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

[PATCH] ASoC: amd: Allow I2S wake event after ACP is powerd On
by Akshu Agrawal 26 Feb '20

26 Feb '20

ACP_PME_EN allows wake interrupt to be generated when I2S wake feature is enabled. On turning ACP On, ACP_PME_EN gets cleared. Setting the bit back ensures that wake event can be received when ACP is On. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/raven/pci-acp3x.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index da60e2ec5535..f25ce50f1a90 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -38,8 +38,13 @@ static int acp3x_power_on(void __iomem *acp3x_base) timeout = 0; while (++timeout < 500) { val = rv_readl(acp3x_base + mmACP_PGFSM_STATUS); - if (!val) + if (!val) { + /* Set PME_EN as after ACP power On, + * PME_EN gets cleared + */ + rv_writel(0x1, acp3x_base + mmACP_PME_EN); return 0; + } udelay(1); } return -ETIMEDOUT; -- 2.17.1

2 1

Applied "ASoC: dt-bindings: Add TAS2563 compatible to the TAS2562 binding" to the asoc tree
by Mark Brown 26 Feb '20

26 Feb '20

The patch ASoC: dt-bindings: Add TAS2563 compatible to the TAS2562 binding has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 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 >From f87cdb1f9937e6f5234e3300804ac156e639bc00 Mon Sep 17 00:00:00 2001 From: Dan Murphy <dmurphy(a)ti.com> Date: Wed, 26 Feb 2020 07:03:03 -0600 Subject: [PATCH] ASoC: dt-bindings: Add TAS2563 compatible to the TAS2562 binding Add the Texas Instruments TAS2563 audio amplifier to the TAS262 binding. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> CC: Rob Herring <robh(a)kernel.org> Link: https://lore.kernel.org/r/20200226130305.12043-1-dmurphy@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/tas2562.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt index 658e1fb18a99..94796b547184 100644 --- a/Documentation/devicetree/bindings/sound/tas2562.txt +++ b/Documentation/devicetree/bindings/sound/tas2562.txt @@ -8,7 +8,7 @@ real time monitoring of loudspeaker behavior. Required properties: - #address-cells - Should be <1>. - #size-cells - Should be <0>. - - compatible: - Should contain "ti,tas2562". + - compatible: - Should contain "ti,tas2562", "ti,tas2563". - reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f. - ti,imon-slot-no:- TDM TX current sense time slot. -- 2.20.1

1 0

Applied "ASoC: tas2562: Add entries for the TAS2563 audio amplifier" to the asoc tree
by Mark Brown 26 Feb '20

26 Feb '20

The patch ASoC: tas2562: Add entries for the TAS2563 audio amplifier has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 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 >From 14f8c8d8fd62207f081549d45099a90dd3717696 Mon Sep 17 00:00:00 2001 From: Dan Murphy <dmurphy(a)ti.com> Date: Wed, 26 Feb 2020 07:03:04 -0600 Subject: [PATCH] ASoC: tas2562: Add entries for the TAS2563 audio amplifier The TAS2563 is register compatible with the TAS2562. The main difference is the TAS2563 has a programmable DSP to manage different audio profiles. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> Link: https://lore.kernel.org/r/20200226130305.12043-2-dmurphy@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tas2562.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index d5e04030a0c1..79c3c3d79766 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -55,6 +55,11 @@ struct tas2562_data { int volume_lvl; }; +enum tas256x_model { + TAS2562, + TAS2563, +}; + static int tas2562_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level) { @@ -664,13 +669,15 @@ static int tas2562_probe(struct i2c_client *client, } static const struct i2c_device_id tas2562_id[] = { - { "tas2562", 0 }, + { "tas2562", TAS2562 }, + { "tas2563", TAS2563 }, { } }; MODULE_DEVICE_TABLE(i2c, tas2562_id); static const struct of_device_id tas2562_of_match[] = { { .compatible = "ti,tas2562", }, + { .compatible = "ti,tas2563", }, { }, }; MODULE_DEVICE_TABLE(of, tas2562_of_match); -- 2.20.1

1 0

[RESEND PATCH for-next 1/3] ASoC: tas2562: Fix sample rate error message
by Dan Murphy 26 Feb '20

26 Feb '20

Fix error message for setting the sample rate. It says bitwidth but should say sample rate. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tas2562.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index d5e04030a0c1..c7a4b3b74654 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -271,7 +271,7 @@ static int tas2562_hw_params(struct snd_pcm_substream *substream, ret = tas2562_set_samplerate(tas2562, params_rate(params)); if (ret) - dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret); + dev_err(tas2562->dev, "set sample rate failed, %d\n", ret); return ret; } -- 2.25.0

1 2

[PATCH for-next 1/3] ASoC: dt-bindings: Add TAS2563 compatible to the TAS2562 binding
by Dan Murphy 26 Feb '20

26 Feb '20

Add the Texas Instruments TAS2563 audio amplifier to the TAS262 binding. CC: Rob Herring <robh(a)kernel.org> Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- Documentation/devicetree/bindings/sound/tas2562.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt index 658e1fb18a99..94796b547184 100644 --- a/Documentation/devicetree/bindings/sound/tas2562.txt +++ b/Documentation/devicetree/bindings/sound/tas2562.txt @@ -8,7 +8,7 @@ real time monitoring of loudspeaker behavior. Required properties: - #address-cells - Should be <1>. - #size-cells - Should be <0>. - - compatible: - Should contain "ti,tas2562". + - compatible: - Should contain "ti,tas2562", "ti,tas2563". - reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f. - ti,imon-slot-no:- TDM TX current sense time slot. -- 2.25.0

2 4

[alsa-devel] [RFC PATCH 00/34] sun8i-codec fixes and new features
by Samuel Holland 26 Feb '20

26 Feb '20

Hello all, The sun8i-codec driver, as used in the Allwinner A33 and A64, currently only exposes a small subset of the available hardware features. In order to use the A64 in a smartphone (the PinePhone), I've added the necessary functionality to the driver: * The full set of supported DAI format options * Support for AIF2 and AIF3 * Additional routing knobs * Additional volume controls Unfortunately, due to preexisting issues with the driver, there are some breaking changes, as explained further in the commit messages: * The LRCK inversion issue means we need a new compatible for the A64. * Some controls are named inaccurately, so they are renamed. * Likewise, the DAPM widgets used in device trees were either named wrong, or the device trees were using the wrong widgets in the first place. (Specifically, the links between the analog codec and digital codec happen at the ADC and DAC, not AIF1.) I tended to take the philosophy of "while I'm breaking things, I might as well do them right", so I've probably made a few more changes than absolutely necessary. I'm not sure about where all of the policy boundaries are, about how far I should go to maintain compatibility. For example, for the DT widget usage, I could: * Rename everything and update the DTS files (which is what I did) * Keep the old (misleading/wrong) name for the widgets, but repurpose them to work correctly (i.e. "ADC Left" would be named "AIF1 Slot 0 Left ADC", but it would work just like "ADC Left" does in this patchset) * Keep the old widgets around as a compatibility layer, but add new widgets and update the in-tree DTS files to use them (i.e. "ADC Left" would have a path from "AIF1 Slot 0 Left ADC", but "AIF1 Slot 0 Left ADC" would be a no-op widget) * Something else entirely There are several trivial fixes in here, and there are several commits that just add new features without changing any existing behavior, but there is enough changing that I thought it would be best to send the whole thing as an RFC. I'm more than happy to reorganize this into one or several patchsets in future revisions. It doesn't have to all go in at once. This has all been tested on the PinePhone. For example, I've successfully routed a voice call between the modem at AIF2 and a Bluetooth headset at AIF3, merged with an audio file played from AIF1. Unfortunately, the PinePhone does not have a DTS upstream yet (partially due to the audio driver situation), so I don't have an example DTS in this patchset. Thanks for any feedback, Samuel Samuel Holland (34): ASoC: dt-bindings: Add a separate compatible for the A64 codec ASoC: sun8i-codec: LRCK is not inverted on A64 arm64: dts: allwinner: a64: Fix the audio codec compatible ASoC: sun8i-codec: Remove unused dev from codec struct ASoC: sun8i-codec: Remove incorrect SND_SOC_DAIFMT_DSP_B ASoC: sun8i-codec: Fix setting DAI data format ASoC: sun8i-codec: Remove extraneous widgets ASoC: sun8i-codec: Fix direction of AIF1 outputs ASoC: sun8i-codec: Fix broken DAPM routing ASoC: sun8i-codec: Advertise only hardware-supported rates ASoC: sun8i-codec: Enforce parameter symmetry ASoC: sun8i-codec: Fix AIF1 MODCLK widget name ASoC: sun8i-codec: Fix AIF1_ADCDAT_CTRL field names ASoC: sun8i-codec: Fix AIF1_MXR_SRC field names ASoC: sun8i-codec: Fix ADC_DIG_CTRL field name ASoC: sun8i-codec: Fix field bit number indentation ASoC: sun8i-codec: Sort masks in a consistent order ASoC: sun8i-codec: Allow all clock inversion permutations ASoC: sun8i-codec: Support mono DAI configurations ASoC: sun8i-codec: Support 8/20/24-bit word sizes ASoC: sun8i-codec: Clean up module/clock hierarchy ASoC: sun8i-codec: Clean up AIF1 Slot 0 widgets ASoC: sun8i-codec: Clean up DAC widgets ASoC: sun8i-codec: Prepare to support multiple AIFs ASoC: sun8i-codec: Add support for AIF2 ASoC: sun8i-codec: Add support for AIF3 ASoC: sun8i-codec: Add AIF mono/stereo controls ASoC: sun8i-codec: Add AIF loopback controls ASoC: sun8i-codec: Add AIF, ADC, and DAC volume controls ASoC: dt-bindings: Bump sound-dai-cells on sun8i-codec ARM: dts: sun8i-a33: Allow using multiple codec DAIs arm64: dts: allwinner: a64: Allow using multiple codec DAIs arm64: dts: allwinner: a64: Allow multiple DAI links arm64: dts: allwinner: a64: Add pinmux for AIF2/AIF3 .../sound/allwinner,sun8i-a33-codec.yaml | 6 +- arch/arm/boot/dts/sun8i-a33-olinuxino.dts | 6 +- arch/arm/boot/dts/sun8i-a33.dtsi | 10 +- .../dts/allwinner/sun50i-a64-bananapi-m64.dts | 8 +- .../dts/allwinner/sun50i-a64-orangepi-win.dts | 8 +- .../boot/dts/allwinner/sun50i-a64-pine64.dts | 8 +- .../dts/allwinner/sun50i-a64-pinebook.dts | 8 +- .../boot/dts/allwinner/sun50i-a64-pinetab.dts | 14 +- .../allwinner/sun50i-a64-sopine-baseboard.dts | 8 +- .../boot/dts/allwinner/sun50i-a64-teres-i.dts | 8 +- arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 47 +- sound/soc/sunxi/sun8i-codec.c | 1056 +++++++++++++---- 12 files changed, 890 insertions(+), 297 deletions(-) -- 2.24.1

5 68

[PATCH] soundwire: bus: provide correct return value on error
by Pierre-Louis Bossart 26 Feb '20

26 Feb '20

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> It seems to be a typo. It makes more sense to return the return value of sdw_update() instead of the value we want to update. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- drivers/soundwire/bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 13887713f311..b8a7a84aca1c 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1070,7 +1070,7 @@ static int sdw_initialize_slave(struct sdw_slave *slave) if (ret < 0) { dev_err(slave->bus->dev, "SDW_DP0_INTMASK read failed:%d\n", ret); - return val; + return ret; } return 0; -- 2.20.1

2 2

[PATCH v3 0/2] Introduce the TLV320ADCx140 codec family
by Dan Murphy 26 Feb '20

26 Feb '20

Hello Introducing the Texas Instruments TLV320ADCx140 quad channel audio ADC. This device supports 4 analog inputs, 8 digital inputs or a combination of analog and digital microphones. TLV320ADC3140 - http://www.ti.com/lit/gpn/tlv320adc3140 TLV320ADC5140 - http://www.ti.com/lit/gpn/tlv320adc5140 TLV320ADC6140 - http://www.ti.com/lit/gpn/tlv320adc6140 Dan Murphy (2): dt-bindings: sound: Add TLV320ADCx140 dt bindings ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family .../bindings/sound/tlv320adcx140.yaml | 83 ++ sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adcx140.c | 849 ++++++++++++++++++ sound/soc/codecs/tlv320adcx140.h | 130 +++ 5 files changed, 1073 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320adcx140.yaml create mode 100644 sound/soc/codecs/tlv320adcx140.c create mode 100644 sound/soc/codecs/tlv320adcx140.h -- 2.25.0

3 9

[PATCH v2 0/3] Add new module driver for new ASRC
by Shengjiu Wang 26 Feb '20

26 Feb '20

Add new module driver for new ASRC in i.MX8MN. Shengjiu Wang (3): ASoC: fsl_asrc: Move common definition to fsl_asrc_common ASoC: dt-bindings: fsl_easrc: Add document for EASRC ASoC: fsl_easrc: Add EASRC ASoC CPU DAI and platform drivers changes in v2 - change i.MX815 to i.MX8MN - Add changes in Kconfig and Makefile .../devicetree/bindings/sound/fsl,easrc.txt | 57 + sound/soc/fsl/Kconfig | 10 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_asrc.h | 11 +- sound/soc/fsl/fsl_asrc_common.h | 22 + sound/soc/fsl/fsl_easrc.c | 2265 +++++++++++++++++ sound/soc/fsl/fsl_easrc.h | 668 +++++ sound/soc/fsl/fsl_easrc_dma.c | 440 ++++ 8 files changed, 3465 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,easrc.txt create mode 100644 sound/soc/fsl/fsl_asrc_common.h create mode 100644 sound/soc/fsl/fsl_easrc.c create mode 100644 sound/soc/fsl/fsl_easrc.h create mode 100644 sound/soc/fsl/fsl_easrc_dma.c -- 2.21.0

3 5

[PATCH] ASoC: rt1015: modify some structure to be static and add operation callback function for rt1015_dai
by jack.yu＠realtek.com 26 Feb '20

26 Feb '20

From: Jack Yu <jack.yu(a)realtek.com> 1. Add "static" for rt1015_aif_dai_ops. 2. Add "static" for rt1015_dai[]. 3. Add operation callback for rt1015_dai[]. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index d300b417dd50..bb310bc7febd 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -664,7 +664,7 @@ static int rt1015_hw_params(struct snd_pcm_substream *substream, snd_soc_component_update_bits(component, RT1015_TDM_MASTER, RT1015_I2S_DL_MASK, val_len); snd_soc_component_update_bits(component, RT1015_CLK2, - RT1015_FS_PD_MASK, pre_div); + RT1015_FS_PD_MASK, pre_div << RT1015_FS_PD_SFT); return 0; } @@ -841,12 +841,12 @@ static void rt1015_remove(struct snd_soc_component *component) #define RT1015_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -struct snd_soc_dai_ops rt1015_aif_dai_ops = { +static struct snd_soc_dai_ops rt1015_aif_dai_ops = { .hw_params = rt1015_hw_params, .set_fmt = rt1015_set_dai_fmt, }; -struct snd_soc_dai_driver rt1015_dai[] = { +static struct snd_soc_dai_driver rt1015_dai[] = { { .name = "rt1015-aif", .id = 0, @@ -857,6 +857,7 @@ struct snd_soc_dai_driver rt1015_dai[] = { .rates = RT1015_STEREO_RATES, .formats = RT1015_FORMATS, }, + .ops = &rt1015_aif_dai_ops, } }; -- 2.25.1

2 1