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

[PATCH] ASoC: wm_adsp: Support new metadata block ID's
by Charles Keepax 23 Jul '20

23 Jul '20

From: James Schulman <james.schulman(a)cirrus.com> Coefficient files now support additional metadata blocks, these contain machine parsable text strings describing the parameters contained in the coefficient file. Signed-off-by: James Schulman <james.schulman(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 1 + sound/soc/codecs/wmfw.h | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 88c397c700ee9..410cca57da52d 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2621,6 +2621,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) switch (type) { case (WMFW_NAME_TEXT << 8): case (WMFW_INFO_TEXT << 8): + case (WMFW_METADATA << 8): break; case (WMFW_ABSOLUTE << 8): /* diff --git a/sound/soc/codecs/wmfw.h b/sound/soc/codecs/wmfw.h index 4278aa6aeb01b..7423272c30e94 100644 --- a/sound/soc/codecs/wmfw.h +++ b/sound/soc/codecs/wmfw.h @@ -180,6 +180,7 @@ struct wmfw_coeff_item { #define WMFW_ABSOLUTE 0xf0 #define WMFW_ALGORITHM_DATA 0xf2 +#define WMFW_METADATA 0xfc #define WMFW_NAME_TEXT 0xfe #define WMFW_INFO_TEXT 0xff -- 2.11.0

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[PATCH] ASoC: fsl: Replace HTTP links with HTTPS ones
by Alexander A. Klimov 23 Jul '20

23 Jul '20

Rationale: Reduces attack surface on kernel devs opening the links for MITM as HTTPS traffic is much harder to manipulate. Deterministic algorithm: For each file: If not .svg: For each line: If doesn't contain `\bxmlns\b`: For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`: If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`: If both the HTTP and HTTPS versions return 200 OK and serve the same content: Replace HTTP with HTTPS. Signed-off-by: Alexander A. Klimov <grandmaster(a)al2klimov.de> --- Continuing my work started at 93431e0607e5. See also: git log --oneline '--author=Alexander A. Klimov <grandmaster(a)al2klimov.de>' v5.7..master If there are any URLs to be removed completely or at least not (just) HTTPSified: Just clearly say so and I'll *undo my change*. See also: https://lkml.org/lkml/2020/6/27/64 If there are any valid, but yet not changed URLs: See: https://lkml.org/lkml/2020/6/26/837 If you apply the patch, please let me know. sound/soc/fsl/imx-audmix.c | 4 ++-- sound/soc/fsl/imx-audmux.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index e09b45de0efd..96980cb0497f 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -6,8 +6,8 @@ * License. You may obtain a copy of the GNU General Public License * Version 2 or later at the following locations: * - * http://www.opensource.org/licenses/gpl-license.html - * http://www.gnu.org/copyleft/gpl.html + * https://www.opensource.org/licenses/gpl-license.html + * https://www.gnu.org/copyleft/gpl.html */ #include <linux/module.h> diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 3ce85a43e08f..25c18b9e348f 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -5,7 +5,7 @@ // Copyright 2009 Pengutronix, Sascha Hauer <s.hauer(a)pengutronix.de> // // Initial development of this code was funded by -// Phytec Messtechnik GmbH, http://www.phytec.de +// Phytec Messtechnik GmbH, https://www.phytec.de #include <linux/clk.h> #include <linux/debugfs.h> -- 2.27.0

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[PATCH -next] ASoC: ti: fix SND_SOC_J721E_EVM warnings & errors
by Randy Dunlap 23 Jul '20

23 Jul '20

From: Randy Dunlap <rdunlap(a)infradead.org> SND_SOC_J721E_EVM should not select SND_SOC_PCM3168A_I2C when I2C is not enabled. That causes build errors, so make this driver's symbol depend on I2C. WARNING: unmet direct dependencies detected for SND_SOC_PCM3168A_I2C Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && I2C [=n] Selected by [m]: - SND_SOC_J721E_EVM [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && (DMA_OMAP [=y] || TI_EDMA [=m] || TI_K3_UDMA [=n] || COMPILE_TEST [=y]) && (ARCH_K3_J721E_SOC [=n] || COMPILE_TEST [=y]) ../sound/soc/codecs/pcm3168a-i2c.c:59:1: warning: data definition has no type or storage class module_i2c_driver(pcm3168a_i2c_driver); ^~~~~~~~~~~~~~~~~ ../sound/soc/codecs/pcm3168a-i2c.c:59:1: error: type defaults to ‘int’ in declaration of ‘module_i2c_driver’ [-Werror=implicit-int] ../sound/soc/codecs/pcm3168a-i2c.c:59:1: warning: parameter names (without types) in function declaration ../sound/soc/codecs/pcm3168a-i2c.c:49:26: warning: ‘pcm3168a_i2c_driver’ defined but not used [-Wunused-variable] static struct i2c_driver pcm3168a_i2c_driver = { ^~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors Fixes: 6748d0559059 ("ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI)") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/ti/Kconfig | 1 + 1 file changed, 1 insertion(+) --- linux-next-20200720.orig/sound/soc/ti/Kconfig +++ linux-next-20200720/sound/soc/ti/Kconfig @@ -222,6 +222,7 @@ config SND_SOC_DM365_VOICE_CODEC_MODULE config SND_SOC_J721E_EVM tristate "SoC Audio support for j721e EVM" depends on ARCH_K3_J721E_SOC || COMPILE_TEST + depends on I2C select SND_SOC_PCM3168A_I2C select SND_SOC_DAVINCI_MCASP help

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[PATCH 1/4] ASoC: tas2562: Add right and left channel slot programming
by Dan Murphy 23 Jul '20

23 Jul '20

Add programming for the tdm slots for the right and left. This also requires configuring the RX/TX offsets for the DAI format type. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tas2562.c | 71 +++++++++++++++++++++++++++++--------- sound/soc/codecs/tas2562.h | 5 ++- 2 files changed, 59 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index d26e30a2948c..1d3c381aeefe 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -175,7 +175,37 @@ static int tas2562_set_dai_tdm_slot(struct snd_soc_dai *dai, { struct snd_soc_component *component = dai->component; struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); - int ret = 0; + int left_slot, right_slot; + int slots_cfg; + int ret; + + if (!tx_mask) { + dev_err(component->dev, "tx masks must not be 0\n"); + return -EINVAL; + } + + if (slots == 1) { + if (tx_mask != 1) + return -EINVAL; + + left_slot = 0; + right_slot = 0; + } else { + left_slot = __ffs(tx_mask); + tx_mask &= ~(1 << left_slot); + if (tx_mask == 0) { + right_slot = left_slot; + } else { + right_slot = __ffs(tx_mask); + tx_mask &= ~(1 << right_slot); + } + } + + slots_cfg = (right_slot << TAS2562_RIGHT_SLOT_SHIFT) | left_slot; + + ret = snd_soc_component_write(component, TAS2562_TDM_CFG3, slots_cfg); + if (ret < 0) + return ret; switch (slot_width) { case 16: @@ -208,6 +238,18 @@ static int tas2562_set_dai_tdm_slot(struct snd_soc_dai *dai, if (ret < 0) return ret; + ret = snd_soc_component_update_bits(component, TAS2562_TDM_CFG5, + TAS2562_TDM_CFG5_VSNS_SLOT_MASK, + tas2562->v_sense_slot); + if (ret < 0) + return ret; + + ret = snd_soc_component_update_bits(component, TAS2562_TDM_CFG6, + TAS2562_TDM_CFG6_ISNS_SLOT_MASK, + tas2562->i_sense_slot); + if (ret < 0) + return ret; + return 0; } @@ -285,7 +327,8 @@ static int tas2562_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_component *component = dai->component; struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); - u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0; + u8 asi_cfg_1 = 0; + u8 tdm_rx_start_slot = 0; int ret; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -307,27 +350,23 @@ static int tas2562_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) dev_err(tas2562->dev, "Failed to set RX edge\n"); return ret; } - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case (SND_SOC_DAIFMT_I2S): - case (SND_SOC_DAIFMT_DSP_A): - case (SND_SOC_DAIFMT_DSP_B): - tdm_rx_start_slot = BIT(1); - break; - case (SND_SOC_DAIFMT_LEFT_J): + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_DSP_B: tdm_rx_start_slot = 0; break; - default: - dev_err(tas2562->dev, "DAI Format is not found, fmt=0x%x\n", - fmt); - ret = -EINVAL; + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_DSP_A: + tdm_rx_start_slot = 1; break; + default: + dev_err(tas2562->dev, + "DAI Format is not found, fmt=0x%x\n", fmt); + return -EINVAL; } ret = snd_soc_component_update_bits(component, TAS2562_TDM_CFG1, - TAS2562_TDM_CFG1_RX_OFFSET_MASK, - tdm_rx_start_slot); - + TAS2562_RX_OFF_MASK, (tdm_rx_start_slot << 1)); if (ret < 0) return ret; diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 28e75fc431d0..18209f397921 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -34,6 +34,10 @@ #define TAS2562_TDM_DET TAS2562_REG(0, 0x11) #define TAS2562_REV_ID TAS2562_REG(0, 0x7d) +#define TAS2562_RX_OFF_MASK GENMASK(5, 1) +#define TAS2562_TX_OFF_MASK GENMASK(3, 1) +#define TAS2562_RIGHT_SLOT_SHIFT 4 + /* Page 2 */ #define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x0c) #define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x0d) @@ -49,7 +53,6 @@ #define TAS2562_TDM_CFG1_RX_EDGE_MASK BIT(0) #define TAS2562_TDM_CFG1_RX_FALLING 1 -#define TAS2562_TDM_CFG1_RX_OFFSET_MASK GENMASK(4, 0) #define TAS2562_TDM_CFG0_RAMPRATE_MASK BIT(5) #define TAS2562_TDM_CFG0_RAMPRATE_44_1 BIT(5) -- 2.26.2

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[PATCH v8] ASoC: Intel: kbl_rt5663_rt5514_max98927: Fix kabylake_ssp_fixup function
by Harsha Priya 23 Jul '20

23 Jul '20

kabylake_ssp_fixup function uses snd_soc_dpcm to identify the codecs DAIs. The HW parameters are changed based on the codec DAI of the stream. The earlier approach to get snd_soc_dpcm was using container_of() macro on snd_pcm_hw_params. The structures have been modified over time and snd_soc_dpcm does not have snd_pcm_hw_params as a reference but as a copy. This causes the current driver to crash when used. This patch changes the way snd_soc_dpcm is extracted. snd_soc_pcm_runtime holds 2 dpcm instances (one for playback and one for capture). 2 codecs on the SSP are dmic (capture) and speakers (playback). Based on the stream direction, snd_soc_dpcm is extracted from snd_soc_pcm_runtime. Tested for all use cases of the driver. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> Tested-by: Lukasz Majczak <lma(a)semihalf.com> --- v1 -> v2: - Extract dmic from SSP0 as every BE should have own fixup function. v2 -> v3: - Restore naming in the dapm route table to not confuse with other drivers - Fixed indentations v3 -> v4: - Updated code and commit description according to solution proposed by Harsha v4 -> v5: - Cosmetic Changes v5 -> v6: - Dmic regression seen with v4 fixed - Using available routines for obtaining dpcm information v6 -> v7: - Updated comments - initilize rtd_dpcm variable - added break statement in the loop v7 -> v8: - Updated commit message - return on rtd_dpcm being NULL - typo fixed --- --- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 41 +++++++++++++++---- 1 file changed, 32 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 584e4f9cedc2..21808fe13481 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -379,22 +379,45 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL; + + /* + * The following loop will be called only for playback stream + * In this platform, there is only one playback device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { + rtd_dpcm = dpcm; + break; + } + + /* + * This following loop will be called only for capture stream + * In this platform, there is only one capture device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { + rtd_dpcm = dpcm; + break; + } + + if (!rtd_dpcm) + return -EINVAL; + + /* + * The above 2 loops are mutually exclusive based on the stream direction, + * thus rtd_dpcm variable will never be overwritten + */ /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; chan->min = chan->max = 2; snd_mask_none(fmt); snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { + } else if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio DMIC cap")) { if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) chan->min = chan->max = 2; @@ -405,7 +428,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) + if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; -- 2.17.1

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[PATCH 0/4] ASoC: fsl_asrc: allow selecting arbitrary clocks
by Arnaud Ferraris 23 Jul '20

23 Jul '20

The current ASRC driver hardcodes the input and output clocks used for sample rate conversions. In order to allow greater flexibility and to cover more use cases, it would be preferable to select the clocks using device-tree properties. This series also fix register configuration and clock assignment so conversion can be conducted effectively in both directions with a good quality. Arnaud Ferraris (4): dt-bindings: sound: fsl,asrc: add properties to select in/out clocks ASoC: fsl_asrc: allow using arbitrary input and output clocks ASoC: fsl_asrc: always use ratio for conversion ASoC: fsl_asrc: swap input and output clocks in capture mode Documentation/devicetree/bindings/sound/fsl,asrc.txt | 8 ++++++++ sound/soc/fsl/fsl_asrc.c | 69 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++----- sound/soc/fsl/fsl_asrc_common.h | 3 +++ 3 files changed, 75 insertions(+), 5 deletions(-)

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Re: linux-next: Tree for Jul 22 (sound/soc/tegra/snd-soc-tegra210-dmic)
by Randy Dunlap 22 Jul '20

22 Jul '20

On 7/22/20 6:16 AM, Stephen Rothwell wrote: > Hi all, > > Changes since 20200721: > on i386: ERROR: modpost: "__udivdi3" [sound/soc/tegra/snd-soc-tegra210-dmic.ko] undefined! ERROR: modpost: "__divdi3" [sound/soc/tegra/snd-soc-tegra210-dmic.ko] undefined! -- ~Randy Reported-by: Randy Dunlap <rdunlap(a)infradead.org>

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[PATCH] dt-bindings: sound: convert ROHM BD28623 amplifier binding to yaml
by Katsuhiro Suzuki 22 Jul '20

22 Jul '20

This patch converts ROHM BD28623UMV class D speaker amplifier binding to DT schema. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- .../bindings/sound/rohm,bd28623.txt | 29 --------- .../bindings/sound/rohm,bd28623.yaml | 65 +++++++++++++++++++ 2 files changed, 65 insertions(+), 29 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rohm,bd28623.txt create mode 100644 Documentation/devicetree/bindings/sound/rohm,bd28623.yaml diff --git a/Documentation/devicetree/bindings/sound/rohm,bd28623.txt b/Documentation/devicetree/bindings/sound/rohm,bd28623.txt deleted file mode 100644 index d84557c2686e..000000000000 --- a/Documentation/devicetree/bindings/sound/rohm,bd28623.txt +++ /dev/null @@ -1,29 +0,0 @@ -ROHM BD28623MUV Class D speaker amplifier for digital input - -This codec does not have any control buses such as I2C, it detect format and -rate of I2S signal automatically. It has two signals that can be connected -to GPIOs: reset and mute. - -Required properties: -- compatible : should be "rohm,bd28623" -- #sound-dai-cells: should be 0. -- VCCA-supply : regulator phandle for the VCCA supply -- VCCP1-supply : regulator phandle for the VCCP1 supply -- VCCP2-supply : regulator phandle for the VCCP2 supply - -Optional properties: -- reset-gpios : GPIO specifier for the active low reset line -- mute-gpios : GPIO specifier for the active low mute line - -Example: - - codec { - compatible = "rohm,bd28623"; - #sound-dai-cells = <0>; - - VCCA-supply = <&vcc_reg>; - VCCP1-supply = <&vcc_reg>; - VCCP2-supply = <&vcc_reg>; - reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>; - mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>; - }; diff --git a/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml b/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml new file mode 100644 index 000000000000..acd8609252b4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml @@ -0,0 +1,65 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rohm,bd28623.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ROHM BD28623MUV Class D speaker amplifier for digital input + +description: + This codec does not have any control buses such as I2C, it detect + format and rate of I2S signal automatically. It has two signals + that can be connected to GPIOs reset and mute. + +maintainers: + - Katsuhiro Suzuki <katsuhiro(a)katsuster.net> + +properties: + compatible: + const: rohm,bd28623 + + "#sound-dai-cells": + const: 0 + + VCCA-supply: + description: + regulator phandle for the VCCA (for analog) power supply + + VCCP1-supply: + description: + regulator phandle for the VCCP1 (for ch1) power supply + + VCCP2-supply: + description: + regulator phandle for the VCCP2 (for ch2) power supply + + reset-gpios: + maxItems: 1 + description: + GPIO specifier for the active low reset line + + mute-gpios: + maxItems: 1 + description: + GPIO specifier for the active low mute line + +required: + - compatible + - VCCA-supply + - VCCP1-supply + - VCCP2-supply + - "#sound-dai-cells" + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + codec { + compatible = "rohm,bd28623"; + #sound-dai-cells = <0>; + + VCCA-supply = <&vcc_reg>; + VCCP1-supply = <&vcc_reg>; + VCCP2-supply = <&vcc_reg>; + reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>; + mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>; + }; -- 2.27.0

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Several issues with HP OMEN dc-0019-ur
by GitHub issues - edited 22 Jul '20

22 Jul '20

alsa-project/alsa-lib issue #73 was edited from UnicodeMan: Hi. I am using HP OMEN dc0019ur laptop with linux 5.7-xanmod kernel and KDE neon (Ubuntu-based) system. Laptop has combined 3.5mm jack and a seperate stereo microphone jack. Tried various quirks by creating /etc/modprobe.d/mute.conf file and adding line "options snd-hda-intel model=XX", where XX is the name of the quirk from documentation here: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/Doc… With no modprobe changes: mute led does not work, headphones work, but microphone is not detected, and if i plug anything into microphone jack, it switches to microphone in combined jack. With hp-line1-mic1 quirk: mute led works fine, headphones are detected, except microphone, though it works anyway. Seperate microphone jack does not work at all. With headset-mic quirk: mute led does not work, headphones jack works, microphone (from headphones jack) is not detected when present but works when switched to anyway Microphone jack works perfectly (detects a plugged in mic, works in stereo). --- alsa-info.sh logs --- no quirks: http://alsa-project.org/db/?f=a73a83ba0610e3fd88310999b197fdde4372249a hp-line1-mic1: http://alsa-project.org/db/?f=f4d08dec7918c79a391fb1898d50c57a2448d872 headset-mic: https://alsa-project.org/db/?f=4b1eeef821f39eea28071ac2f2c0babcf515ef14 Not sure if it matters, but I want to add that I'm using nvidia render offload with reverse PRIME support, the gpu has one DP and one HDMI port wired directly to it. aaand, I hope I am writing this issue in correct place :D Issue URL : https://github.com/alsa-project/alsa-lib/issues/73 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH v2 0/9] ASoC: qdsp6: add gapless compressed audio support
by Srinivas Kandagatla 22 Jul '20

22 Jul '20

This patchset adds gapless compressed audio support on q6asm. Gapless on q6asm is implemented using 2 streams in a single q6asm session. First few patches such as stream id per each command, gapless flags and silence meta data are for preparedness for adding gapless support. Last patch implements copy callback to allow finer control over buffer offsets, specially in partial drain cases. This patchset is tested on RB3 aka DB845c platform. This patchset as it is will support gapless however QDSP can also support switching decoders on a single stream. Patches to support such feature are send in different patchset which involves adding generic interfaces. Thanks, srini Changes since v1: - Fixed all the comments to use "q6asm" wording correctly. - dropped patches that are already applied Srinivas Kandagatla (9): ASoC: q6asm: rename misleading session id variable ASoC: q6asm: make commands specific to streams ASoC: q6asm: use flags directly from q6asm-dai ASoC: q6asm: add length to write command token ASoC: q6asm: add support to remove intial and trailing silence ASoC: q6asm: add support to gapless flag in q6asm open ASoC: q6asm-dai: add next track metadata support ASoC: qdsp6-dai: add gapless support ASoC: q6asm-dai: add support to copy callback sound/soc/qcom/qdsp6/q6asm-dai.c | 413 +++++++++++++++++++++++-------- sound/soc/qcom/qdsp6/q6asm.c | 169 +++++++++---- sound/soc/qcom/qdsp6/q6asm.h | 48 ++-- 3 files changed, 467 insertions(+), 163 deletions(-) -- 2.21.0

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