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

[PATCH v2 0/3] ASoC: fsl_sai: update the register list
by Shengjiu Wang 17 Sep '20

17 Sep '20

As sai ip is upgraded, so update sai register list. Shengjiu Wang (3): ASoC: fsl_sai: Add new added registers and new bit definition ASoC: fsl_sai: Add fsl_sai_check_version function ASoC: fsl_sai: Set MCLK input or output direction changes in v2: - update commit message for first commit - Add acked-by Nicolin sound/soc/fsl/fsl_sai.c | 77 ++++++++++++++++++++++++++++++++++++ sound/soc/fsl/fsl_sai.h | 87 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 164 insertions(+) -- 2.27.0

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Re: [PATCH] ASoC: fsl_audmix: make clock and output src write only
by Mark Brown 17 Sep '20

17 Sep '20

On Mon, 14 Sep 2020 20:24:34 +0300, Viorel Suman (OSS) wrote: > "alsactl -f state.conf store/restore" sequence fails because setting > "mixing clock source" and "output source" requires active TDM clock > being started for configuration propagation. Make these two controls > write only so that their values are not stored at "alsactl store". Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: fsl_audmix: make clock and output src write only commit: 944c517b8c838832a166f1c89afbf8724f4a6b49 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

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[PATCH] ASoC: dt-bindings: aries-wm8994: Match compatibles with enum
by Krzysztof Kozlowski 17 Sep '20

17 Sep '20

The common pattern for enumerating compatibles is enum, not oneOf. Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- .../bindings/sound/samsung,aries-wm8994.yaml | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml index 902a0b66628e..1c6947294825 100644 --- a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml @@ -11,12 +11,11 @@ maintainers: properties: compatible: - oneOf: - - const: samsung,aries-wm8994 - description: With FM radio and modem master - - - const: samsung,fascinate4g-wm8994 - description: Without FM radio and modem slave + enum: + # With FM radio and modem master + - samsung,aries-wm8994 + # Without FM radio and modem slave + - samsung,fascinate4g-wm8994 model: $ref: /schemas/types.yaml#/definitions/string -- 2.17.1

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[PATCH 1/6] ASoC: tlv320adcx140: Idle the device while writing registers
by Dan Murphy 17 Sep '20

17 Sep '20

It was observed that if the device was active and register writes were performed there were some unwanted behaviors particularly when writing the word length and some filter options. So when writing to the device the device should be placed in sleep mode and then exit sleep mode once the register update is complete. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tlv320adcx140.c | 43 ++++++++++++++++++++++++-------- sound/soc/codecs/tlv320adcx140.h | 2 ++ 2 files changed, 34 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 8efe20605f9b..3909c1cf52be 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -30,7 +30,7 @@ struct adcx140_priv { struct regmap *regmap; struct device *dev; - int micbias_vg; + bool micbias_vg; unsigned int dai_fmt; unsigned int tdm_delay; @@ -614,11 +614,26 @@ static int adcx140_reset(struct adcx140_priv *adcx140) return ret; } +static void adcx140_pwr_ctrl(struct adcx140_priv *adcx140, bool power_state) +{ + int pwr_ctrl = 0; + + if (power_state) + pwr_ctrl = ADCX140_PWR_CFG_ADC_PDZ | ADCX140_PWR_CFG_PLL_PDZ; + + if (adcx140->micbias_vg && power_state) + pwr_ctrl |= ADCX140_PWR_CFG_BIAS_PDZ; + + regmap_update_bits(adcx140->regmap, ADCX140_PWR_CFG, + ADCX140_PWR_CTRL_MSK, pwr_ctrl); +} + static int adcx140_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; + struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); u8 data = 0; switch (params_width(params)) { @@ -640,9 +655,13 @@ static int adcx140_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + adcx140_pwr_ctrl(adcx140, false); + snd_soc_component_update_bits(component, ADCX140_ASI_CFG0, ADCX140_WORD_LEN_MSK, data); + adcx140_pwr_ctrl(adcx140, true); + return 0; } @@ -709,6 +728,8 @@ static int adcx140_set_dai_fmt(struct snd_soc_dai *codec_dai, adcx140->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + adcx140_pwr_ctrl(adcx140, false); + snd_soc_component_update_bits(component, ADCX140_ASI_CFG0, ADCX140_FSYNCINV_BIT | ADCX140_BCLKINV_BIT | @@ -721,6 +742,7 @@ static int adcx140_set_dai_fmt(struct snd_soc_dai *codec_dai, snd_soc_component_update_bits(component, ADCX140_ASI_CFG1, ADCX140_TX_OFFSET_MASK, offset); + adcx140_pwr_ctrl(adcx140, true); return 0; } @@ -818,12 +840,11 @@ static int adcx140_codec_probe(struct snd_soc_component *component) ret = device_property_read_u32(adcx140->dev, "ti,mic-bias-source", &bias_source); - if (ret) + if (ret || bias_source > ADCX140_MIC_BIAS_VAL_AVDD) { bias_source = ADCX140_MIC_BIAS_VAL_VREF; - - if (bias_source > ADCX140_MIC_BIAS_VAL_AVDD) { - dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); - return -EINVAL; + adcx140->micbias_vg = false; + } else { + adcx140->micbias_vg = true; } ret = device_property_read_u32(adcx140->dev, "ti,vref-source", @@ -906,6 +927,8 @@ static int adcx140_codec_probe(struct snd_soc_component *component) ADCX140_MIC_BIAS_VREF_MSK, bias_cfg); if (ret) dev_err(adcx140->dev, "setting MIC bias failed %d\n", ret); + + adcx140_pwr_ctrl(adcx140, true); out: return ret; } @@ -914,21 +937,19 @@ static int adcx140_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level) { struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); - int pwr_cfg = 0; switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: - pwr_cfg = ADCX140_PWR_CFG_BIAS_PDZ | ADCX140_PWR_CFG_PLL_PDZ | - ADCX140_PWR_CFG_ADC_PDZ; + adcx140_pwr_ctrl(adcx140, true); break; case SND_SOC_BIAS_OFF: - pwr_cfg = 0x0; + adcx140_pwr_ctrl(adcx140, false); break; } - return regmap_write(adcx140->regmap, ADCX140_PWR_CFG, pwr_cfg); + return 0; } static const struct snd_soc_component_driver soc_codec_driver_adcx140 = { diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h index eedbc1d7221f..94c6d1fd2977 100644 --- a/sound/soc/codecs/tlv320adcx140.h +++ b/sound/soc/codecs/tlv320adcx140.h @@ -123,6 +123,7 @@ #define ADCX140_MIC_BIAS_VREF_1375V 2 #define ADCX140_MIC_BIAS_VREF_MSK GENMASK(1, 0) +#define ADCX140_PWR_CTRL_MSK GENMASK(7, 5) #define ADCX140_PWR_CFG_BIAS_PDZ BIT(7) #define ADCX140_PWR_CFG_ADC_PDZ BIT(6) #define ADCX140_PWR_CFG_PLL_PDZ BIT(5) @@ -145,4 +146,5 @@ #define ADCX140_GPO_CFG_MAX 4 #define ADCX140_GPO_DRV_MAX 5 + #endif /* _TLV320ADCX140_ */ -- 2.28.0

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[PATCH] ASoC: qcom: lpass-sc7180: Add MODULE_DEVICE_TABLE
by Douglas Anderson 17 Sep '20

17 Sep '20

The lpass-sc7180 driver can be built as a module but is lacking a MODULE_DEVICE_TABLE. This means it won't auto-load. Fix this oversight. Fixes: 24caf8d9eb10 ("ASoC: qcom: lpass-sc7180: Add platform driver for lpass audio") Signed-off-by: Douglas Anderson <dianders(a)chromium.org> --- sound/soc/qcom/lpass-sc7180.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 167bf2cbc556..a8a3d8f8f567 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -200,6 +200,7 @@ static const struct of_device_id sc7180_lpass_cpu_device_id[] = { {.compatible = "qcom,sc7180-lpass-cpu", .data = &sc7180_data}, {} }; +MODULE_DEVICE_TABLE(of, sc7180_lpass_cpu_device_id); static struct platform_driver sc7180_lpass_cpu_platform_driver = { .driver = { -- 2.28.0.618.gf4bc123cb7-goog

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[PATCH] ASoC: dt-bindings: Correct interrupt flags in examples
by Krzysztof Kozlowski 17 Sep '20

17 Sep '20

GPIO_ACTIVE_x flags are not correct in the context of interrupt flags. These are simple defines so they could be used in DTS but they will not have the same meaning: 1. GPIO_ACTIVE_HIGH = 0 = IRQ_TYPE_NONE 2. GPIO_ACTIVE_LOW = 1 = IRQ_TYPE_EDGE_RISING Correct the interrupt flags, assuming the author of the code wanted some logical behavior behind the name "ACTIVE_xxx", this is: ACTIVE_HIGH => IRQ_TYPE_LEVEL_HIGH Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- Documentation/devicetree/bindings/sound/max98090.txt | 2 +- Documentation/devicetree/bindings/sound/rt5640.txt | 2 +- Documentation/devicetree/bindings/sound/rt5659.txt | 2 +- Documentation/devicetree/bindings/sound/rt5665.txt | 2 +- Documentation/devicetree/bindings/sound/rt5668.txt | 2 +- Documentation/devicetree/bindings/sound/rt5677.txt | 2 +- Documentation/devicetree/bindings/sound/rt5682.txt | 2 +- 7 files changed, 7 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt index 7e1bbd5c27fd..39d640294c62 100644 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -55,5 +55,5 @@ audio-codec@10 { compatible = "maxim,max98090"; reg = <0x10>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(H, 4) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(H, 4) IRQ_TYPE_LEVEL_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index e40e4893eed8..ff1228713f7e 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -88,7 +88,7 @@ rt5640 { compatible = "realtek,rt5640"; reg = <0x1c>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5659.txt b/Documentation/devicetree/bindings/sound/rt5659.txt index 1766e0543fc5..56788f50b6cf 100644 --- a/Documentation/devicetree/bindings/sound/rt5659.txt +++ b/Documentation/devicetree/bindings/sound/rt5659.txt @@ -72,7 +72,7 @@ rt5659 { compatible = "realtek,rt5659"; reg = <0x1b>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5665.txt b/Documentation/devicetree/bindings/sound/rt5665.txt index 8df170506986..f6ca96b4ce98 100644 --- a/Documentation/devicetree/bindings/sound/rt5665.txt +++ b/Documentation/devicetree/bindings/sound/rt5665.txt @@ -62,7 +62,7 @@ rt5659 { compatible = "realtek,rt5665"; reg = <0x1b>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5668.txt b/Documentation/devicetree/bindings/sound/rt5668.txt index c88b96e7764b..a2b7e9a2f2f3 100644 --- a/Documentation/devicetree/bindings/sound/rt5668.txt +++ b/Documentation/devicetree/bindings/sound/rt5668.txt @@ -41,7 +41,7 @@ rt5668 { compatible = "realtek,rt5668b"; reg = <0x1a>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; realtek,dmic1-data-pin = <1>; diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index 1b3c13d206ff..da2430099181 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -64,7 +64,7 @@ rt5677 { compatible = "realtek,rt5677"; reg = <0x2c>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; gpio-controller; #gpio-cells = <2>; diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt index ade1ece8b45f..707fa98d1310 100644 --- a/Documentation/devicetree/bindings/sound/rt5682.txt +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -58,7 +58,7 @@ rt5682 { compatible = "realtek,rt5682i"; reg = <0x1a>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; realtek,dmic1-data-pin = <1>; -- 2.17.1

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[PATCH] ASoC: tlv320adcx140: Fix digital gain range
by Camel Guo 17 Sep '20

17 Sep '20

From: Camel Guo <camelg(a)axis.com> According to its datasheet, the digital gain should be -100 dB when CHx_DVOL is 1 and 27 dB when CHx_DVOL is 255. But with the current dig_vol_tlv, "Digital CHx Out Volume" shows 27.5 dB if CHx_DVOL is 255 and -95.5 dB if CHx_DVOL is 1. This commit fixes this bug. Fixes: 689c7655b50c ("ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family") Signed-off-by: Camel Guo <camelg(a)axis.com> --- sound/soc/codecs/tlv320adcx140.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 6d456aa269ad..13ece039e9fe 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -161,7 +161,7 @@ static const struct regmap_config adcx140_i2c_regmap = { }; /* Digital Volume control. From -100 to 27 dB in 0.5 dB steps */ -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10000, 50, 0); +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10050, 50, 0); /* ADC gain. From 0 to 42 dB in 1 dB steps */ static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 100, 0); -- 2.20.1

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[PATCH v3 1/2] ASoC: dt-bindings: ak4458: Add dsd-path property
by Shengjiu Wang 17 Sep '20

17 Sep '20

Add "dsd-path" property, which is used for ak4497 codec to select the DSD input pin. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/ak4458.txt | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/ak4458.txt b/Documentation/devicetree/bindings/sound/ak4458.txt index 8f6c84f21468..0416c14895d6 100644 --- a/Documentation/devicetree/bindings/sound/ak4458.txt +++ b/Documentation/devicetree/bindings/sound/ak4458.txt @@ -12,6 +12,9 @@ Optional properties: - mute-gpios: A GPIO specifier for the soft mute pin - AVDD-supply: Analog power supply - DVDD-supply: Digital power supply +- dsd-path: Select DSD input pins for ak4497 + 0: select #16, #17, #19 pins + 1: select #3, #4, #5 pins Example: -- 2.27.0

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[PATCH 0/2] ASoC: q6afe: add clocks support
by Srinivas Kandagatla 17 Sep '20

17 Sep '20

q6afe already exposes clocks using apis, but not as proper clock controller driver. This patch puts those clocks in to a proper clock controller so that other drivers that depend on those clocks can be properly expressed. Srinivas Kandagatla (2): ASoC: q6afe: dt-bindings: add q6afe clock bindings ASoC: q6afe-clocks: add q6afe clock controller .../devicetree/bindings/sound/qcom,q6afe.txt | 23 ++ include/dt-bindings/sound/qcom,q6afe.h | 74 ++++- sound/soc/qcom/Kconfig | 4 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-clocks.c | 270 ++++++++++++++++++ 5 files changed, 371 insertions(+), 1 deletion(-) create mode 100644 sound/soc/qcom/qdsp6/q6afe-clocks.c -- 2.21.0

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[PATCH 0/8] ASoC: SOF: small fixes for 5.10
by Kai Vehmanen 17 Sep '20

17 Sep '20

Series that adds debug support for IMX platforms, more details to FW version information, adds missing -EACCESS handling to pm_runtime_get_sync() calls and a set of minor cosmetic, trace verbosity and coding style issues. Guennadi Liakhovetski (3): ASoC: SOF: (cosmetic) remove redundant "ret" variable uses ASoC: SOF: remove several superfluous type-casts ASoC: SOF: fix range checks Iulian Olaru (1): ASoC: SOF: imx: Add debug support for imx platforms Karol Trzcinski (1): ASoC: SOF: Add `src_hash` to `sof_ipc_fw_version` structure Pierre-Louis Bossart (3): ASoC: SOF: debug: update test for pm_runtime_get_sync() ASoC: SOF: control: update test for pm_runtime_get_sync() ASoC: SOF: Intel: hda: reduce verbosity of boot error logs include/sound/sof/info.h | 4 +- sound/soc/sof/control.c | 62 +++++++++++++-------------- sound/soc/sof/debug.c | 2 +- sound/soc/sof/imx/Kconfig | 8 ++++ sound/soc/sof/imx/Makefile | 3 ++ sound/soc/sof/imx/imx-common.c | 72 ++++++++++++++++++++++++++++++++ sound/soc/sof/imx/imx-common.h | 16 +++++++ sound/soc/sof/imx/imx8.c | 23 +++++++++- sound/soc/sof/imx/imx8m.c | 17 +++++++- sound/soc/sof/intel/hda-loader.c | 16 +++---- sound/soc/sof/intel/hda.c | 12 ++++-- sound/soc/sof/intel/hda.h | 2 + sound/soc/sof/sof-audio.c | 6 +-- sound/soc/sof/sof-priv.h | 8 ++++ sound/soc/sof/topology.c | 44 ++++++++++--------- 15 files changed, 226 insertions(+), 69 deletions(-) create mode 100644 sound/soc/sof/imx/imx-common.c create mode 100644 sound/soc/sof/imx/imx-common.h -- 2.27.0

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