCLK in trigger event because: + * 1. These trigger events are paired, so the BCLK + * enable_count is balanced. + * 2. the BCLK can be shared (ex: headset and headset mic), + * we need to increase the enable_count so that we don't + * turn off the shared BCLK while other devices are using + * it. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); @@ -288,7 +321,8 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, ret = clk_enable(drvdata->mi2s_bit_clk[id]); if (ret) { - dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + dev_err(dai->dev, + "error in enabling mi2s bit clk: %d\n", ret); clk_disable(drvdata->mi2s_osr_clk[id]); return ret; } @@ -296,6 +330,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* + * To ensure lpass BCLK/LRCLK is disabled during + * device suspend. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); @@ -315,12 +353,55 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, return ret; } +static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret; + /* + * To ensure lpass BCLK/LRCLK is enabled bit before playback/capture + * data flow starts. This allows other codec to have some delay before + * the data flow. + * (ex: to drop start up pop noise before capture starts). + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = regmap_fields_write(i2sctl->spken, id, + LPAIF_I2SCTL_SPKEN_ENABLE); + else + ret = regmap_fields_write(i2sctl->micen, id, + LPAIF_I2SCTL_MICEN_ENABLE); + + if (ret) { + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + return ret; + } + + /* + * Check BCLK is off before enabling it as lpass_cpu_daiops_prepare can + * be called multiple times. It's paired with the clk_disable in + * lpass_cpu_daiops_shutdown. + */ + if (!__clk_is_enabled(drvdata->mi2s_bit_clk[dai->driver->id])) { + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + if (ret) { + dev_err(dai->dev, + "error in enabling mi2s bit clk: %d\n", ret); + clk_disable(drvdata->mi2s_osr_clk[id]); + return ret; + } + } + return 0; +} + const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, .trigger = lpass_cpu_daiops_trigger, + .prepare = lpass_cpu_daiops_prepare, }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); -- 2.31.0

2 1

[PATCH] ALSA: hda: update the power_state during the direct-complete
by Hui Wang 02 Jun '21

02 Jun '21

The patch_realtek.c needs to check if the power_state.event equals PM_EVENT_SUSPEND, after using the direct-complete, the suspend() and resume() will be skipped if the codec is already rt_suspended, in this case, the patch_realtek.c will always get PM_EVENT_ON even the system is really resumed from S3. We could set power_state to PMSG_SUSPEND in the prepare(), if other PM functions are called before complete(), those functions will override power_state; if no other PM functions are called before complete(), we could know the suspend() and resume() are skipped since only S3 pm functions could be skipped by direct-complete, in this case set power_state to PMSG_RESUME in the complete(). This could guarantee the first time of calling hda_codec_runtime_resume() after complete() has the correct power_state. Fixes: 215a22ed31a1 ("ALSA: hda: Refactor codec PM to use direct-complete optimization") Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_codec.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a31009afc025..5462f771c2f9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2917,6 +2917,7 @@ static int hda_codec_runtime_resume(struct device *dev) #ifdef CONFIG_PM_SLEEP static int hda_codec_pm_prepare(struct device *dev) { + dev->power.power_state = PMSG_SUSPEND; return pm_runtime_suspended(dev); } @@ -2924,6 +2925,10 @@ static void hda_codec_pm_complete(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); + /* If no other pm-functions are called between prepare() and complete() */ + if (dev->power.power_state.event == PM_EVENT_SUSPEND) + dev->power.power_state = PMSG_RESUME; + if (pm_runtime_suspended(dev) && (codec->jackpoll_interval || hda_codec_need_resume(codec) || codec->forced_resume)) pm_request_resume(dev); -- 2.25.1

2 1

[syzbot] UBSAN: shift-out-of-bounds in snd_timer_user_ccallback
by Dongliang Mu 02 Jun '21

02 Jun '21

> Hello, > > syzbot found the following issue on: > > HEAD commit: 5ff2756a Merge tag 'nfs-for-5.13-2' of git://git.linux-nfs.. > git tree: upstream > console output: https://syzkaller.appspot.com/x/log.txt?x=17872d5bd00000 > kernel config: https://syzkaller.appspot.com/x/.config?x=770708ea7cfd4916 > dashboard link: https://syzkaller.appspot.com/bug?extid=d102fa5b35335a7e544e > > Unfortunately, I don't have any reproducer for this issue yet. > > IMPORTANT: if you fix the issue, please add the following tag to the commit: > Reported-by: syzbot+d102fa...(a)syzkaller.appspotmail.com > > ================================================================================ > UBSAN: shift-out-of-bounds in sound/core/timer.c:1376:23 > shift exponent 105 is too large for 32-bit type 'int' > CPU: 1 PID: 10368 Comm: syz-executor.1 Not tainted 5.13.0-rc3-syzkaller #0 > Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 > Call Trace: > __dump_stack lib/dump_stack.c:79 [inline] > dump_stack+0x141/0x1d7 lib/dump_stack.c:120 > ubsan_epilogue+0xb/0x5a lib/ubsan.c:148 > __ubsan_handle_shift_out_of_bounds.cold+0xb1/0x181 lib/ubsan.c:327 > snd_timer_user_ccallback.cold+0x19/0x1e sound/core/timer.c:1376 > > snd_timer_notify1+0x243/0x3b0 sound/core/timer.c:525 The root cause of this bug is in the snd_timer_notify1 [1]. At the end of this function, it calls "ts->ccallback(ts, event + 100, &tstamp, resolution)". Here the variable event is 5. It adds 100 and is passed as 2nd argument of snd_timer_user_ccallback. >From the variable naming, the 2nd argument should an event, and in the range of event enumeration. In fact, 105 (event + 100) is out of this range. I don't quite understand the meaning of adding 100. Any thought here? [1] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/sou… -- My best regards to you. No System Is Safe! Dongliang Mu

4 4

[PATCH RESEND] ALSA: timer: Fix master timer notification
by Takashi Iwai 02 Jun '21

02 Jun '21

snd_timer_notify1() calls the notification to each slave for a master event, but it passes a wrong event number. It should be +10 offset, corresponding to SNDRV_TIMER_EVENT_MXXX, but it's incorrectly with +100 offset. Casually this was spotted by UBSAN check via syzkaller. Reported-by: syzbot+d102fa5b35335a7e544e(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000e5560e05c3bd1d63@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/timer.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 6898b1ac0d7f..92b7008fcdb8 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -520,9 +520,10 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event) return; if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE) return; + event += 10; /* convert to SNDRV_TIMER_EVENT_MXXX */ list_for_each_entry(ts, &ti->slave_active_head, active_list) if (ts->ccallback) - ts->ccallback(ts, event + 100, &tstamp, resolution); + ts->ccallback(ts, event, &tstamp, resolution); } /* start/continue a master timer */ -- 2.26.2

1 0

[PATCH][next] ASoC: rk817: remove redundant assignment to pointer node
by Colin King 02 Jun '21

02 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> The pointer node is being initialized with a value that is never read and it is being updated later with a new value. The initialization is redundant and can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rk817_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 17e672b85ee5..0d7cc26ded57 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -457,7 +457,7 @@ static const struct snd_soc_component_driver soc_codec_dev_rk817 = { static void rk817_codec_parse_dt_property(struct device *dev, struct rk817_codec_priv *rk817) { - struct device_node *node = dev->parent->of_node; + struct device_node *node; node = of_get_child_by_name(dev->parent->of_node, "codec"); if (!node) { -- 2.31.1

3 2

no sound via HDMI
by reinerschrutz＠web.de 02 Jun '21

02 Jun '21

Hi, missing sound via monitor and HDMI cable following ubuntuusers "HDA" article I found: 0[generic ] - HD-Audio Generic HD-Audio Generic at 0xfcc 88000 irg 62 1[generic_1 ] - HD-Audio Generic HD-Audio Generic at 0xfcc 88000 irq 63 looking for a pci device I found: 05:00.1 Audio device Advanced Micro Devices, Inc. [AMD/ATI] Device 1637 05:00.6 Audio device Advanced Mirco Devices, Inc. [AMD] Family 17h (Models 10h 1fh) HD Audio Controller Codec found: ATI R6xx HDMI My system: Asus Prime A520M-K, AMD Ryzen 7 Pro 4750G, 8x 3,6 GHz, 16 GD DDR4 RAM, 500 GB Samsung 870 EVO Advanced Linux Sound Architecture - Driver Cofiguration Guide said to give notice. Thanks. Stefan Ocken

1 0

[PATCH] ASoC: rk817: fix a warning in rk817_probe()
by Chris Morgan 02 Jun '21

02 Jun '21

From: Chris Morgan <macromorgan(a)hotmail.com> The return value of snd_soc_component_write() is stored but not evaluated and this results in a warning when W=1 is set. Stop storing the return value to be consistent with all other calls of snd_soc_component_write() and to remove the warning. Fixes: 0d6a04da9b25 ("ASoC: Add Rockchip rk817 audio CODEC support") Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- sound/soc/codecs/rk817_codec.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 17e672b85ee5..fd3a5ba034a9 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -415,12 +415,11 @@ static int rk817_probe(struct snd_soc_component *component) { struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); struct rk808 *rk808 = dev_get_drvdata(component->dev->parent); - int ret; snd_soc_component_init_regmap(component, rk808->regmap); rk817->component = component; - ret = snd_soc_component_write(component, RK817_CODEC_DTOP_LPT_SRST, 0x40); + snd_soc_component_write(component, RK817_CODEC_DTOP_LPT_SRST, 0x40); rk817_init(component); -- 2.25.1

2 1

[PATCH 0/3] ALSA: firewire-motu: media clock recovery for sph-aware devices
by Takashi Sakamoto 02 Jun '21

02 Jun '21

Hi, In a commit f9e5ecdfc2c2 ("ALSA: firewire-lib: add replay target to cache sequence of packet"), I categorize devices supported by drivers in ALSA firewire stack in terms of the way to deliver effective sampling transfer frequency. This patchset is for the devices in group 3. The devices are known to have problems when ALSA firewire-motu driver handles. Many of them generate sound with noise. In the worst case, it generates no sound. The devices interpret presentation time to decide playback timing. Unlike the syt-aware devices, the devices interpret the presentation time in source packet header (SPH) per data block, instead of the presentation time in syt field of CIP header. Current implementation of the driver processes the sequence of outgoing packet by computation according to nominal sampling transfer frequency. However, the ideal sequence is not adequate to the devices, actually. With this patchset, the drivers are going to replay the sequence of incoming packets for media clock recovery, instead of nominal sampling transfer frequency. For the detail of sequence replay, please refer to a commit 39c2649c71d8 ("ALSA: firewire-lib: replay sequence of incoming packets for outgoing packets"). The sequence replay is done by two levels; the sequence of the number of data blocks per packet, and the sequence of SPH per data blocks in the packet. Takashi Sakamoto (3): ALSA: firewire-motu: use macro for magic numbers relevant to IEC 61883-1 ALSA: firewire-motu: cache event ticks in source packet header per data block ALSA: firewire-motu: sequence replay for source packet header sound/firewire/motu/amdtp-motu.c | 136 +++++++++++++++--------------- sound/firewire/motu/motu-stream.c | 27 +++++- sound/firewire/motu/motu.h | 14 ++- 3 files changed, 104 insertions(+), 73 deletions(-) -- 2.27.0

2 4

[PATCH v2] ALSA: control led: fix memory leak in snd_ctl_led_register
by Dongliang Mu 02 Jun '21

02 Jun '21

The snd_ctl_led_sysfs_add and snd_ctl_led_sysfs_remove should contain the refcount operations in pair. However, snd_ctl_led_sysfs_remove fails to decrease the refcount to zero, which causes device_release never to be invoked. This leads to memory leak to some resources, like struct device_private. In addition, we also free some other similar memory leaks in snd_ctl_led_init/snd_ctl_led_exit. Fix this by replacing device_del to device_unregister in snd_ctl_led_sysfs_remove/snd_ctl_led_init/snd_ctl_led_exit. Note that, when CONFIG_DEBUG_KOBJECT_RELEASE is enabled, put_device will call kobject_release and delay the release of kobject, which will cause use-after-free when the memory backing the kobject is freed at once. Reported-by: syzbot+08a7d8b51ea048a74ffb(a)syzkaller.appspotmail.com Fixes: a135dfb5de1 ("ALSA: led control - add sysfs kcontrol LED marking layer") Signed-off-by: Dongliang Mu <mudongliangabcd(a)gmail.com> --- v1->v2: fix other similar memory leaks; move kfree(led_card) to a release function. sound/core/control_led.c | 33 ++++++++++++++++++++++++++------- 1 file changed, 26 insertions(+), 7 deletions(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 25f57c14f294..a90e31dbde61 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -17,6 +17,9 @@ MODULE_LICENSE("GPL"); #define MAX_LED (((SNDRV_CTL_ELEM_ACCESS_MIC_LED - SNDRV_CTL_ELEM_ACCESS_SPK_LED) \ >> SNDRV_CTL_ELEM_ACCESS_LED_SHIFT) + 1) +#define to_led_card_dev(_dev) \ + container_of(_dev, struct snd_ctl_led_card, dev) + enum snd_ctl_led_mode { MODE_FOLLOW_MUTE = 0, MODE_FOLLOW_ROUTE, @@ -371,6 +374,21 @@ static void snd_ctl_led_disconnect(struct snd_card *card) snd_ctl_led_refresh(); } +static void snd_ctl_led_card_release(struct device *dev) +{ + struct snd_ctl_led_card *led_card = to_led_card_dev(dev); + + kfree(led_card); +} + +static void snd_ctl_led_release(struct device *dev) +{ +} + +static void snd_ctl_led_dev_release(struct device *dev) +{ +} + /* * sysfs */ @@ -663,6 +681,7 @@ static void snd_ctl_led_sysfs_add(struct snd_card *card) led_card->number = card->number; led_card->led = led; device_initialize(&led_card->dev); + led_card->dev.release = snd_ctl_led_card_release; if (dev_set_name(&led_card->dev, "card%d", card->number) < 0) goto cerr; led_card->dev.parent = &led->dev; @@ -681,7 +700,6 @@ static void snd_ctl_led_sysfs_add(struct snd_card *card) put_device(&led_card->dev); cerr2: printk(KERN_ERR "snd_ctl_led: unable to add card%d", card->number); - kfree(led_card); } } @@ -700,8 +718,7 @@ static void snd_ctl_led_sysfs_remove(struct snd_card *card) snprintf(link_name, sizeof(link_name), "led-%s", led->name); sysfs_remove_link(&card->ctl_dev.kobj, link_name); sysfs_remove_link(&led_card->dev.kobj, "card"); - device_del(&led_card->dev); - kfree(led_card); + device_unregister(&led_card->dev); led->cards[card->number] = NULL; } } @@ -723,6 +740,7 @@ static int __init snd_ctl_led_init(void) device_initialize(&snd_ctl_led_dev); snd_ctl_led_dev.class = sound_class; + snd_ctl_led_dev.release = snd_ctl_led_dev_release; dev_set_name(&snd_ctl_led_dev, "ctl-led"); if (device_add(&snd_ctl_led_dev)) { put_device(&snd_ctl_led_dev); @@ -733,15 +751,16 @@ static int __init snd_ctl_led_init(void) INIT_LIST_HEAD(&led->controls); device_initialize(&led->dev); led->dev.parent = &snd_ctl_led_dev; + led->dev.release = snd_ctl_led_release; led->dev.groups = snd_ctl_led_dev_attr_groups; dev_set_name(&led->dev, led->name); if (device_add(&led->dev)) { put_device(&led->dev); for (; group > 0; group--) { led = &snd_ctl_leds[group - 1]; - device_del(&led->dev); + device_unregister(&led->dev); } - device_del(&snd_ctl_led_dev); + device_unregister(&snd_ctl_led_dev); return -ENOMEM; } } @@ -767,9 +786,9 @@ static void __exit snd_ctl_led_exit(void) } for (group = 0; group < MAX_LED; group++) { led = &snd_ctl_leds[group]; - device_del(&led->dev); + device_unregister(&led->dev); } - device_del(&snd_ctl_led_dev); + device_unregister(&snd_ctl_led_dev); snd_ctl_led_clean(NULL); } -- 2.25.1

4 3

bytcr-rt5640 initiializes but has wrong settings
by GitHub issues - opened 02 Jun '21

02 Jun '21

alsa-project/alsa-ucm-conf issue #96 was opened from AnterCreeper: My device is Elitepad 1000 G2. The headphone and the mic inside cannot work. However, The speaker works properly. The mic IN1 is the headphone mic insted of the mic inside. I think that it may be because the audio was ported to the wrong place. Sorry for my bad English :D Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/96 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[syzbot] UBSAN: shift-out-of-bounds in snd_timer_user_ccallback
by syzbot 02 Jun '21

02 Jun '21

Hello, syzbot found the following issue on: HEAD commit: 5ff2756a Merge tag 'nfs-for-5.13-2' of git://git.linux-nfs.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=17872d5bd00000 kernel config: https://syzkaller.appspot.com/x/.config?x=770708ea7cfd4916 dashboard link: https://syzkaller.appspot.com/bug?extid=d102fa5b35335a7e544e Unfortunately, I don't have any reproducer for this issue yet. IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+d102fa5b35335a7e544e(a)syzkaller.appspotmail.com ================================================================================ UBSAN: shift-out-of-bounds in sound/core/timer.c:1376:23 shift exponent 105 is too large for 32-bit type 'int' CPU: 1 PID: 10368 Comm: syz-executor.1 Not tainted 5.13.0-rc3-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:79 [inline] dump_stack+0x141/0x1d7 lib/dump_stack.c:120 ubsan_epilogue+0xb/0x5a lib/ubsan.c:148 __ubsan_handle_shift_out_of_bounds.cold+0xb1/0x181 lib/ubsan.c:327 snd_timer_user_ccallback.cold+0x19/0x1e sound/core/timer.c:1376 snd_timer_notify1+0x243/0x3b0 sound/core/timer.c:525 snd_timer_stop1+0x641/0x890 sound/core/timer.c:656 snd_timer_pause+0x6e/0x90 sound/core/timer.c:738 seq_timer_stop sound/core/seq/seq_timer.c:331 [inline] snd_seq_timer_stop+0x93/0xd0 sound/core/seq/seq_timer.c:341 queue_delete+0x42/0xa0 sound/core/seq/seq_queue.c:133 snd_seq_queue_client_leave+0x37/0x1a0 sound/core/seq/seq_queue.c:552 seq_free_client1.part.0+0x10a/0x260 sound/core/seq/seq_clientmgr.c:280 seq_free_client1 sound/core/seq/seq_clientmgr.c:273 [inline] seq_free_client+0x7b/0xf0 sound/core/seq/seq_clientmgr.c:301 snd_seq_release+0x4d/0xe0 sound/core/seq/seq_clientmgr.c:382 __fput+0x288/0x920 fs/file_table.c:280 task_work_run+0xdd/0x1a0 kernel/task_work.c:164 tracehook_notify_resume include/linux/tracehook.h:189 [inline] exit_to_user_mode_loop kernel/entry/common.c:174 [inline] exit_to_user_mode_prepare+0x272/0x280 kernel/entry/common.c:208 __syscall_exit_to_user_mode_work kernel/entry/common.c:290 [inline] syscall_exit_to_user_mode+0x19/0x60 kernel/entry/common.c:301 do_syscall_64+0x47/0xb0 arch/x86/entry/common.c:57 entry_SYSCALL_64_after_hwframe+0x44/0xae RIP: 0033:0x4193eb Code: 0f 05 48 3d 00 f0 ff ff 77 45 c3 0f 1f 40 00 48 83 ec 18 89 7c 24 0c e8 63 fc ff ff 8b 7c 24 0c 41 89 c0 b8 03 00 00 00 0f 05 <48> 3d 00 f0 ff ff 77 35 44 89 c7 89 44 24 0c e8 a1 fc ff ff 8b 44 RSP: 002b:0000000000a9fb80 EFLAGS: 00000293 ORIG_RAX: 0000000000000003 RAX: 0000000000000000 RBX: 0000000000000005 RCX: 00000000004193eb RDX: 0000000000571b58 RSI: ffffffff89166efe RDI: 0000000000000004 RBP: 0000000000000001 R08: 0000000000000000 R09: 0000001b2ea280a8 R10: 00000000000001d1 R11: 0000000000000293 R12: 000000000014d248 R13: 00000000000003e8 R14: 000000000056bf80 R15: 000000000014d1dd ================================================================================ --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot.

1 0

[PATCH 0/4] ASoC: rsnd: adjust disabled module for R-Car D3
by Kuninori Morimoto 01 Jun '21

01 Jun '21

Hi Mark, Geert Renesas Sound driver is assuming that all SSI/SRC are connected. But some SSI/SRC are not connected on Some Renesas SoC. ex) H2 E2 SRC0 <= SRC1 SRC1 SRC2 SRC2 ... ... We accepted it by using "status = disabled" on DT before. ex) rcar_sound,src { src-0 { => status = "disabled"; }; src1: src-1 { ... }; ... But R-Car D3 have many disabled modules (It has SSI3/SSI4, SRC5/SRC6), and Renesas SoC maintainer don't want above style on DT. ex) rcar_sound,src { => src0: src-0 { status = "disabled"; }; => src1: src-1 { status = "disabled"; }; => src2: src-2 { status = "disabled"; }; => src3: src-3 { status = "disabled"; }; => src4: src-4 { status = "disabled"; }; src5: src-5 { ... }; ... This patch-set adjust to this situation, and enables to intuitive DT settings. rcar_sound,src { src5: src-5 { ... }; src6: src-6 { ... }; }; Kuninori Morimoto (4): ASoC: rsnd: tidyup rsnd_parse_connect_common() ASoC: rsnd: tidyup rsnd_dma_request_channel() ASoC: rsnd: tidyup rsnd_parse_connect_xxx() ASoC: rsnd: adjust disabled module sound/soc/sh/rcar/core.c | 58 ++++++++++++++++++++++++++++++++++++++-- sound/soc/sh/rcar/dma.c | 8 +++--- sound/soc/sh/rcar/dvc.c | 2 +- sound/soc/sh/rcar/rsnd.h | 16 ++++++----- sound/soc/sh/rcar/src.c | 6 +++-- sound/soc/sh/rcar/ssi.c | 12 ++++++--- sound/soc/sh/rcar/ssiu.c | 10 ++++--- 7 files changed, 91 insertions(+), 21 deletions(-) -- 2.25.1

2 5

[PATCH 1/2] ASoC: SOF: Intel: hda: clean up hda_dsp_dump()
by Kai Vehmanen 01 Jun '21

01 Jun '21

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Clean up the hda_dsp_dump() function to avoid duplicating the ROM status and error. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/intel/hda.c | 17 +++++------------ 1 file changed, 5 insertions(+), 12 deletions(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 126232a76a10..e1e368ff2b12 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -394,28 +394,21 @@ void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags) struct sof_ipc_dsp_oops_xtensa xoops; struct sof_ipc_panic_info panic_info; u32 stack[HDA_DSP_STACK_DUMP_SIZE]; - u32 status, panic; - /* try APL specific status message types first */ + /* print ROM/FW status */ hda_dsp_get_status(sdev); - /* now try generic SOF status messages */ - status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, - HDA_DSP_SRAM_REG_FW_STATUS); - panic = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_TRACEP); - + /* print panic info if FW boot is complete. Otherwise, print the extended ROM status */ if (sdev->fw_state == SOF_FW_BOOT_COMPLETE) { + u32 status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_STATUS); + u32 panic = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_TRACEP); + hda_dsp_get_registers(sdev, &xoops, &panic_info, stack, HDA_DSP_STACK_DUMP_SIZE); snd_sof_get_status(sdev, status, panic, &xoops, &panic_info, stack, HDA_DSP_STACK_DUMP_SIZE); } else { - sof_dev_dbg_or_err(sdev->dev, flags & SOF_DBG_DUMP_FORCE_ERR_LEVEL, - "status = 0x%8.8x panic = 0x%8.8x\n", - status, panic); - hda_dsp_dump_ext_rom_status(sdev, flags); - hda_dsp_get_status(sdev); } } base-commit: 058efb40641845432c52777443b3372dbc97c032 -- 2.31.1

2 2

[PATCH] ASoC: SOF: Intel: pci-tgl: add ADL-M support
by Kai Vehmanen 01 Jun '21

01 Jun '21

Add PCI DID for Intel AlderLake-M. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/pci-tgl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index 88c3bf404dd7..a00262184efa 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -116,6 +116,8 @@ static const struct pci_device_id sof_pci_ids[] = { .driver_data = (unsigned long)&adls_desc}, { PCI_DEVICE(0x8086, 0x51c8), /* ADL-P */ .driver_data = (unsigned long)&adl_desc}, + { PCI_DEVICE(0x8086, 0x51cc), /* ADL-M */ + .driver_data = (unsigned long)&adl_desc}, { 0, } }; MODULE_DEVICE_TABLE(pci, sof_pci_ids); base-commit: 058efb40641845432c52777443b3372dbc97c032 -- 2.31.1

3 2

[PATCH] ASoC: fsl-asoc-card: Set .owner attribute when registering card.
by Nicolas Cavallari 01 Jun '21

01 Jun '21

Otherwise, when compiled as module, a WARN_ON is triggered: WARNING: CPU: 0 PID: 5 at sound/core/init.c:208 snd_card_new+0x310/0x39c [snd] [...] CPU: 0 PID: 5 Comm: kworker/0:0 Not tainted 5.10.39 #1 Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) Workqueue: events deferred_probe_work_func [<c0111988>] (unwind_backtrace) from [<c010c8ac>] (show_stack+0x10/0x14) [<c010c8ac>] (show_stack) from [<c092784c>] (dump_stack+0xdc/0x104) [<c092784c>] (dump_stack) from [<c0129710>] (__warn+0xd8/0x114) [<c0129710>] (__warn) from [<c0922a48>] (warn_slowpath_fmt+0x5c/0xc4) [<c0922a48>] (warn_slowpath_fmt) from [<bf0496f8>] (snd_card_new+0x310/0x39c [snd]) [<bf0496f8>] (snd_card_new [snd]) from [<bf1d7df8>] (snd_soc_bind_card+0x334/0x9c4 [snd_soc_core]) [<bf1d7df8>] (snd_soc_bind_card [snd_soc_core]) from [<bf1e9cd8>] (devm_snd_soc_register_card+0x30/0x6c [snd_soc_core]) [<bf1e9cd8>] (devm_snd_soc_register_card [snd_soc_core]) from [<bf22d964>] (fsl_asoc_card_probe+0x550/0xcc8 [snd_soc_fsl_asoc_card]) [<bf22d964>] (fsl_asoc_card_probe [snd_soc_fsl_asoc_card]) from [<c060c930>] (platform_drv_probe+0x48/0x98) [...] Signed-off-by: Nicolas Cavallari <nicolas.cavallari(a)green-communications.fr> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index c62bfd1c3ac7..4f55b316cf0f 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -744,6 +744,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Initialize sound card */ priv->pdev = pdev; priv->card.dev = &pdev->dev; + priv->card.owner = THIS_MODULE; ret = snd_soc_of_parse_card_name(&priv->card, "model"); if (ret) { snprintf(priv->name, sizeof(priv->name), "%s-audio", -- 2.32.0.rc0

3 2

[PATCH] ASoC: SOF: reset enabled_cores state at suspend
by Kai Vehmanen 01 Jun '21

01 Jun '21

The recent changes to use common code to power up/down DSP cores also removed the reset of the core state at suspend. It turns out this is still needed. When the firmware state is reset to SOF_FW_BOOT_NOT_STARTED, also enabled_cores should be reset, and existing DSP drivers depend on this. BugLink: https://github.com/thesofproject/linux/issues/2824 Fixes: 42077f08b3 ("ASoC: SOF: update dsp core power status in common APIs") Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/pm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index fd265803f7bc..c83fb6255961 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -256,6 +256,7 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) /* reset FW state */ sdev->fw_state = SOF_FW_BOOT_NOT_STARTED; + sdev->enabled_cores_mask = 0; return ret; } base-commit: 058efb40641845432c52777443b3372dbc97c032 -- 2.31.1

2 1

[PATCH] ASoC: topology: Fix spelling mistake "vesion" -> "version"
by Colin King 01 Jun '21

01 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> There are spelling mistakes in comments. Fix them. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/soc-topology.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 5e65b72910e9..0a24d0d409d2 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1881,7 +1881,7 @@ static void stream_caps_new_ver(struct snd_soc_tplg_stream_caps *dest, * @src: older version of pcm as a source * @pcm: latest version of pcm created from the source * - * Support from vesion 4. User should free the returned pcm manually. + * Support from version 4. User should free the returned pcm manually. */ static int pcm_new_ver(struct soc_tplg *tplg, struct snd_soc_tplg_pcm *src, @@ -2069,7 +2069,7 @@ static void set_link_hw_format(struct snd_soc_dai_link *link, * @src: old version of phyical link config as a source * @link: latest version of physical link config created from the source * - * Support from vesion 4. User need free the returned link config manually. + * Support from version 4. User need free the returned link config manually. */ static int link_new_ver(struct soc_tplg *tplg, struct snd_soc_tplg_link_config *src, @@ -2380,7 +2380,7 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg, * @src: old version of manifest as a source * @manifest: latest version of manifest created from the source * - * Support from vesion 4. Users need free the returned manifest manually. + * Support from version 4. Users need free the returned manifest manually. */ static int manifest_new_ver(struct soc_tplg *tplg, struct snd_soc_tplg_manifest *src, -- 2.31.1

2 1

[PATCH -next] ASoC: ti: omap-mcbsp: use DEVICE_ATTR_RW macro
by YueHaibing 01 Jun '21

01 Jun '21

Use DEVICE_ATTR_RW() helper instead of plain DEVICE_ATTR(), which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/ti/omap-mcbsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index db47981768c5..4479d74f0a45 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -539,7 +539,7 @@ static ssize_t prop##_store(struct device *dev, \ return size; \ } \ \ -static DEVICE_ATTR(prop, 0644, prop##_show, prop##_store) +static DEVICE_ATTR_RW(prop) THRESHOLD_PROP_BUILDER(max_tx_thres); THRESHOLD_PROP_BUILDER(max_rx_thres); -- 2.17.1

4 3

[PATCH 0/5] ASoC: Constify snd_compress_ops
by Rikard Falkeborn 01 Jun '21

01 Jun '21

The only use of the static and global snd_compress_ops structs is to assign their address to the compress_ops field in the snd_soc_component_driver struct which is a pointer to const. Make them const to allow the compiler to put them in read-only memory. Rikard Falkeborn (5): ASoC: cs47125: Constify static struct snd_compress_ops ASoC: wm5102: Constify static struct snd_compress_ops ASoC: wm5110: Constify static struct snd_compress_ops ASoC: qcom: q6asm-dai: Constify static struct snd_compress_ops ASoC: SOF: Intel: Constify sof_probe_compressed_ops sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 2 +- sound/soc/sof/compress.c | 2 +- sound/soc/sof/compress.h | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) -- 2.31.1

3 7

alsa-lib 1.2.5 no devices on Lenovo Carbon (gen 7) - pipewire
by GitHub issues - edited 01 Jun '21

01 Jun '21

alsa-project/alsa-lib issue #142 was edited from sambo57u: I use Fedora 34 and updated alsa-lib and all other alsa products to version 1.2.5. They work fine on my desktop machines but on Lenovo laptop I am getting the message alsactl cannot find hw:0 and daemon deactivates with no devices showing. Downgrading to 1.2.4 solves the problem. Issue URL : https://github.com/alsa-project/alsa-lib/issues/142 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Incomplete sound device detection in PulseAudio with alsa-lib 1.2.5
by GitHub issues - opened 01 Jun '21

01 Jun '21

alsa-project/alsa-lib issue #143 was opened from foutrelis: (Copying my comments over from #142 since it appears to be a different issue to that one. Sorry for the noise there!) After upgrading to alsa-lib 1.2.5 Chromium stopped playing audio until pulseaudio was restarted. Upon closer inspection, `pactl list cards` is now missing the `HDA Intel HDMI` card and only shows the `HDA Intel PCH` one. Some error messages I've gathered: - After upgrading to alsa-lib and before restarting pulseaudio, attempting to play a video in Chromium would result in: `pulseaudio[625]: Error opening PCM device front:0: Invalid argument` - Restarting pulseaudio with alsa-lib 1.2.5 installed logs the following to the journal: ``` pulseaudio[66345]: Failed to find a working profile. pulseaudio[66345]: Failed to load module "module-alsa-card" (argument: "device_id="1" name="pci-0000_00_03.0" card_name="alsa_card.pci-0000_00_03.0" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. ``` Git bisect points to commit 63f7745be504e447923f2cde421177a2fca99340 as the first commit that results in the `Failed to find a working profile` error. After restarting pulseaudio with alsa-lib 1.2.5 installed, it appears that `pactl list cards` shows one fewer cards on my laptop which has this audio hardware: ``` $ lspci | grep Audio 00:03.0 Audio device: Intel Corporation Haswell-ULT HD Audio Controller (rev 0b) 00:1b.0 Audio device: Intel Corporation 8 Series HD Audio Controller (rev 04) ``` [alsa-lib-1.2.4-pactl-list-cards.txt](https://github.com/alsa-project/alsa-l… [alsa-lib-1.2.5-pactl-list-cards.txt](https://github.com/alsa-project/alsa-l… `pulseaudio -v` output: [pulseaudio-verbose.txt](https://github.com/alsa-project/alsa-lib/files/6577… Issue URL : https://github.com/alsa-project/alsa-lib/issues/143 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/5] ALSA: usb-audio: Delay account cleanup and latency reduction
by Takashi Iwai 01 Jun '21

01 Jun '21

Hi, this is a patch set for USB-audio driver for the code refactoring of PCM delay account as well as the reduction of the playback latency. The last change is a bit intrusive (despite of the smallness), and hopefully it won't break things. My initial tests through various backends were OK, but if anyone finds a problem, please let me know. Takashi === Takashi Iwai (5): ALSA: usb-audio: Make snd_usb_pcm_delay() static ALSA: usb-audio: Pre-calculate buffer byte size ALSA: usb-audio: Refactoring delay account code ALSA: usb-audio: Factor out DSD bitrev copy function ALSA: usb-audio: Reduce latency at playback start sound/usb/card.h | 8 +- sound/usb/endpoint.c | 1 + sound/usb/pcm.c | 202 ++++++++++++++++++++----------------------- sound/usb/pcm.h | 3 - 4 files changed, 101 insertions(+), 113 deletions(-) -- 2.26.2

1 5

[PATCH] arm64: dts: qcom: sc7180: add label for secondary mi2s
by Judy Hsiao 01 Jun '21

01 Jun '21

Adds label for MI2S secondary block to allow follower projects to override for the four speaker support which uses I2S SD1 line on gpio52 pin. Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> --- arch/arm64/boot/dts/qcom/sc7180-trogdor.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/qcom/sc7180-trogdor.dtsi b/arch/arm64/boot/dts/qcom/sc7180-trogdor.dtsi index 24d293ef56d7..2027914a0bed 100644 --- a/arch/arm64/boot/dts/qcom/sc7180-trogdor.dtsi +++ b/arch/arm64/boot/dts/qcom/sc7180-trogdor.dtsi @@ -772,7 +772,7 @@ mi2s@0 { qcom,capture-sd-lines = <0>; }; - mi2s@1 { + secondary_mi2s: mi2s@1 { reg = <MI2S_SECONDARY>; qcom,playback-sd-lines = <0>; }; -- 2.32.0.rc0.204.g9fa02ecfa5-goog

3 2

[PATCH] ALSA: control led: fix memory leak in snd_ctl_led_register
by Dongliang Mu 01 Jun '21

01 Jun '21

The snd_ctl_led_sysfs_add and snd_ctl_led_sysfs_remove should contain the refcount operations in pair. However, snd_ctl_led_sysfs_remove fails to decrease the refcount to zero, which causes device_release never to be invoked. This leads to memory leak to some resources, like struct device_private. Fix this by calling put_device at the end of snd_ctl_led_sysfs_remove Reported-by: syzbot+08a7d8b51ea048a74ffb(a)syzkaller.appspotmail.com Fixes: a135dfb5de1 ("ALSA: led control - add sysfs kcontrol LED marking layer") Signed-off-by: Dongliang Mu <mudongliangabcd(a)gmail.com> --- sound/core/control_led.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 25f57c14f294..fff2688b5019 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -371,6 +371,10 @@ static void snd_ctl_led_disconnect(struct snd_card *card) snd_ctl_led_refresh(); } +static void snd_ctl_led_release(struct device *dev) +{ +} + /* * sysfs */ @@ -663,6 +667,7 @@ static void snd_ctl_led_sysfs_add(struct snd_card *card) led_card->number = card->number; led_card->led = led; device_initialize(&led_card->dev); + led_card->dev.release = snd_ctl_led_release; if (dev_set_name(&led_card->dev, "card%d", card->number) < 0) goto cerr; led_card->dev.parent = &led->dev; @@ -701,6 +706,7 @@ static void snd_ctl_led_sysfs_remove(struct snd_card *card) sysfs_remove_link(&card->ctl_dev.kobj, link_name); sysfs_remove_link(&led_card->dev.kobj, "card"); device_del(&led_card->dev); + put_device(&led_card->dev); kfree(led_card); led->cards[card->number] = NULL; } -- 2.25.1

3 26

[PATCH v3] ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
by Srinivasa Rao Mandadapu 01 Jun '21

01 Jun '21

This patch fixes PoP noise of around 15ms observed during audio capture begin. Enables BCLK and LRCLK in snd_soc_dai_ops prepare call for introducing some delay before capture start and clock enable. Co-developed-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes Since V2: -- Updated comments as per linux style -- Removed unrelated changes. Changes Since V1: -- Enableed BCLK and LRCLK in dai ops prepare API instead of startup API -- Added comments sound/soc/qcom/lpass-cpu.c | 54 +++++++++++++++++++++++++++++++++++++- 1 file changed, 53 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 28c7497344e3..1855eae22aad 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -93,9 +93,21 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); - clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); + /* + * To ensure BCLK/LRCLK disabled even in device node validation + * Will not impact if disabled in lpass_cpu_daiops_trigger() + * suspend. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); + else + regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE); + + clk_disable_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, @@ -275,6 +287,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * To ensure lpass BCLK/LRCLK is enabled during + * device resume. Will not impact if enabled in lpass_cpu_daiops_prepare(). + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); @@ -296,6 +312,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* + * To ensure lpass BCLK/LRCLK is disabled during + * device suspend. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); @@ -315,12 +335,44 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, return ret; } +static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret; + /* + * To ensure lpass BCLK/LRCLK is enabled bit before + * playback/capture data flow starts. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); + else + ret = regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_ENABLE); + + if (ret) { + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + return ret; + } + + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + clk_disable(drvdata->mi2s_osr_clk[id]); + return ret; + } + return 0; +} + const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, .trigger = lpass_cpu_daiops_trigger, + .prepare = lpass_cpu_daiops_prepare, }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 2

[PATCH] mfd: arizona: Allow building arizona MFD-core as module
by Hans de Goede 01 Jun '21

01 Jun '21

There is no reason why the arizona core,irq and codec model specific regmap bits cannot be build as a module. All they do is export symbols which are used by the arizona-spi/i2c and arizona-codec modules, which themselves can be built as module. Change the Kconfig and Makefile arizona bits so that the arizona MFD-core can be built as a module. This is especially useful on x86 platforms with a WM5102 codec, this allows the arizona MFD driver necessary for the WM5102 codec to be enabled in generic distro-kernels without growing the base kernel-image size. Note this also adds an explicit "depends on MFD_ARIZONA" to all the arizona codec Kconfig options. The codec drivers use functions from mfd arizona-core. These new depends are necessary to disallow the codec drivers being builtin when the arizona-core is build as a module, otherwise we end up with missing symbol errors when building vmlinuz. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Changes in v2: - Add explicit "depends on MFD_ARIZONA" to all the arizona codec Kconfigs --- drivers/mfd/Kconfig | 2 +- drivers/mfd/Makefile | 14 +++++++------- drivers/mfd/arizona-core.c | 2 ++ sound/soc/codecs/Kconfig | 10 +++++----- 4 files changed, 15 insertions(+), 13 deletions(-) diff --git a/drivers/mfd/Kconfig b/drivers/mfd/Kconfig index 5c7f2b100191..3fb048361b11 100644 --- a/drivers/mfd/Kconfig +++ b/drivers/mfd/Kconfig @@ -1788,7 +1788,7 @@ config MFD_ARIZONA select REGMAP select REGMAP_IRQ select MFD_CORE - bool + tristate config MFD_ARIZONA_I2C tristate "Cirrus Logic/Wolfson Microelectronics Arizona platform with I2C" diff --git a/drivers/mfd/Makefile b/drivers/mfd/Makefile index 4f6d2b8a5f76..ebc9b6704e87 100644 --- a/drivers/mfd/Makefile +++ b/drivers/mfd/Makefile @@ -41,24 +41,24 @@ obj-$(CONFIG_MFD_TQMX86) += tqmx86.o obj-$(CONFIG_MFD_LOCHNAGAR) += lochnagar-i2c.o -obj-$(CONFIG_MFD_ARIZONA) += arizona-core.o -obj-$(CONFIG_MFD_ARIZONA) += arizona-irq.o +arizona-objs := arizona-core.o arizona-irq.o +obj-$(CONFIG_MFD_ARIZONA) += arizona.o obj-$(CONFIG_MFD_ARIZONA_I2C) += arizona-i2c.o obj-$(CONFIG_MFD_ARIZONA_SPI) += arizona-spi.o ifeq ($(CONFIG_MFD_WM5102),y) -obj-$(CONFIG_MFD_ARIZONA) += wm5102-tables.o +arizona-objs += wm5102-tables.o endif ifeq ($(CONFIG_MFD_WM5110),y) -obj-$(CONFIG_MFD_ARIZONA) += wm5110-tables.o +arizona-objs += wm5110-tables.o endif ifeq ($(CONFIG_MFD_WM8997),y) -obj-$(CONFIG_MFD_ARIZONA) += wm8997-tables.o +arizona-objs += wm8997-tables.o endif ifeq ($(CONFIG_MFD_WM8998),y) -obj-$(CONFIG_MFD_ARIZONA) += wm8998-tables.o +arizona-objs += wm8998-tables.o endif ifeq ($(CONFIG_MFD_CS47L24),y) -obj-$(CONFIG_MFD_ARIZONA) += cs47l24-tables.o +arizona-objs += cs47l24-tables.o endif obj-$(CONFIG_MFD_WCD934X) += wcd934x.o obj-$(CONFIG_MFD_WM8400) += wm8400-core.o diff --git a/drivers/mfd/arizona-core.c b/drivers/mfd/arizona-core.c index ce6fe6de34f8..9323b1e3a69e 100644 --- a/drivers/mfd/arizona-core.c +++ b/drivers/mfd/arizona-core.c @@ -1447,3 +1447,5 @@ int arizona_dev_exit(struct arizona *arizona) return 0; } EXPORT_SYMBOL_GPL(arizona_dev_exit); + +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2a7b3e363069..45aa92598066 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -689,7 +689,7 @@ config SND_SOC_CS47L15 config SND_SOC_CS47L24 tristate - depends on MFD_CS47L24 + depends on MFD_CS47L24 && MFD_ARIZONA config SND_SOC_CS47L35 tristate @@ -1558,11 +1558,11 @@ config SND_SOC_WM5100 config SND_SOC_WM5102 tristate - depends on MFD_WM5102 + depends on MFD_WM5102 && MFD_ARIZONA config SND_SOC_WM5110 tristate - depends on MFD_WM5110 + depends on MFD_WM5110 && MFD_ARIZONA config SND_SOC_WM8350 tristate @@ -1727,11 +1727,11 @@ config SND_SOC_WM8996 config SND_SOC_WM8997 tristate - depends on MFD_WM8997 + depends on MFD_WM8997 && MFD_ARIZONA config SND_SOC_WM8998 tristate - depends on MFD_WM8998 + depends on MFD_WM8998 && MFD_ARIZONA config SND_SOC_WM9081 tristate -- 2.31.1

3 2

[PATCH 0/2] Cirrus Logic CS35L41 Amplifier
by David Rhodes 01 Jun '21

01 Jun '21

ASoC driver and devicetree documentation for a new Cirrus Logic amplifier CS35L41 David Rhodes (2): ASoC: cs35l41: CS35L41 Boosted Smart Amplifier ASoC: cs35l41: Add bindings for CS35L41 .../devicetree/bindings/sound/cs35l41.yaml | 395 ++++ include/sound/cs35l41.h | 83 + sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs35l41-i2c.c | 126 ++ sound/soc/codecs/cs35l41-spi.c | 148 ++ sound/soc/codecs/cs35l41-tables.c | 688 +++++++ sound/soc/codecs/cs35l41.c | 1758 +++++++++++++++++ sound/soc/codecs/cs35l41.h | 755 +++++++ 9 files changed, 3969 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l41.yaml create mode 100644 include/sound/cs35l41.h create mode 100644 sound/soc/codecs/cs35l41-i2c.c create mode 100644 sound/soc/codecs/cs35l41-spi.c create mode 100644 sound/soc/codecs/cs35l41-tables.c create mode 100644 sound/soc/codecs/cs35l41.c create mode 100644 sound/soc/codecs/cs35l41.h -- 2.25.1

2 4

trivial patch
by josef radinger 01 Jun '21

01 Jun '21

just fix some trivial typo in comments, sorry for the second try. i should have read more carefully the rule about NOT sending attachments. diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 3c68df82c95c..1855b805eba2 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -373,7 +373,7 @@ static int q6adm_device_open(struct q6adm *adm, struct q6copp *copp, * @rate: rate at which copp is required. * @channel_mode: channel mode * @topology: adm topology id - * @perf_mode: performance mode. + * @perf_mode: performace mode. * @bit_width: audio sample bit width * @app_type: Application type. * @acdb_id: ACDB id @@ -452,7 +452,7 @@ EXPORT_SYMBOL_GPL(q6adm_get_copp_id); * @dev: Pointer to adm child device. * @path: playback or capture path. * @payload_map: map between session id and afe ports. - * @perf_mode: Performance mode. + * @perf_mode: Performace mode. * * Return: Will be an negative on error or a zero on success. */ yours josef

2 1

[PATCH] ALSA: hda: Fix for mute key LED for HP Pavilion 15-CK0xx
by Carlos M 01 Jun '21

01 Jun '21

For the HP Pavilion 15-CK0xx, with audio subsystem ID 0x103c:0x841c, adding a line in patch_realtek.c to apply the ALC269_FIXUP_HP_MUTE_LED_MIC3 fix activates the mute key LED. Signed-off-by: Carlos M <carlos.marr.pz(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 61a60c420..43e37145e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8303,6 +8303,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x82bf, "HP G3 mini", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x82c0, "HP G3 mini premium", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x841c, "HP Pavilion 15-CK0xx", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), -- 2.31.1

2 1

[PATCH v1 1/1] ALSA: hda/cirrus: Set Initial DMIC volume to -26 dB
by Vitaly Rodionov 01 Jun '21

01 Jun '21

From: Stefan Binding <sbinding(a)opensource.cirrus.com> Previously this fix was applied only to Bullseye variant laptops, and should be applied to Cyborg and Warlock variants. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/pci/hda/patch_cirrus.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 726507d0b04c..8629e84fef23 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -2206,10 +2206,9 @@ static void cs8409_cs42l42_fixups(struct hda_codec *codec, break; case HDA_FIXUP_ACT_PROBE: - /* Set initial volume on Bullseye to -26 dB */ - if (codec->fixup_id == CS8409_BULLSEYE) - snd_hda_codec_amp_init_stereo(codec, CS8409_CS42L42_DMIC_ADC_PIN_NID, - HDA_INPUT, 0, 0xff, 0x19); + /* Set initial DMIC volume to -26 dB */ + snd_hda_codec_amp_init_stereo(codec, CS8409_CS42L42_DMIC_ADC_PIN_NID, + HDA_INPUT, 0, 0xff, 0x19); snd_hda_gen_add_kctl(&spec->gen, NULL, &cs8409_cs42l42_hp_volume_mixer); snd_hda_gen_add_kctl(&spec->gen, -- 2.25.1

2 1

[PATCH 0/6] ALSA: firewire: media clock recovery for syt-unaware devices
by Takashi Sakamoto 01 Jun '21

01 Jun '21

Hi, In a commit f9e5ecdfc2c2 ("ALSA: firewire-lib: add replay target to cache sequence of packet"), I categorize devices supported by drivers in ALSA firewire stack in terms of the way to deliver effective sampling transfer frequency. This patchset is for the devices in group 1. The devices in the group ignore presentation time to decide playback timing, therefore drivers to handle them can ignore processing presentation time expressed in syt field of CIP header. The sequence of the number of data blocks per packet is important for media clock recovery. With the patchset, the drivers are going to replay the sequence. For the detail of sequence replay, please refer to a commit 39c2649c71d8 ("ALSA: firewire-lib: replay sequence of incoming packets for outgoing packets"). Takashi Sakamoto (6): ALSA: fireworks: delete SYTMATCH clock source ALSA: fireworks: perform sequence replay for media clock recovery ALSA: oxfw: perform sequence replay for media clock recovery ALSA: firewire-digi00x: perform sequence replay for media clock recovery ALSA: firewire-tascam: perform sequence replay for media clock recovery ALSA: fireface: perform sequence replay for media clock recovery sound/firewire/digi00x/amdtp-dot.c | 9 +++------ sound/firewire/digi00x/digi00x-stream.c | 8 ++++++-- sound/firewire/fireface/amdtp-ff.c | 2 +- sound/firewire/fireface/ff-stream.c | 6 +++++- sound/firewire/fireworks/fireworks.h | 2 +- sound/firewire/fireworks/fireworks_stream.c | 18 ++++++++++++++--- sound/firewire/oxfw/oxfw-stream.c | 22 +++++++++++++++++---- sound/firewire/tascam/tascam-stream.c | 21 ++++++++++++++++++-- sound/firewire/tascam/tascam.h | 1 + 9 files changed, 69 insertions(+), 20 deletions(-) -- 2.27.0

2 7

[PATCH] ALSA: hda: Fix a regression in Capture Switch mixer read
by Takashi Iwai 31 May '21

31 May '21

The recent commit to drop the HDA-specific mute-LED control, e65bf99718b5 ("ALSA: HDA - remove the custom implementation for the audio LED trigger"), caused a regression on the mixer element read for "Capture Switch" when it's built from bind controls. The function create_bind_cap_vol_ctl() creates the snd_kcontrol_new object directly via snd_hda_gen_add_kctl() instead of add_control(). Although the commit above added a workaround for the SNDRV_CTL_ACCESS_READWRITE in add_control() as default, this code path fell out from the radar. As a result, now the driver gives -EPERM error because of the lack of the proper access bit at reading "Capture Switch" element value. Fix the regression by setting the access bit properly. Fixes: e65bf99718b5 ("ALSA: HDA - remove the custom implementation for the audio LED trigger") BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1186634 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b638fc2ef6f7..1f8018f9ce57 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3520,6 +3520,7 @@ static int cap_sw_put(struct snd_kcontrol *kcontrol, static const struct snd_kcontrol_new cap_sw_temp = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, .info = cap_sw_info, .get = cap_sw_get, .put = cap_sw_put, -- 2.26.2

1 0

Release tarballs ship other version, break build scripts
by GitHub issues - opened 31 May '21

31 May '21

alsa-project/alsa-ucm-conf issue #95 was opened from dvzrv: Hi! I package this for Arch Linux. The release tarballs on the ALSA website ship the 1.2.5 data in a directory with a differing version number: `alsa-ucm-conf-1.2.4.81.g4884e/` This breaks build system assumptions about the contents of these tarballs. Please provide the data in directories that match the release version. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/95 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] arm64: dts: qcom: sc7180: add lable for secondary mi2s
by Judy Hsiao 31 May '21

31 May '21

Adds lable for secondary mi2s to allow overriding. Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> --- arch/arm64/boot/dts/qcom/sc7180-trogdor.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/qcom/sc7180-trogdor.dtsi b/arch/arm64/boot/dts/qcom/sc7180-trogdor.dtsi index 24d293ef56d7..2027914a0bed 100644 --- a/arch/arm64/boot/dts/qcom/sc7180-trogdor.dtsi +++ b/arch/arm64/boot/dts/qcom/sc7180-trogdor.dtsi @@ -772,7 +772,7 @@ mi2s@0 { qcom,capture-sd-lines = <0>; }; - mi2s@1 { + secondary_mi2s: mi2s@1 { reg = <MI2S_SECONDARY>; qcom,playback-sd-lines = <0>; }; -- 2.32.0.rc0.204.g9fa02ecfa5-goog

2 1

alsa-lib 1.2.5 no devices on Lenovo Carbon (gen 7)
by GitHub issues - opened 31 May '21

31 May '21

alsa-project/alsa-lib issue #142 was opened from sambo57u: I use Fedora 34 and updated alsa-lib and all other alsa products to version 1.2.5. They work fine on my desktop machines but on Lenovo laptop I am getting the message alsactl cannot find hw:0 and daemon deactivates with no devices showing. Downgrading to 1.2.4 solves the problem. Issue URL : https://github.com/alsa-project/alsa-lib/issues/142 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ALSA 1.2.4 release
by Jaroslav Kysela 31 May '21

31 May '21

Hello all, new ALSA userspace packages were released. You may download them from the ALSA website http://www.alsa-project.org or directly: FTP: ftp://ftp.alsa-project.org/pub HTTP: https://www.alsa-project.org/files/pub Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.3.2_v1.2.4 All files on the primary ALSA site have also GPG signature now. The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 1

[PATCH] soundwire: cadence: remove the repeated declaration
by Shaokun Zhang 31 May '21

31 May '21

Function 'cdns_reset_page_addr' is declared twice, so remove the repeated declaration. Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Sanyog Kale <sanyog.r.kale(a)intel.com> Signed-off-by: Shaokun Zhang <zhangshaokun(a)hisilicon.com> --- drivers/soundwire/cadence_master.h | 3 --- 1 file changed, 3 deletions(-) diff --git a/drivers/soundwire/cadence_master.h b/drivers/soundwire/cadence_master.h index 4d1aab5b5ec2..0e7f8b35bb21 100644 --- a/drivers/soundwire/cadence_master.h +++ b/drivers/soundwire/cadence_master.h @@ -180,9 +180,6 @@ enum sdw_command_response cdns_xfer_msg_defer(struct sdw_bus *bus, struct sdw_msg *msg, struct sdw_defer *defer); -enum sdw_command_response -cdns_reset_page_addr(struct sdw_bus *bus, unsigned int dev_num); - int cdns_bus_conf(struct sdw_bus *bus, struct sdw_bus_params *params); int cdns_set_sdw_stream(struct snd_soc_dai *dai, -- 2.7.4

3 2

Add UCM for Nvidia Tegra HDMI Audio
by GitHub pull_request - edited 31 May '21

31 May '21

alsa-project/alsa-ucm-conf pull request #94 was edited from clamor-s: This were used to have audio over HDMI on Asus Transformers. You may find original configs [in pmOS](https://gitlab.com/postmarketOS/pmaports/-/tree/master/device/testing… and paths used [here](https://gitlab.com/postmarketOS/pmaports/-/blob/master/device/testing…. I have adjusted them to use syntax 4 and re-use common codecs. This ucm is universal for all Tegra 3 devices and hopefully Tegra 2 as well. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/94 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/94.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add UCM for Nvidia Tegra HDMI Audio
by GitHub pull_request - opened 31 May '21

31 May '21

alsa-project/alsa-ucm-conf pull request #94 was opened from clamor-s: This were used to have audio over HDMI on Asus Transformers. You may find original configs [in pmOS](https://gitlab.com/postmarketOS/pmaports/-/tree/master/device/testing… and paths used [here](https://gitlab.com/postmarketOS/pmaports/-/blob/master/device/testing…. I have adjusted them to use syntax 4 and re-use common codecs. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/94 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/94.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

topology: fix potential null pointer from strchr
by GitHub pull_request - opened 31 May '21

31 May '21

alsa-project/alsa-utils pull request #91 was opened from aiChaoSONG: This patch adds check to the return pointer from strchr, because it may be null and cause segment fault, if component is not properly constructed. Signed-off-by: Chao Song <chao.song(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/91 Patch URL : https://github.com/alsa-project/alsa-utils/pull/91.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ALSA: hda: Add AlderLake-M PCI ID
by Kai Vehmanen 30 May '21

30 May '21

Add HD Audio PCI ID for Intel AlderLake-M. Add rules to snd_intel_dsp_find_config() to choose SOF driver for ADL-M systems with PCH-DMIC or Soundwire codecs, and legacy driver for the rest. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/hda/intel-dsp-config.c | 4 ++++ sound/pci/hda/hda_intel.c | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index ab5ff7867eb9..d8be146793ee 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -331,6 +331,10 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x51c8, }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x51cc, + }, #endif }; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 79ade335c8a0..470753b36c8a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2485,6 +2485,9 @@ static const struct pci_device_id azx_ids[] = { /* Alderlake-P */ { PCI_DEVICE(0x8086, 0x51c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Alderlake-M */ + { PCI_DEVICE(0x8086, 0x51cc), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, base-commit: 873fd81377b82c00f3660c5a37e36f8846ebe3f5 -- 2.31.1

2 1