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

[PATCH v4 1/2] dt-bindings: tas2562: Convert the tas2562 binding to yaml
by Dan Murphy 28 Jul '20

28 Jul '20

Convert the TAS2562 text file to yaml format. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2562.txt | 34 --------- .../devicetree/bindings/sound/tas2562.yaml | 69 +++++++++++++++++++ 2 files changed, 69 insertions(+), 34 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/tas2562.txt create mode 100644 Documentation/devicetree/bindings/sound/tas2562.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt deleted file mode 100644 index 94796b547184..000000000000 --- a/Documentation/devicetree/bindings/sound/tas2562.txt +++ /dev/null @@ -1,34 +0,0 @@ -Texas Instruments TAS2562 Smart PA - -The TAS2562 is a mono, digital input Class-D audio amplifier optimized for -efficiently driving high peak power into small loudspeakers. -Integrated speaker voltage and current sense provides for -real time monitoring of loudspeaker behavior. - -Required properties: - - #address-cells - Should be <1>. - - #size-cells - Should be <0>. - - 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. - -Optional properties: -- interrupt-parent: phandle to the interrupt controller which provides - the interrupt. -- interrupts: (GPIO) interrupt to which the chip is connected. -- shut-down: GPIO used to control the state of the device. - -Examples: -tas2562@4c { - #address-cells = <1>; - #size-cells = <0>; - compatible = "ti,tas2562"; - reg = <0x4c>; - - interrupt-parent = <&gpio1>; - interrupts = <14>; - - shut-down = <&gpio1 15 0>; - ti,imon-slot-no = <0>; -}; - diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml new file mode 100644 index 000000000000..d2d3401bf251 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -0,0 +1,69 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2562.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2562 Smart PA + +maintainers: + - Dan Murphy <dmurphy(a)ti.com> + +description: | + The TAS2562 is a mono, digital input Class-D audio amplifier optimized for + efficiently driving high peak power into small loudspeakers. + Integrated speaker voltage and current sense provides for + real time monitoring of loudspeaker behavior. + +properties: + compatible: + enum: + - ti,tas2562 + - ti,tas2563 + + reg: + maxItems: 1 + description: | + I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + shut-down-gpio: + description: GPIO used to control the state of the device. + deprecated: true + + shutdown-gpio: + description: GPIO used to control the state of the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tas2562"; + reg = <0x4c>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + shutdown-gpio = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + }; + }; + -- 2.26.2

2 5

[PATCH v2 5/5] ASoC: amd: Added hw_params support for ALC1015
by Ravulapati Vishnu vardhan rao 28 Jul '20

28 Jul '20

Adding rt1015 hw_params which set Bit-clock ratio, PLL and appropriate sys clk specific with RTK1015. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 29 ++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index e0c767716b9a..55815fdaa1aa 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -127,6 +127,34 @@ static int rt5682_clk_enable(struct snd_pcm_substream *substream) return ret; } +static int acp3x_1015_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; + int srate, i, ret; + + ret = 0; + srate = params_rate(params); + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + if (strcmp(codec_dai->component->name, "rt1015-aif")) + continue; + ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64); + if (ret < 0) + return ret; + ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK, + 64 * srate, 256 * srate); + if (ret < 0) + return ret; + ret = snd_soc_dai_set_sysclk(codec_dai, RT1015_SCLK_S_PLL, + 256 * srate, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + } + return ret; +} + static void rt5682_clk_disable(void) { clk_disable_unprepare(rt5682_dai_wclk); @@ -232,6 +260,7 @@ static const struct snd_soc_ops acp3x_5682_ops = { static const struct snd_soc_ops acp3x_max_play_ops = { .startup = acp3x_max_startup, .shutdown = rt5682_shutdown, + .hw_params = acp3x_1015_hw_params, }; static const struct snd_soc_ops acp3x_ec_cap0_ops = { -- 2.17.1

1 0

[PATCH v2 4/5] ASoC: amd: Adding DAI LINK for rt1015 codec
by Ravulapati Vishnu vardhan rao 28 Jul '20

28 Jul '20

DAI link support for RTK 1015 and providing the codec details depending on the snd_soc_card selected by ACPI ID. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 35 ++++++++++++++++++++++++---- 1 file changed, 30 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 015b6a3312f5..e0c767716b9a 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -33,6 +33,12 @@ static struct clk *rt5682_dai_bclk; static struct gpio_desc *dmic_sel; void *soc_is_rltk_max(struct device *dev); +enum { + RT5682 = 0, + MAX, + EC, +}; + static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) { int ret; @@ -242,6 +248,9 @@ SND_SOC_DAILINK_DEF(rt5682, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", "rt5682-aif1"))); SND_SOC_DAILINK_DEF(max, DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", "HiFi"))); +SND_SOC_DAILINK_DEF(rt1015, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC1015:00", "rt1015-aif"), + COMP_CODEC("i2c-10EC1015:01", "rt1015-aif"))); SND_SOC_DAILINK_DEF(cros_ec, DAILINK_COMP_ARRAY(COMP_CODEC("GOOG0013:00", "EC Codec I2S RX"))); @@ -260,7 +269,7 @@ static struct snd_soc_codec_conf rt1015_conf[] = { }; static struct snd_soc_dai_link acp3x_dai[] = { - { + [RT5682] = { .name = "acp3x-5682-play", .stream_name = "Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF @@ -271,16 +280,19 @@ static struct snd_soc_dai_link acp3x_dai[] = { .ops = &acp3x_5682_ops, SND_SOC_DAILINK_REG(acp3x_i2s, rt5682, platform), }, - { + [MAX] = { .name = "acp3x-max98357-play", .stream_name = "HiFi Playback", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBS_CFS, .dpcm_playback = 1, .ops = &acp3x_max_play_ops, - SND_SOC_DAILINK_REG(acp3x_bt, max, platform), + .cpus = acp3x_bt, + .num_cpus = ARRAY_SIZE(acp3x_bt), + .platforms = platform, + .num_platforms = ARRAY_SIZE(platform), }, - { + [EC] = { .name = "acp3x-ec-dmic0-capture", .stream_name = "Capture DMIC0", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF @@ -390,6 +402,18 @@ void *soc_is_rltk_max(struct device *dev) return (void *)match->driver_data; } +static void card_spk_dai_link_present(struct snd_soc_dai_link *links, + const char *card_name) +{ + if (!strcmp(card_name, "acp3xalc56821015")) { + links[1].codecs = rt1015; + links[1].num_codecs = ARRAY_SIZE(rt1015); + } else { + links[1].codecs = max; + links[1].num_codecs = ARRAY_SIZE(max); + } +} + static int acp3x_probe(struct platform_device *pdev) { int ret; @@ -405,6 +429,7 @@ static int acp3x_probe(struct platform_device *pdev) if (!machine) return -ENOMEM; + card_spk_dai_link_present(card->dai_link, card->name); card->dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); -- 2.17.1

1 0

[PATCH v2 3/5] ASoC: amd: Adding support for ALC1015 codec in machine driver
by Ravulapati Vishnu vardhan rao 28 Jul '20

28 Jul '20

Adding support for ALC1015 RTK codec in machine driver. Passing specific card structure based on its ACPI ID. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/Kconfig | 1 + sound/soc/amd/acp3x-rt5682-max9836.c | 63 ++++++++++++++++++++++++++-- 2 files changed, 60 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index e37cf72f2bab..a6ce000fac3f 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -33,6 +33,7 @@ config SND_SOC_AMD_RV_RT5682_MACH select SND_SOC_MAX98357A select SND_SOC_CROS_EC_CODEC select I2C_CROS_EC_TUNNEL + select SND_SOC_RT1015 depends on SND_SOC_AMD_ACP3x && I2C && CROS_EC help This option enables machine driver for RT5682 and MAX9835. diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 2c672657d139..015b6a3312f5 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -21,6 +21,7 @@ #include "raven/acp3x.h" #include "../codecs/rt5682.h" +#include "../codecs/rt1015.h" #define PCO_PLAT_CLK 48000000 #define RT5682_PLL_FREQ (48000 * 512) @@ -247,7 +248,18 @@ SND_SOC_DAILINK_DEF(cros_ec, SND_SOC_DAILINK_DEF(platform, DAILINK_COMP_ARRAY(COMP_PLATFORM("acp3x_rv_i2s_dma.0"))); -static struct snd_soc_dai_link acp3x_dai_5682_98357[] = { +static struct snd_soc_codec_conf rt1015_conf[] = { + { + .dlc = COMP_CODEC_CONF("i2c-10EC1015:00"), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF("i2c-10EC1015:01"), + .name_prefix = "Right", + }, +}; + +static struct snd_soc_dai_link acp3x_dai[] = { { .name = "acp3x-5682-play", .stream_name = "Playback", @@ -317,8 +329,8 @@ static const struct snd_kcontrol_new acp3x_5682_mc_controls[] = { static struct snd_soc_card acp3x_5682 = { .name = "acp3xalc5682m98357", .owner = THIS_MODULE, - .dai_link = acp3x_dai_5682_98357, - .num_links = ARRAY_SIZE(acp3x_dai_5682_98357), + .dai_link = acp3x_dai, + .num_links = ARRAY_SIZE(acp3x_dai), .dapm_widgets = acp3x_5682_widgets, .num_dapm_widgets = ARRAY_SIZE(acp3x_5682_widgets), .dapm_routes = acp3x_5682_audio_route, @@ -327,6 +339,47 @@ static struct snd_soc_card acp3x_5682 = { .num_controls = ARRAY_SIZE(acp3x_5682_mc_controls), }; +static const struct snd_soc_dapm_widget acp3x_1015_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MUX("Dmic Mux", SND_SOC_NOPM, 0, 0, + &acp3x_dmic_mux_control), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + +static const struct snd_soc_dapm_route acp3x_1015_route[] = { + {"Headphone Jack", NULL, "HPOL"}, + {"Headphone Jack", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"Dmic Mux", "Front Mic", "DMIC"}, + {"Dmic Mux", "Rear Mic", "DMIC"}, + {"Left Spk", NULL, "Left SPO"}, + {"Right Spk", NULL, "Right SPO"}, +}; + +static const struct snd_kcontrol_new acp3x_mc_1015_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static struct snd_soc_card acp3x_1015 = { + .name = "acp3xalc56821015", + .owner = THIS_MODULE, + .dai_link = acp3x_dai, + .num_links = ARRAY_SIZE(acp3x_dai), + .dapm_widgets = acp3x_1015_widgets, + .num_dapm_widgets = ARRAY_SIZE(acp3x_1015_widgets), + .dapm_routes = acp3x_1015_route, + .num_dapm_routes = ARRAY_SIZE(acp3x_1015_route), + .codec_conf = rt1015_conf, + .num_configs = ARRAY_SIZE(rt1015_conf), + .controls = acp3x_mc_1015_controls, + .num_controls = ARRAY_SIZE(acp3x_mc_1015_controls), +}; + void *soc_is_rltk_max(struct device *dev) { const struct acpi_device_id *match; @@ -375,6 +428,7 @@ static int acp3x_probe(struct platform_device *pdev) static const struct acpi_device_id acp3x_audio_acpi_match[] = { { "AMDI5682", (unsigned long)&acp3x_5682}, + { "AMDI1015", (unsigned long)&acp3x_1015}, {}, }; MODULE_DEVICE_TABLE(acpi, acp3x_audio_acpi_match); @@ -391,5 +445,6 @@ static struct platform_driver acp3x_audio = { module_platform_driver(acp3x_audio); MODULE_AUTHOR("akshu.agrawal(a)amd.com"); -MODULE_DESCRIPTION("ALC5682 & MAX98357 audio support"); +MODULE_AUTHOR("Vishnuvardhanrao.Ravulapati(a)amd.com"); +MODULE_DESCRIPTION("ALC5682 ALC1015 & MAX98357 audio support"); MODULE_LICENSE("GPL v2"); -- 2.17.1

1 0

[PATCH v2 2/5] ASoC: amd: Passing card structure based on codec
by Ravulapati Vishnu vardhan rao 28 Jul '20

28 Jul '20

Passing specific snd_soc_card structure depending on the ACPI ID. In future we can add other IDs in the ACPI table and pass the structure. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 25 ++++++++++++++++++++----- 1 file changed, 20 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 8b5af064864f..2c672657d139 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -30,6 +30,7 @@ static struct snd_soc_jack pco_jack; static struct clk *rt5682_dai_wclk; static struct clk *rt5682_dai_bclk; static struct gpio_desc *dmic_sel; +void *soc_is_rltk_max(struct device *dev); static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) { @@ -326,18 +327,32 @@ static struct snd_soc_card acp3x_5682 = { .num_controls = ARRAY_SIZE(acp3x_5682_mc_controls), }; +void *soc_is_rltk_max(struct device *dev) +{ + const struct acpi_device_id *match; + + match = acpi_match_device(dev->driver->acpi_match_table, dev); + if (!match) + return NULL; + return (void *)match->driver_data; +} + static int acp3x_probe(struct platform_device *pdev) { int ret; struct snd_soc_card *card; struct acp3x_platform_info *machine; + struct device *dev = &pdev->dev; + + card = (struct snd_soc_card *)soc_is_rltk_max(dev); + if (!card) + return -ENODEV; machine = devm_kzalloc(&pdev->dev, sizeof(*machine), GFP_KERNEL); if (!machine) return -ENOMEM; - card = &acp3x_5682; - acp3x_5682.dev = &pdev->dev; + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); @@ -348,18 +363,18 @@ static int acp3x_probe(struct platform_device *pdev) return PTR_ERR(dmic_sel); } - ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_5682); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", - acp3x_5682.name, ret); + card->name, ret); return ret; } return 0; } static const struct acpi_device_id acp3x_audio_acpi_match[] = { - { "AMDI5682", 0 }, + { "AMDI5682", (unsigned long)&acp3x_5682}, {}, }; MODULE_DEVICE_TABLE(acpi, acp3x_audio_acpi_match); -- 2.17.1

1 0

[PATCH 1/6] ASoC: amd: Renaming snd-soc-card structure and fields.
by Ravulapati Vishnu vardhan rao 28 Jul '20

28 Jul '20

As in future our machine driver supports multiple codecs So changing naming convention of snd_soc_card struct and its fields. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 6009e444b858..8b5af064864f 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -290,7 +290,7 @@ 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[] = { +static const struct snd_soc_dapm_widget acp3x_5682_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Spk", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -298,7 +298,7 @@ static const struct snd_soc_dapm_widget acp3x_widgets[] = { &acp3x_dmic_mux_control), }; -static const struct snd_soc_dapm_route acp3x_audio_route[] = { +static const struct snd_soc_dapm_route acp3x_5682_audio_route[] = { {"Headphone Jack", NULL, "HPOL"}, {"Headphone Jack", NULL, "HPOR"}, {"IN1P", NULL, "Headset Mic"}, @@ -307,23 +307,23 @@ static const struct snd_soc_dapm_route acp3x_audio_route[] = { {"Dmic Mux", "Rear Mic", "DMIC"}, }; -static const struct snd_kcontrol_new acp3x_mc_controls[] = { +static const struct snd_kcontrol_new acp3x_5682_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Spk"), SOC_DAPM_PIN_SWITCH("Headset Mic"), }; -static struct snd_soc_card acp3x_card = { +static struct snd_soc_card acp3x_5682 = { .name = "acp3xalc5682m98357", .owner = THIS_MODULE, .dai_link = acp3x_dai_5682_98357, .num_links = ARRAY_SIZE(acp3x_dai_5682_98357), - .dapm_widgets = acp3x_widgets, - .num_dapm_widgets = ARRAY_SIZE(acp3x_widgets), - .dapm_routes = acp3x_audio_route, - .num_dapm_routes = ARRAY_SIZE(acp3x_audio_route), - .controls = acp3x_mc_controls, - .num_controls = ARRAY_SIZE(acp3x_mc_controls), + .dapm_widgets = acp3x_5682_widgets, + .num_dapm_widgets = ARRAY_SIZE(acp3x_5682_widgets), + .dapm_routes = acp3x_5682_audio_route, + .num_dapm_routes = ARRAY_SIZE(acp3x_5682_audio_route), + .controls = acp3x_5682_mc_controls, + .num_controls = ARRAY_SIZE(acp3x_5682_mc_controls), }; static int acp3x_probe(struct platform_device *pdev) @@ -336,8 +336,8 @@ static int acp3x_probe(struct platform_device *pdev) if (!machine) return -ENOMEM; - card = &acp3x_card; - acp3x_card.dev = &pdev->dev; + card = &acp3x_5682; + acp3x_5682.dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); @@ -348,11 +348,11 @@ static int acp3x_probe(struct platform_device *pdev) return PTR_ERR(dmic_sel); } - ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_card); + ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_5682); if (ret) { dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", - acp3x_card.name, ret); + acp3x_5682.name, ret); return ret; } return 0; -- 2.17.1

6 16

arecord command stop when output file set /dev/null
by GitHub issues - opened 28 Jul '20

28 Jul '20

alsa-project/alsa-utils issue #48 was opened from garyfcg: How to reproduce it? 1. Input the below command, then record go on arecord -r 96000 -c 4 -f S32_LE 1> /dev/null 2. Wait for some minutes, it depended on system memory size. For example,system memory had 2G, (2*1024*1024*1024)/(96k*4*32/8)=1400s The results: 1. this command quit auto. Expect: 1. This command should not be interrupted. Issue URL : https://github.com/alsa-project/alsa-utils/issues/48 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ALSA: hda/hdmi: Fix keep_power assignment for non-component devices
by Takashi Iwai 28 Jul '20

28 Jul '20

It's been reported that, when neither nouveau nor Nvidia graphics driver is used, the screen starts flickering. And, after comparing between the working case (stable 4.4.x) and the broken case, it turned out that the problem comes from the audio component binding. The Nvidia and AMD audio binding code clears the bus->keep_power flag whenever snd_hdac_acomp_init() succeeds. But this doesn't mean that the component is actually bound, but it merely indicates that it's ready for binding. So, when both nouveau and Nvidia are blacklisted or not ready, the driver keeps running without the audio component but also with bus->keep_power = false. This made the driver runtime PM kicked in and powering down when unused, which results in flickering in the graphics side, as it seems. For fixing the bug, this patch moves the bus->keep_power flag change into generic_acomp_notifier_set() that is the function called from the master_bind callback of component ops; i.e. it's guaranteed that the binding succeeded. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208609 Fixes: 5a858e79c911 ("ALSA: hda - Disable audio component for legacy Nvidia HDMI codecs") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 41eaa89660c3..cd46247988e4 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2440,6 +2440,7 @@ static void generic_acomp_notifier_set(struct drm_audio_component *acomp, mutex_lock(&spec->bind_lock); spec->use_acomp_notifier = use_acomp; spec->codec->relaxed_resume = use_acomp; + spec->codec->bus->keep_power = 0; /* reprogram each jack detection logic depending on the notifier */ for (i = 0; i < spec->num_pins; i++) reprogram_jack_detect(spec->codec, @@ -2534,7 +2535,6 @@ static void generic_acomp_init(struct hda_codec *codec, if (!snd_hdac_acomp_init(&codec->bus->core, &spec->drm_audio_ops, match_bound_vga, 0)) { spec->acomp_registered = true; - codec->bus->keep_power = 0; } } -- 2.16.4

1 0

[PATCH 3/3] ASoC: max98390: update dsm param bin max size
by Steve Lee 28 Jul '20

28 Jul '20

MAX98390_DSM_PARAM_MAX_SIZE is changed to support extended register update. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98390.h b/sound/soc/codecs/max98390.h index 5f444e7779b0..dff884f68e3e 100644 --- a/sound/soc/codecs/max98390.h +++ b/sound/soc/codecs/max98390.h @@ -650,7 +650,7 @@ /* DSM register offset */ #define MAX98390_DSM_PAYLOAD_OFFSET 16 -#define MAX98390_DSM_PARAM_MAX_SIZE 770 +#define MAX98390_DSM_PARAM_MAX_SIZE 1024 #define MAX98390_DSM_PARAM_MIN_SIZE 670 struct max98390_priv { -- 2.17.1

2 4

USB interface: Mixer controls out-of-sync?
by Tanjeff Moos 27 Jul '20

27 Jul '20

Hi all, after connecting my USB interface (Focusrite Scarlett 6i6 1st Gen), the alsa control element values reported by libalsa are not in sync with the hardware. For example, the "Master 1 (Monitor)" control is shown as muted, but a signal can be heard. Unmuting changes nothing, and muting again disables the signal. In practice I often have to go through many controls and alter them a little to synchronize alsamixer with the actual hardware settings. Is this a known behavior? Kind regards, Tanjeff

1 0