October 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH AUTOSEL 6.1 01/19] ASoC: Intel: sof_sdw: add support for SKU 0B14
by Sasha Levin 18 Oct '23

18 Oct '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit fb0b8d299781be8d46b3612aa96cef28da0d93f4 ] One more missing SKU in the list. Closes: https://github.com/thesofproject/linux/issues/4543 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Chao Song <chao.song(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20230919092125.1922468-1-yung-chuan.liao@linux.in… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 414ac90273810..985012f2003e2 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -347,6 +347,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { /* No Jack */ .driver_data = (void *)SOF_SDW_TGL_HDMI, }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B14"), + }, + /* No Jack */ + .driver_data = (void *)SOF_SDW_TGL_HDMI, + }, + { .callback = sof_sdw_quirk_cb, .matches = { -- 2.40.1

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[PATCH AUTOSEL 6.5 22/31] ALSA: usb-audio: Fix microphone sound on Nexigo webcam.
by Sasha Levin 18 Oct '23

18 Oct '23

From: Christos Skevis <xristos.thes(a)gmail.com> [ Upstream commit 4a63e68a295187ae3c1cb3fa0c583c96a959714f ] I own an external usb Webcam, model NexiGo N930AF, which had low mic volume and inconsistent sound quality. Video works as expected. (snip) [ +0.047857] usb 5-1: new high-speed USB device number 2 using xhci_hcd [ +0.003406] usb 5-1: New USB device found, idVendor=1bcf, idProduct=2283, bcdDevice=12.17 [ +0.000007] usb 5-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ +0.000004] usb 5-1: Product: NexiGo N930AF FHD Webcam [ +0.000003] usb 5-1: Manufacturer: SHENZHEN AONI ELECTRONIC CO., LTD [ +0.000004] usb 5-1: SerialNumber: 20201217011 [ +0.003900] usb 5-1: Found UVC 1.00 device NexiGo N930AF FHD Webcam (1bcf:2283) [ +0.025726] usb 5-1: 3:1: cannot get usb sound sample rate freq at ep 0x86 [ +0.071482] usb 5-1: 3:2: cannot get usb sound sample rate freq at ep 0x86 [ +0.004679] usb 5-1: 3:3: cannot get usb sound sample rate freq at ep 0x86 [ +0.051607] usb 5-1: Warning! Unlikely big volume range (=4096), cval->res is probably wrong. [ +0.000005] usb 5-1: [7] FU [Mic Capture Volume] ch = 1, val = 0/4096/1 Set up quirk cval->res to 16 for 256 levels, Set GET_SAMPLE_RATE quirk flag to stop trying to get the sample rate. Confirmed that happened anyway later due to the backoff mechanism, after 3 failures All audio stream on device interfaces share the same values, apart from wMaxPacketSize and tSamFreq : (snip) Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 8 bDelay 1 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 1 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 44100 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x005c 1x 92 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0x0000 (snip) Based on the usb data about manufacturer, SPCA2281B3 is the most likely controller IC Manufacturer does not provide link for datasheet nor detailed specs. No way to confirm if the firmware supports any other way of getting the sample rate. Testing patch provides consistent good sound recording quality and volume range. (snip) [ +0.045764] usb 5-1: new high-speed USB device number 2 using xhci_hcd [ +0.106290] usb 5-1: New USB device found, idVendor=1bcf, idProduct=2283, bcdDevice=12.17 [ +0.000006] usb 5-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ +0.000004] usb 5-1: Product: NexiGo N930AF FHD Webcam [ +0.000003] usb 5-1: Manufacturer: SHENZHEN AONI ELECTRONIC CO., LTD [ +0.000004] usb 5-1: SerialNumber: 20201217011 [ +0.043700] usb 5-1: set resolution quirk: cval->res = 16 [ +0.002585] usb 5-1: Found UVC 1.00 device NexiGo N930AF FHD Webcam (1bcf:2283) Signed-off-by: Christos Skevis <xristos.thes(a)gmail.com> Link: https://lore.kernel.org/r/20231006155330.399393-1-xristos.thes@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer.c | 7 +++++++ sound/usb/quirks.c | 2 ++ 2 files changed, 9 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 9105ec623120a..783a2493707ea 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1204,6 +1204,13 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->res = 16; } break; + case USB_ID(0x1bcf, 0x2283): /* NexiGo N930AF FHD Webcam */ + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + usb_audio_info(chip, + "set resolution quirk: cval->res = 16\n"); + cval->res = 16; + } + break; } } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d4bbef70d2f7b..4e64842245e19 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2177,6 +2177,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_FIXED_RATE), DEVICE_FLG(0x0ecb, 0x2069, /* JBL Quantum810 Wireless */ QUIRK_FLAG_FIXED_RATE), + DEVICE_FLG(0x1bcf, 0x2283, /* NexiGo N930AF FHD Webcam */ + QUIRK_FLAG_GET_SAMPLE_RATE), /* Vendor matches */ VENDOR_FLG(0x045e, /* MS Lifecam */ -- 2.40.1

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[PATCH AUTOSEL 6.5 13/31] ALSA: usb-audio: Fix microphone sound on Opencomm2 Headset
by Sasha Levin 18 Oct '23

18 Oct '23

From: WhaleChang <whalechang(a)google.com> [ Upstream commit 6a83d6f3bb3c329a73e3483651fb77b78bac1878 ] When a Opencomm2 Headset is connected to a Bluetooth USB dongle, the audio playback functions properly, but the microphone does not work. In the dmesg logs, there are messages indicating that the init_pitch function fails when the capture process begins. The microphone only functions when the ep pitch control is not set. Toggling the pitch control off bypasses the init_piatch function and allows the microphone to work. Signed-off-by: WhaleChang <whalechang(a)google.com> Link: https://lore.kernel.org/r/20231006044852.4181022-1-whalechang@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 598659d761cc9..d4bbef70d2f7b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1994,7 +1994,11 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, /* mic works only when ep packet size is set to wMaxPacketSize */ fp->attributes |= UAC_EP_CS_ATTR_FILL_MAX; break; - + case USB_ID(0x3511, 0x2b1e): /* Opencomm2 UC USB Bluetooth dongle */ + /* mic works only when ep pitch control is not set */ + if (stream == SNDRV_PCM_STREAM_CAPTURE) + fp->attributes &= ~UAC_EP_CS_ATTR_PITCH_CONTROL; + break; } } -- 2.40.1

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[PATCH AUTOSEL 6.5 09/31] ASoC: tlv320adc3xxx: BUG: Correct micbias setting
by Sasha Levin 18 Oct '23

18 Oct '23

From: Antoine Gennart <gennartan(a)disroot.org> [ Upstream commit e930bea4124b8a4a47ba4092d99da30099b9242d ] The micbias setting for tlv320adc can also have the value '3' which means that the micbias ouput pin is connected to the input pin AVDD. Signed-off-by: Antoine Gennart <gennartan(a)disroot.org> Link: https://lore.kernel.org/r/20230929130117.77661-1-gennartan@disroot.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tlv320adc3xxx.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c index b976c1946286a..420bbf588efea 100644 --- a/sound/soc/codecs/tlv320adc3xxx.c +++ b/sound/soc/codecs/tlv320adc3xxx.c @@ -293,7 +293,7 @@ #define ADC3XXX_BYPASS_RPGA 0x80 /* MICBIAS control bits */ -#define ADC3XXX_MICBIAS_MASK 0x2 +#define ADC3XXX_MICBIAS_MASK 0x3 #define ADC3XXX_MICBIAS1_SHIFT 5 #define ADC3XXX_MICBIAS2_SHIFT 3 @@ -1099,7 +1099,7 @@ static int adc3xxx_parse_dt_micbias(struct adc3xxx *adc3xxx, unsigned int val; if (!of_property_read_u32(np, propname, &val)) { - if (val >= ADC3XXX_MICBIAS_AVDD) { + if (val > ADC3XXX_MICBIAS_AVDD) { dev_err(dev, "Invalid property value for '%s'\n", propname); return -EINVAL; } -- 2.40.1

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[PATCH AUTOSEL 6.5 08/31] ASoC: core: Do not call link_exit() on uninitialized rtd objects
by Sasha Levin 18 Oct '23

18 Oct '23

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit dd9f9cc1e6b9391140afa5cf27bb47c9e2a08d02 ] On init we have sequence: for_each_card_prelinks(card, i, dai_link) { ret = snd_soc_add_pcm_runtime(card, dai_link); ret = init_some_other_things(...); if (ret) goto probe_end: for_each_card_rtds(card, rtd) { ret = soc_init_pcm_runtime(card, rtd); probe_end: while on exit: for_each_card_rtds(card, rtd) snd_soc_link_exit(rtd); If init_some_other_things() step fails due to error we end up with not fully setup rtds and try to call snd_soc_link_exit on them, which depending on contents on .link_exit handler, can end up dereferencing NULL pointer. Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Link: https://lore.kernel.org/r/20230929103243.705433-2-amadeuszx.slawinski@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 20 +++++++++++++++----- 2 files changed, 17 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index b27f84580c5b0..cf34810882347 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1125,6 +1125,8 @@ struct snd_soc_pcm_runtime { unsigned int pop_wait:1; unsigned int fe_compr:1; /* for Dynamic PCM */ + bool initialized; + int num_components; struct snd_soc_component *components[]; /* CPU/Codec/Platform */ }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1a0bde23f5e6f..2d85164457f73 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1259,7 +1259,7 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, snd_soc_runtime_get_dai_fmt(rtd); ret = snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt); if (ret) - return ret; + goto err; /* add DPCM sysfs entries */ soc_dpcm_debugfs_add(rtd); @@ -1284,17 +1284,26 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, /* create compress_device if possible */ ret = snd_soc_dai_compress_new(cpu_dai, rtd, num); if (ret != -ENOTSUPP) - return ret; + goto err; /* create the pcm */ ret = soc_new_pcm(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", dai_link->stream_name, ret); - return ret; + goto err; } - return snd_soc_pcm_dai_new(rtd); + ret = snd_soc_pcm_dai_new(rtd); + if (ret < 0) + goto err; + + rtd->initialized = true; + + return 0; +err: + snd_soc_link_exit(rtd); + return ret; } static void soc_set_name_prefix(struct snd_soc_card *card, @@ -1892,7 +1901,8 @@ static void soc_cleanup_card_resources(struct snd_soc_card *card) /* release machine specific resources */ for_each_card_rtds(card, rtd) - snd_soc_link_exit(rtd); + if (rtd->initialized) + snd_soc_link_exit(rtd); /* remove and free each DAI */ soc_remove_link_dais(card); soc_remove_link_components(card); -- 2.40.1

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[PATCH AUTOSEL 6.5 07/31] ASoC: fsl-asoc-card: use integer type for fll_id and pll_id
by Sasha Levin 18 Oct '23

18 Oct '23

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit 2b21207afd06714986a3d22442ed4860ba4f9ced ] As the pll_id and pll_id can be zero (WM8960_SYSCLK_AUTO) with the commit 2bbc2df46e67 ("ASoC: wm8960: Make automatic the default clocking mode") Then the machine driver will skip to call set_sysclk() and set_pll() for codec, when the sysclk rate is different with what wm8960 read at probe, the output sound frequency is wrong. So change the fll_id and pll_id initial value, still keep machine driver's behavior same as before. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Link: https://lore.kernel.org/r/1695202992-24864-1-git-send-email-shengjiu.wang@n… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl-asoc-card.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 76b5bfc288fde..bab7d34cf585b 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -52,8 +52,8 @@ struct codec_priv { unsigned long mclk_freq; unsigned long free_freq; u32 mclk_id; - u32 fll_id; - u32 pll_id; + int fll_id; + int pll_id; }; /** @@ -206,7 +206,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, } /* Specific configuration for PLL */ - if (codec_priv->pll_id && codec_priv->fll_id) { + if (codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) { if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) pll_out = priv->sample_rate * 384; else @@ -248,7 +248,7 @@ static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) priv->streams &= ~BIT(substream->stream); - if (!priv->streams && codec_priv->pll_id && codec_priv->fll_id) { + if (!priv->streams && codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) { /* Force freq to be free_freq to avoid error message in codec */ ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), codec_priv->mclk_id, @@ -621,6 +621,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.dapm_routes = audio_map; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); priv->card.driver_name = DRIVER_NAME; + + priv->codec_priv.fll_id = -1; + priv->codec_priv.pll_id = -1; + /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { codec_dai_name = "cs42888"; -- 2.40.1

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[PATCH AUTOSEL 6.5 04/31] ASoC: simple-card: fixup asoc_simple_probe() error handling
by Sasha Levin 18 Oct '23

18 Oct '23

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit 41bae58df411f9accf01ea660730649b2fab1dab ] asoc_simple_probe() is used for both "DT probe" (A) and "platform probe" (B). It uses "goto err" when error case, but it is not needed for "platform probe" case (B). Thus it is using "return" directly there. static int asoc_simple_probe(...) { ^ if (...) { | ... (A) if (ret < 0) | goto err; v } else { ^ ... | if (ret < 0) (B) return -Exxx; v } ... ^ if (ret < 0) (C) goto err; v ... err: (D) simple_util_clean_reference(card); return ret; } Both case are using (C) part, and it calls (D) when err case. But (D) will do nothing for (B) case. Because of these behavior, current code itself is not wrong, but is confusable, and more, static analyzing tool will warning on (B) part (should use goto err). To avoid static analyzing tool warning, this patch uses "goto err" on (B) part. Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)linaro.org> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/87o7hy7mlh.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/generic/simple-card.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 0745bf6a09aa0..5005d3c9c659a 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -701,10 +701,12 @@ static int asoc_simple_probe(struct platform_device *pdev) struct snd_soc_dai_link *dai_link = priv->dai_link; struct simple_dai_props *dai_props = priv->dai_props; + ret = -EINVAL; + cinfo = dev->platform_data; if (!cinfo) { dev_err(dev, "no info for asoc-simple-card\n"); - return -EINVAL; + goto err; } if (!cinfo->name || @@ -713,7 +715,7 @@ static int asoc_simple_probe(struct platform_device *pdev) !cinfo->platform || !cinfo->cpu_dai.name) { dev_err(dev, "insufficient asoc_simple_card_info settings\n"); - return -EINVAL; + goto err; } cpus = dai_link->cpus; -- 2.40.1

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[PATCH AUTOSEL 6.5 03/31] ASoC: Intel: sof_sdw: add support for SKU 0B14
by Sasha Levin 18 Oct '23

18 Oct '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit fb0b8d299781be8d46b3612aa96cef28da0d93f4 ] One more missing SKU in the list. Closes: https://github.com/thesofproject/linux/issues/4543 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Chao Song <chao.song(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20230919092125.1922468-1-yung-chuan.liao@linux.in… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index c86f8f9a61003..7d9b3a07945d7 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -366,6 +366,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { /* No Jack */ .driver_data = (void *)SOF_SDW_TGL_HDMI, }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B14"), + }, + /* No Jack */ + .driver_data = (void *)SOF_SDW_TGL_HDMI, + }, + { .callback = sof_sdw_quirk_cb, .matches = { -- 2.40.1

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[PATCH AUTOSEL 6.5 02/31] ASoC: Intel: soc-acpi: Add entry for sof_es8336 in MTL match table.
by Sasha Levin 18 Oct '23

18 Oct '23

From: Balamurugan C <balamurugan.c(a)intel.com> [ Upstream commit 381ddcd5875e496f2eae06bb65853271b7150fee ] Adding support for ES83x6 codec in MTL match table. Signed-off-by: Balamurugan C <balamurugan.c(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20230919091136.1922253-2-yung-chuan.liao@linux.in… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index ed9821adc1d9d..18da4b7e378d1 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -30,6 +30,11 @@ static const struct snd_soc_acpi_codecs mtl_rt5682_rt5682s_hp = { .codecs = {"10EC5682", "RTL5682"}, }; +static const struct snd_soc_acpi_codecs mtl_essx_83x6 = { + .num_codecs = 3, + .codecs = { "ESSX8316", "ESSX8326", "ESSX8336"}, +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { { .comp_ids = &mtl_rt5682_rt5682s_hp, @@ -52,6 +57,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { .quirk_data = &mtl_rt1019p_amp, .sof_tplg_filename = "sof-mtl-rt1019-rt5682.tplg", }, + { + .comp_ids = &mtl_essx_83x6, + .drv_name = "sof-essx8336", + .sof_tplg_filename = "sof-mtl-es8336", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER | + SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | + SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_machines); -- 2.40.1

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USB-Audio: Add FiiO K11 config
by GitHub pull_request - edited 18 Oct '23

18 Oct '23

alsa-project/alsa-ucm-conf pull request #360 was edited from ZVNexus: Without an ALSA config for the DAC autosuspend seems to be broken, every 10 seconds after pausing audio playback there is a 1 second delay to start it back up. Added a config for it and now there is no delay. Let me know if something else would make more sense. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/360 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/360.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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