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

[PATCH 0/3] ASoC: fsl_easrc: Fix several warnings
by Shengjiu Wang 16 Jun '20

16 Jun '20

Fix several warnings with "make W=1" Shengjiu Wang (3): ASoC: fsl_easrc: Fix -Wmissing-prototypes warning ASoC: fsl_easrc: Fix -Wunused-but-set-variable ASoC: fsl_easrc: Fix "Function parameter not described" warnings sound/soc/fsl/fsl_easrc.c | 42 +++++++++++++++++---------------------- 1 file changed, 18 insertions(+), 24 deletions(-) -- 2.21.0

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[PATCH v2 0/2] ASoC: mediatek: mt6358: support DMIC one-wire mode
by Jiaxin Yu 16 Jun '20

16 Jun '20

v2 changes: 1. Uses a DT property to select DMIC mode instead of a mixer control. v1 changes: 1. Uses a mixer control to select DMIC mode. 2. patchwork list: https://patchwork.kernel.org/patch/11578309 Jiaxin Yu (2): ASoC: mediatek: mt6358: support DMIC one-wire mode ASoC: dt-bindings: mediatek: mt6358: add dmic-mode property Documentation/devicetree/bindings/sound/mt6358.txt | 6 ++++++ sound/soc/codecs/mt6358.c | 23 +++++++++++++++++++++- 2 files changed, 28 insertions(+), 1 deletion(-) -- 1.8.1.1.dirty

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[PATCH] ASoC: img-parallel-out: Fix a reference count leak
by wu000273＠umn.edu 16 Jun '20

16 Jun '20

From: Qiushi Wu <wu000273(a)umn.edu> pm_runtime_get_sync() increments the runtime PM usage counter even when it returns an error code, causing incorrect ref count if pm_runtime_put_noidle() is not called in error handling paths. Thus call pm_runtime_put_noidle() if pm_runtime_get_sync() fails. Signed-off-by: Qiushi Wu <wu000273(a)umn.edu> --- sound/soc/img/img-parallel-out.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/img/img-parallel-out.c b/sound/soc/img/img-parallel-out.c index 5ddbe3a31c2e..4da49a42e854 100644 --- a/sound/soc/img/img-parallel-out.c +++ b/sound/soc/img/img-parallel-out.c @@ -163,8 +163,10 @@ static int img_prl_out_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } ret = pm_runtime_get_sync(prl->dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put_noidle(prl->dev); return ret; + } reg = img_prl_out_readl(prl, IMG_PRL_OUT_CTL); reg = (reg & ~IMG_PRL_OUT_CTL_EDGE_MASK) | control_set; -- 2.17.1

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[V7 PATCH] dt-bindings: Added device tree binding for max98390
by Steve Lee 16 Jun '20

16 Jun '20

Add DT binding of max98390 amplifier driver. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- Changed since V6: * Re-confirm yaml dt binding check * Add minimum and maximum value for each temperature_calib and r0_calib * Add maxim prefix for naming. Changed since V5: * Change txt to yaml and fix up the examples. Changed since V4: * No changes. Changed since V3: * No changes. Changed since V2: * No changes. Changed since V1: * Modified sample text in example .../bindings/sound/maxim,max98390.yaml | 51 +++++++++++++++++++ 1 file changed, 51 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98390.yaml diff --git a/Documentation/devicetree/bindings/sound/maxim,max98390.yaml b/Documentation/devicetree/bindings/sound/maxim,max98390.yaml new file mode 100644 index 000000000000..e5ac35280da3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98390.yaml @@ -0,0 +1,51 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98390.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98390 Speaker Amplifier with Integrated Dynamic Speaker Management + +maintainers: + - Steve Lee <steves.lee(a)maximintegrated.com> + +properties: + compatible: + const: maxim,max98390 + + reg: + maxItems: 1 + description: I2C address of the device. + + maxim,temperature_calib: + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + description: The calculated temperature data was measured while doing the calibration. + minimum: 0 + maximum: 65535 + + maxim,r0_calib: + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + description: This is r0 calibration data which was measured in factory mode. + minimum: 1 + maximum: 8388607 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98390: amplifier@38 { + compatible = "maxim,max98390"; + reg = <0x38>; + maxim,temperature_calib = <1024>; + maxim,r0_calib = <100232>; + }; + }; -- 2.17.1

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[PATCH 1/2] ASoC: bindings: wm8960: Add property for headphone detection
by Shengjiu Wang 16 Jun '20

16 Jun '20

Add two properties for headphone detection. wlf,hp-cfg: A list of headphone jack detect configuration register values wlf,gpio-cfg: A list of GPIO configuration register values Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/wm8960.txt | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/wm8960.txt b/Documentation/devicetree/bindings/sound/wm8960.txt index 6d29ac3750ee..85d3b287108c 100644 --- a/Documentation/devicetree/bindings/sound/wm8960.txt +++ b/Documentation/devicetree/bindings/sound/wm8960.txt @@ -21,6 +21,17 @@ Optional properties: enabled and disabled together with HP_L and HP_R pins in response to jack detect events. + - wlf,hp-cfg: A list of headphone jack detect configuration register values. + The list must be 3 entries long. + hp-cfg[0]: HPSEL[1:0] of R48 (Additional Control 4). + hp-cfg[1]: {HPSWEN:HPSWPOL} of R24 (Additional Control 2). + hp-cfg[2]: {TOCLKSEL:TOEN} of R23 (Additional Control 1). + + - wlf,gpio-cfg: A list of GPIO configuration register values. + The list must be 2 entries long. + gpio-cfg[0]: ALRCGPIO of R9 (Audio interface) + gpio-cfg[1]: {GPIOPOL:GPIOSEL[2:0]} of R48 (Additional Control 4). + Example: wm8960: codec@1a { -- 2.21.0

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[PATCH 0/2] ASoC: max98357a: support MAX98360A in OF
by Tzung-Bi Shih 16 Jun '20

16 Jun '20

Commit 1a0f2433d738 ("ASoC: max98357a: Add ACPI HID MAX98360A") supports MAX98360A in ACPI world. This series supports MAX98360A in OF world. Tzung-Bi Shih (2): ASoC: max98357a: add compatible string for MAX98360A ASoC: dt-bindings: add compatible string for MAX98360A .../devicetree/bindings/sound/max98357a.txt | 12 +++++++++--- sound/soc/codecs/max98357a.c | 1 + 2 files changed, 10 insertions(+), 3 deletions(-) -- 2.27.0.278.ge193c7cf3a9-goog

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[PATCH v3 0/3] ASoC: Intel: Add KeemBay ASoC platform driver
by Sia Jee Heng 16 Jun '20

16 Jun '20

The below series of patches support the KeemBay ASoC platform driver. The platform driver initialize the i2s to capture and playback the pcm data on the ARM. The i2s is running in polling mode. There is no DSP in the KeemBay SoC. Users are rely on the Gstreamer plugin to perform Audio preprocessing. Audio graph card is used to connect the platform driver with the tlv320aic3204 codec. Change History: v3: - Adjusted header format. - Use Audio graph card instead of custom sound card. - Use if-else instead of conditional operator. - Enabled .set_fmt to configure master clock. v2: - Corrected I2S naming for DT binding. v1: - Initial version. Sia Jee Heng (3): ASoC: Intel: Add KeemBay platform driver ASoC: Intel: Add makefiles and kconfig changes for KeemBay dt-bindings: sound: Add documentation for KeemBay i2s .../bindings/sound/intel,keembay-i2s.yaml | 68 +++ sound/soc/intel/Kconfig | 7 + sound/soc/intel/Makefile | 1 + sound/soc/intel/keembay/Makefile | 4 + sound/soc/intel/keembay/kmb_platform.c | 667 +++++++++++++++++++++ sound/soc/intel/keembay/kmb_platform.h | 145 +++++ 6 files changed, 892 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml create mode 100644 sound/soc/intel/keembay/Makefile create mode 100644 sound/soc/intel/keembay/kmb_platform.c create mode 100644 sound/soc/intel/keembay/kmb_platform.h -- 1.9.1

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[PATCH 1/5] ASoC: rt5682: Enable Vref2 under using PLL2
by derek.fang＠realtek.com 16 Jun '20

16 Jun '20

From: "derek.fang" <derek.fang(a)realtek.com> Enable Vref2 under long term using PLL2 to avoid clock unstable. Signed-off-by: derek.fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682.c | 24 +++++++++++++++++------- 1 file changed, 17 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index d324512..ca8e1f2 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -960,13 +960,12 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); - if (snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) + if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); - else + RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0); + if (!snd_soc_dapm_get_pin_status(dapm, "Vref2")) snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, - RT5682_PWR_VREF2 | RT5682_PWR_MB, 0); + RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, RT5682_PWR_CBJ, 0); @@ -1601,8 +1600,7 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0, rt5682_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, - NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref2", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0), /* ASRC */ @@ -2485,6 +2483,15 @@ static int rt5682_wclk_prepare(struct clk_hw *hw) snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS"); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_MB, RT5682_PWR_MB); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Vref2"); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2 | RT5682_PWR_FV2, + RT5682_PWR_VREF2); + usleep_range(55000, 60000); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_FV2, RT5682_PWR_FV2); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1"); snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2F"); snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2B"); @@ -2510,9 +2517,12 @@ static void rt5682_wclk_unprepare(struct clk_hw *hw) snd_soc_dapm_mutex_lock(dapm); snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Vref2"); if (!rt5682->jack_type) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2 | RT5682_PWR_FV2 | RT5682_PWR_MB, 0); + snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1"); snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2F"); snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2B"); -- 2.7.4

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[PATCH v5 0/3] ASoC: Intel: Add KeemBay ASoC platform driver
by Sia Jee Heng 16 Jun '20

16 Jun '20

The below series of patches support the KeemBay ASoC platform driver. The platform driver initialize the i2s to capture and playback the pcm data on the ARM. The i2s is running in polling mode. There is no DSP in the KeemBay SoC. Users are rely on the Gstreamer plugin to perform Audio preprocessing. Audio graph card is used to connect the platform driver with the tlv320aic3204 codec. Change History: v5: - Remove OF dependency from Kconfig as OF is shifted to audio graph card. v4: - Reduce if-otology at the tx/rx function. - Fix indentation. - specify .rate directly v3: - Adjusted header format. - Use Audio graph card instead of custom sound card. - Use if-else instead of conditional operator. - Enabled .set_fmt to configure master clock. v2: - Corrected I2S naming for DT binding. v1: - Initial version. Sia Jee Heng (3): ASoC: Intel: Add KeemBay platform driver ASoC: Intel: Add makefiles and kconfig changes for KeemBay dt-bindings: sound: Add documentation for KeemBay i2s .../bindings/sound/intel,keembay-i2s.yaml | 68 +++ sound/soc/intel/Kconfig | 7 + sound/soc/intel/Makefile | 1 + sound/soc/intel/keembay/Makefile | 4 + sound/soc/intel/keembay/kmb_platform.c | 654 +++++++++++++++++++++ sound/soc/intel/keembay/kmb_platform.h | 145 +++++ 6 files changed, 879 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml create mode 100644 sound/soc/intel/keembay/Makefile create mode 100644 sound/soc/intel/keembay/kmb_platform.c create mode 100644 sound/soc/intel/keembay/kmb_platform.h -- 1.9.1

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[v3] ASoC: AMD: Use mixer control to switch between DMICs
by Akshu Agrawal 16 Jun '20

16 Jun '20

Having mixer control to switch between DMICs prevents user to initiate capture simultaneously on both the DMIcs. Earlier 2 separate devices, one for each DMIC, gave an option of using them simultaneously, which is not supported. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: Modified "Front Mic" to "DMIC Switch" v3: Changed to using of Mux sound/soc/amd/acp3x-rt5682-max9836.c | 58 +++++++++++++++------------- 1 file changed, 31 insertions(+), 27 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index e499c00e0c66..f745b42dfd23 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -188,25 +188,27 @@ static int acp3x_ec_dmic0_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_BT_INSTANCE; snd_soc_dai_set_bclk_ratio(codec_dai, 64); - if (dmic_sel) - gpiod_set_value(dmic_sel, 0); return rt5682_clk_enable(substream); } -static int acp3x_ec_dmic1_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); +static int dmic_switch; - machine->cap_i2s_instance = I2S_BT_INSTANCE; - snd_soc_dai_set_bclk_ratio(codec_dai, 64); - if (dmic_sel) - gpiod_set_value(dmic_sel, 1); +static int dmic_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = dmic_switch; + return 0; +} - return rt5682_clk_enable(substream); +static int dmic_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (dmic_sel) { + dmic_switch = ucontrol->value.integer.value[0]; + gpiod_set_value(dmic_sel, dmic_switch); + } + return 0; } static void rt5682_shutdown(struct snd_pcm_substream *substream) @@ -229,11 +231,6 @@ static const struct snd_soc_ops acp3x_ec_cap0_ops = { .shutdown = rt5682_shutdown, }; -static const struct snd_soc_ops acp3x_ec_cap1_ops = { - .startup = acp3x_ec_dmic1_startup, - .shutdown = rt5682_shutdown, -}; - SND_SOC_DAILINK_DEF(acp3x_i2s, DAILINK_COMP_ARRAY(COMP_CPU("acp3x_i2s_playcap.0"))); SND_SOC_DAILINK_DEF(acp3x_bt, @@ -279,21 +276,26 @@ static struct snd_soc_dai_link acp3x_dai_5682_98357[] = { .ops = &acp3x_ec_cap0_ops, SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), }, - { - .name = "acp3x-ec-dmic1-capture", - .stream_name = "Capture DMIC1", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS, - .dpcm_capture = 1, - .ops = &acp3x_ec_cap1_ops, - SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), - }, }; +static const char * const dmic_mux_text[] = { + "Front Mic", + "Rear Mic", +}; + +static SOC_ENUM_SINGLE_DECL( + acp3x_dmic_enum, SND_SOC_NOPM, 0, dmic_mux_text); + +static const struct snd_kcontrol_new acp3x_dmic_mux_control = + SOC_DAPM_ENUM_EXT("DMIC Select Mux", acp3x_dmic_enum, + dmic_get, dmic_set); + static const struct snd_soc_dapm_widget acp3x_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Spk", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MUX("Dmic Mux", SND_SOC_NOPM, 0, 0, + &acp3x_dmic_mux_control), }; static const struct snd_soc_dapm_route acp3x_audio_route[] = { @@ -301,6 +303,8 @@ static const struct snd_soc_dapm_route acp3x_audio_route[] = { {"Headphone Jack", NULL, "HPOR"}, {"IN1P", NULL, "Headset Mic"}, {"Spk", NULL, "Speaker"}, + {"Dmic Mux", "Front Mic", "DMIC"}, + {"Dmic Mux", "Rear Mic", "DMIC"}, }; static const struct snd_kcontrol_new acp3x_mc_controls[] = { -- 2.20.1

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