March 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH v0] ASoC: rsnd: core: Check convert rate in rsnd_hw_params
by mdurnev＠gmail.com 16 Mar '21

16 Mar '21

From: Mikhail Durnev <mikhail_durnev(a)mentor.com> snd_pcm_hw_params_set_rate_near can return incorrect sample rate in some cases, e.g. when the backend output rate is set to some value higher than 48000 Hz and the input rate is 8000 Hz. So passing the value returned by snd_pcm_hw_params_set_rate_near to snd_pcm_hw_params will result in "FSO/FSI ratio error" and playing no audio at all while the userland is not properly notified about the issue. If SRC is unable to convert the requested sample rate to the sample rate the backend is using, then the requested sample rate should be adjusted in rsnd_hw_params. The userland will be notified about that change in the returned hw_params structure. Signed-off-by: Mikhail Durnev <mikhail_durnev(a)mentor.com> --- sound/soc/sh/rcar/core.c | 69 +++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 68 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index d1612d3..8696a99 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1424,8 +1424,75 @@ static int rsnd_hw_params(struct snd_soc_component *component, } if (io->converted_chan) dev_dbg(dev, "convert channels = %d\n", io->converted_chan); - if (io->converted_rate) + if (io->converted_rate) { + /* + * SRC supports convert rates from params_rate(hw_params)/k_down + * to params_rate(hw_params)*k_up, where k_up is always 6, and + * k_down depends on number of channels and SRC unit. + * So all SRC units can upsample audio up to 6 times regardless + * its number of channels. And all SRC units can downsample + * 2 channel audio up to 6 times too. + */ + int k_up = 6; + int k_down = 6; + int channel; + struct rsnd_mod *src_mod = rsnd_io_to_mod_src(io); + dev_dbg(dev, "convert rate = %d\n", io->converted_rate); + + channel = io->converted_chan ? io->converted_chan : + params_channels(hw_params); + + switch (rsnd_mod_id(src_mod)) { + /* + * SRC0 can downsample 4, 6 and 8 channel audio up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 4 channel audio + * up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 6 and 8 channel audio + * no more than twice. + */ + case 1: + case 3: + case 4: + if (channel > 4) { + k_down = 2; + break; + } + fallthrough; + case 0: + if (channel > 2) + k_down = 4; + break; + + /* Other SRC units do not support more than 2 channels */ + default: + if (channel > 2) + return -EINVAL; + } + + if (params_rate(hw_params) > io->converted_rate * k_down) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + io->converted_rate * k_down; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + io->converted_rate * k_down; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } else if (params_rate(hw_params) * k_up < io->converted_rate) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + (io->converted_rate + k_up - 1) / k_up; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + (io->converted_rate + k_up - 1) / k_up; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } + + /* + * TBD: Max SRC input and output rates also depend on number + * of channels and SRC unit: + * SRC1, SRC3 and SRC4 do not support more than 128kHz + * for 6 channel and 96kHz for 8 channel audio. + * Perhaps this function should return EINVAL if the input or + * the output rate exceeds the limitation. + */ + } } return rsnd_dai_call(hw_params, io, substream, hw_params); -- 2.7.4

3 2

[PATCH v2] ASoC: imx-hdmi: fix platform_no_drv_owner.cocci warnings
by Yang Li 16 Mar '21

16 Mar '21

./sound/soc/fsl/imx-hdmi.c:226:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- Change in v2: -use imx-hdmi instead of hdmi-codec for Subject sound/soc/fsl/imx-hdmi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index dbbb761..cd0235a 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -223,7 +223,6 @@ static int imx_hdmi_probe(struct platform_device *pdev) static struct platform_driver imx_hdmi_driver = { .driver = { .name = "imx-hdmi", - .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, .of_match_table = imx_hdmi_dt_ids, }, -- 1.8.3.1

3 2

[PATCH] ASoC: fsl_spdif: use snd_ctl_boolean_mono_info
by Shengjiu Wang 16 Mar '21

16 Mar '21

From: Viorel Suman <viorel.suman(a)nxp.com> Remove redundant code and use snd_ctl_boolean_mono_info instead. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_spdif.c | 30 +++--------------------------- 1 file changed, 3 insertions(+), 27 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 174e558224d8..1cd3441d1c03 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -801,18 +801,6 @@ static int fsl_spdif_qget(struct snd_kcontrol *kcontrol, return ret; } -/* Valid bit information */ -static int fsl_spdif_vbit_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - - return 0; -} - /* Get valid good bit from interrupt status register */ static int fsl_spdif_rx_vbit_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -925,18 +913,6 @@ static int fsl_spdif_rxrate_get(struct snd_kcontrol *kcontrol, return 0; } -/* User bit sync mode info */ -static int fsl_spdif_usync_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - - return 0; -} - /* * User bit sync mode: * 1 CD User channel subcode @@ -1018,7 +994,7 @@ static struct snd_kcontrol_new fsl_spdif_ctrls[] = { .name = "IEC958 RX V-Bit Errors", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, - .info = fsl_spdif_vbit_info, + .info = snd_ctl_boolean_mono_info, .get = fsl_spdif_rx_vbit_get, }, { @@ -1027,7 +1003,7 @@ static struct snd_kcontrol_new fsl_spdif_ctrls[] = { .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_WRITE | SNDRV_CTL_ELEM_ACCESS_VOLATILE, - .info = fsl_spdif_vbit_info, + .info = snd_ctl_boolean_mono_info, .get = fsl_spdif_tx_vbit_get, .put = fsl_spdif_tx_vbit_put, }, @@ -1047,7 +1023,7 @@ static struct snd_kcontrol_new fsl_spdif_ctrls[] = { .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_WRITE | SNDRV_CTL_ELEM_ACCESS_VOLATILE, - .info = fsl_spdif_usync_info, + .info = snd_ctl_boolean_mono_info, .get = fsl_spdif_usync_get, .put = fsl_spdif_usync_put, }, -- 2.27.0

2 1

[PATCH] ASoC: dt-bindings: fsl_spdif: Add compatible string for new platforms
by Shengjiu Wang 16 Mar '21

16 Mar '21

Add compatible string for new added platforms which support spdif module. They are i.MX8QXP, i.MX8MM, i.MX8MN, i.MX8MQ. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,spdif.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml index 50449b6d1048..4454aca34d56 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -21,6 +21,10 @@ properties: - fsl,vf610-spdif - fsl,imx6sx-spdif - fsl,imx8qm-spdif + - fsl,imx8qxp-spdif + - fsl,imx8mq-spdif + - fsl,imx8mm-spdif + - fsl,imx8mn-spdif reg: maxItems: 1 -- 2.27.0

2 1

[PATCH] ASoC: hdmi-codec: fix platform_no_drv_owner.cocci warnings
by Yang Li 16 Mar '21

16 Mar '21

./sound/soc/fsl/imx-hdmi.c:226:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/fsl/imx-hdmi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index dbbb761..cd0235a 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -223,7 +223,6 @@ static int imx_hdmi_probe(struct platform_device *pdev) static struct platform_driver imx_hdmi_driver = { .driver = { .name = "imx-hdmi", - .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, .of_match_table = imx_hdmi_dt_ids, }, -- 1.8.3.1

3 2

[PATCH v2 0/7] ASoC: codecs: add wcd938x support
by Srinivas Kandagatla 16 Mar '21

16 Mar '21

This patchset adds support for Qualcomm WCD938X codec. Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 4 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 7 x TX diff inputs, 8 DMICs and MBHC. Even though this device has two SoundWire devices, only tx device has access to main codec Control/Status Registers! This patchset along with other SoundWire patches on the list have been tested on SM8250 MTP device. Am planning to send support for MBHC once this driver gets accepted! Thanks, srini Many thanks for reviewing v1. Changes since v1: - updated PA gain to a proper TLV control - move various if else checks to switch case. - return error on check failure rather than fallback to default - cleaned up micbias pull up control - fixed all the gain tlvs correctly - update dt-bindings to use microvolt for button thresholds - removed unnecessary debug at various places. Srinivas Kandagatla (7): ASoC: dt-bindings: wcd938x: add bindings for wcd938x ASoC: codecs: wcd-clsh: add new version support ASoC: codecs: wcd938x: add basic driver ASoC: codecs: wcd938x: add basic controls ASoC: codecs: wcd938x: add playback dapm widgets ASoC: codecs: wcd938x: add capture dapm widgets ASoC: codecs: wcd938x: add audio routing .../bindings/sound/qcom,wcd938x.yaml | 127 + sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-clsh-v2.c | 350 +- sound/soc/codecs/wcd-clsh-v2.h | 16 + sound/soc/codecs/wcd938x-sdw.c | 291 ++ sound/soc/codecs/wcd938x.c | 3623 +++++++++++++++++ sound/soc/codecs/wcd938x.h | 676 +++ 8 files changed, 5084 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml create mode 100644 sound/soc/codecs/wcd938x-sdw.c create mode 100644 sound/soc/codecs/wcd938x.c create mode 100644 sound/soc/codecs/wcd938x.h -- 2.21.0

2 8

conf: No SPDIF on Plantronics 3220 headset
by GitHub pull_request - opened 16 Mar '21

16 Mar '21

alsa-project/alsa-lib pull request #126 was opened from drjhe: The Plantronics Blackwire 3220 Series headset (USB ID 047f:c056) shows up in GNOME Settings with a non-existent digital output. Add it to the list to suppress this. Request URL : https://github.com/alsa-project/alsa-lib/pull/126 Patch URL : https://github.com/alsa-project/alsa-lib/pull/126.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: Intel: sof_rt5682: Add ALC1015Q-VB speaker amp support
by Brent Lu 16 Mar '21

16 Mar '21

This patch adds jsl_rt5682_rt1015p which supports the RT5682 headset codec and ALC1015Q-VB speaker amplifier combination on JasperLake platform. This driver also applies for ALC1015Q-CG running in auto-mode. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_realtek_common.c | 99 +++++++++++++++++++ sound/soc/intel/boards/sof_realtek_common.h | 7 ++ sound/soc/intel/boards/sof_rt5682.c | 19 +++- .../intel/common/soc-acpi-intel-jsl-match.c | 13 +++ 5 files changed, 137 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index d1d28129a32b..58379393b8e4 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -457,6 +457,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_MAX98373_I2C select SND_SOC_RT1011 select SND_SOC_RT1015 + select SND_SOC_RT1015P select SND_SOC_RT5682_I2C select SND_SOC_DMIC select SND_SOC_HDAC_HDMI diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c index f3cf73c620ba..5dd0eb438aa3 100644 --- a/sound/soc/intel/boards/sof_realtek_common.c +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -7,6 +7,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include <sound/soc-dai.h> #include <sound/soc-dapm.h> #include <uapi/sound/asound.h> @@ -136,3 +137,101 @@ void sof_rt1011_codec_conf(struct snd_soc_card *card) card->codec_conf = rt1011_codec_confs; card->num_configs = ARRAY_SIZE(rt1011_codec_confs); } + +/* + * rt1015: i2c mode driver for ALC1015 and ALC1015Q + * rt1015p: auto-mode driver for ALC1015, ALC1015Q, and ALC1015Q-VB + */ +static const struct snd_soc_dapm_route rt1015p_1dev_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Speaker" }, + { "Right Spk", NULL, "Speaker" }, +}; + +static const struct snd_soc_dapm_route rt1015p_2dev_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Left Speaker" }, + { "Right Spk", NULL, "Right Speaker" }, +}; + +static struct snd_soc_codec_conf rt1015p_codec_confs[] = { + { + .dlc = COMP_CODEC_CONF(RT1015P_DEV0_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(RT1015P_DEV1_NAME), + .name_prefix = "Right", + }, +}; + +static struct snd_soc_dai_link_component rt1015p_dai_link_components[] = { + { + .name = RT1015P_DEV0_NAME, + .dai_name = RT1015P_CODEC_DAI, + }, + { + .name = RT1015P_DEV1_NAME, + .dai_name = RT1015P_CODEC_DAI, + }, +}; + +static int rt1015p_get_num_codecs(void) +{ + static int dev_num; + + if (dev_num) + return dev_num; + + if (!acpi_dev_present("RTL1015", "1", -1)) + dev_num = 1; + else + dev_num = 2; + + return dev_num; +} + +static int rt1015p_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + /* reserved for debugging purpose */ + + return 0; +} + +static const struct snd_soc_ops rt1015p_ops = { + .hw_params = rt1015p_hw_params, +}; + +static int rt1015p_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + if (rt1015p_get_num_codecs() == 1) + ret = snd_soc_dapm_add_routes(&card->dapm, rt1015p_1dev_dapm_routes, + ARRAY_SIZE(rt1015p_1dev_dapm_routes)); + else + ret = snd_soc_dapm_add_routes(&card->dapm, rt1015p_2dev_dapm_routes, + ARRAY_SIZE(rt1015p_2dev_dapm_routes)); + if (ret) + dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + return ret; +} + +void sof_rt1015p_dai_link(struct snd_soc_dai_link *link) +{ + link->codecs = rt1015p_dai_link_components; + link->num_codecs = rt1015p_get_num_codecs(); + link->init = rt1015p_init; + link->ops = &rt1015p_ops; +} + +void sof_rt1015p_codec_conf(struct snd_soc_card *card) +{ + if (rt1015p_get_num_codecs() == 1) + return; + + card->codec_conf = rt1015p_codec_confs; + card->num_configs = ARRAY_SIZE(rt1015p_codec_confs); +} diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h index 87cb3812b926..cb0b49b2855c 100644 --- a/sound/soc/intel/boards/sof_realtek_common.h +++ b/sound/soc/intel/boards/sof_realtek_common.h @@ -21,4 +21,11 @@ void sof_rt1011_dai_link(struct snd_soc_dai_link *link); void sof_rt1011_codec_conf(struct snd_soc_card *card); +#define RT1015P_CODEC_DAI "HiFi" +#define RT1015P_DEV0_NAME "RTL1015:00" +#define RT1015P_DEV1_NAME "RTL1015:01" + +void sof_rt1015p_dai_link(struct snd_soc_dai_link *link); +void sof_rt1015p_codec_conf(struct snd_soc_card *card); + #endif /* __SOF_REALTEK_COMMON_H */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 55505e207bc0..f4b898c1719f 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -45,8 +45,9 @@ #define SOF_RT1011_SPEAKER_AMP_PRESENT BIT(13) #define SOF_RT1015_SPEAKER_AMP_PRESENT BIT(14) #define SOF_RT1015_SPEAKER_AMP_100FS BIT(15) -#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(16) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(17) +#define SOF_RT1015P_SPEAKER_AMP_PRESENT BIT(16) +#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(17) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(18) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -723,6 +724,8 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].num_codecs = ARRAY_SIZE(rt1015_components); links[id].init = speaker_codec_init_lr; links[id].ops = &sof_rt1015_ops; + } else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) { + sof_rt1015p_dai_link(&links[id]); } else if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) { links[id].codecs = max_98373_components; @@ -851,6 +854,8 @@ static int sof_audio_probe(struct platform_device *pdev) sof_max98373_codec_conf(&sof_audio_card_rt5682); else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) sof_rt1011_codec_conf(&sof_audio_card_rt5682); + else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) + sof_rt1015p_codec_conf(&sof_audio_card_rt5682); dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, dmic_be_num, hdmi_num); @@ -940,6 +945,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "jsl_rt5682_rt1015p", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1015P_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; @@ -966,3 +980,4 @@ MODULE_ALIAS("platform:tgl_max98373_rt5682"); MODULE_ALIAS("platform:jsl_rt5682_max98360a"); MODULE_ALIAS("platform:cml_rt1015_rt5682"); MODULE_ALIAS("platform:tgl_rt1011_rt5682"); +MODULE_ALIAS("platform:jsl_rt5682_rt1015p"); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index 52238db0bcb5..73fe4f89a82d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -19,6 +19,11 @@ static struct snd_soc_acpi_codecs rt1015_spk = { .codecs = {"10EC1015"} }; +static struct snd_soc_acpi_codecs rt1015p_spk = { + .num_codecs = 1, + .codecs = {"RTL1015"} +}; + static struct snd_soc_acpi_codecs mx98360a_spk = { .num_codecs = 1, .codecs = {"MX98360A"} @@ -52,6 +57,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { .quirk_data = &rt1015_spk, .sof_tplg_filename = "sof-jsl-rt5682-rt1015.tplg", }, + { + .id = "10EC5682", + .drv_name = "jsl_rt5682_rt1015p", + .sof_fw_filename = "sof-jsl.ri", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt1015p_spk, + .sof_tplg_filename = "sof-jsl-rt5682-rt1015.tplg", + }, { .id = "10EC5682", .drv_name = "jsl_rt5682_max98360a", -- 2.17.1

3 2

aaf plugin xrun:1711 error
by GitHub issues - opened 16 Mar '21

16 Mar '21

alsa-project/alsa-plugins issue #19 was opened from elabbing: Hello, I set 2 host to use aaf plugin for TSN test. I followed TSN-doc guidelines and `aaf.txt` doc. Simply, after hosts is ready and syncing though `ptp4l, phc3sys, qdisc,` then I try to run a basic test, no switch between, back to back linked hosts, obtaining the following error. ``` #Talker sudo speaker-test -p 12500 -F S16_BE -c 2 -r 48000 -D aaf0 #Listener sudo arecord -F 25000 -t raw -f S16_BE -c 2 -r 48000 -D aaf0 | \ > aplay -F 25000 -t raw -f S16_BE -c 2 -r 48000 -D default Playing raw data 'stdin' : Signed 16 bit Big Endian, Rate 48000 Hz, Stereo Recording raw data 'stdin' : Signed 16 bit Big Endian, Rate 48000 Hz, Stereo underrun!!! (at least 24,460 ms long) arecord: xrun:1711: read/write error, state = RUNNING ``` This error happens after a random period, tipically less than 60sec. I followed TSN doc settings, but I noticed that value in aaf plugin configuration and cbs qdisc are suggested differrently between aaf.txt doc and TSN doc. Any prompt to get out of this? Many thanks Issue URL : https://github.com/alsa-project/alsa-plugins/issues/19 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs for HP 850 G8
by Jeremy Szu 16 Mar '21

16 Mar '21

The HP EliteBook 850 G8 Notebook PC is using ALC285 codec which is using 0x04 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e34ba64dd0ae..8239e5efc12d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8067,6 +8067,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), + SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), -- 2.30.1

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