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

[PATCH] ASoC: mmp-sspa: set phase two word length register
by Kyle Russell 20 Nov '20

20 Nov '20

If hw params enables dual phase transmission, then the word length for the second phase should be set to match the sample format instead of remaining at the reset default. This matches the configuration already being done for the first phase. This driver already sets the phase two sample size, so this should complete the phase two configuration. Signed-off-by: Kyle Russell <bkylerussell(a)gmail.com> --- sound/soc/pxa/mmp-sspa.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 52d4d8ace1c3..4803972ee655 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -246,6 +246,9 @@ static int mmp_sspa_hw_params(struct snd_pcm_substream *substream, sspa_ctrl &= ~SSPA_CTL_XWDLEN1_MASK; sspa_ctrl |= SSPA_CTL_XWDLEN1(bitval); + sspa_ctrl &= ~SSPA_CTL_XWDLEN2_MASK; + sspa_ctrl |= SSPA_CTL_XWDLEN2(bitval); + sspa_ctrl &= ~SSPA_CTL_XSSZ1_MASK; sspa_ctrl |= SSPA_CTL_XSSZ1(bitval); -- 2.25.1

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[PATCH] ASoC: codecs: lpass-va-macro: add missing MODULE_DEVICE_TABLE
by Srinivas Kandagatla 20 Nov '20

20 Nov '20

Fix module loading due by adding missing MODULE_DEVICE_TABLE. Fixes: 908e6b1df26e ("ASoC: codecs: lpass-va-macro: Add support to VA Macro") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/lpass-va-macro.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index b604de07e650..3e6bbef26dcb 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -1487,6 +1487,7 @@ static const struct of_device_id va_macro_dt_match[] = { { .compatible = "qcom,sm8250-lpass-va-macro" }, {} }; +MODULE_DEVICE_TABLE(of, va_macro_dt_match); static struct platform_driver va_macro_driver = { .driver = { -- 2.21.0

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[PATCH] ASoC: dt-bindings: renesas, rsnd: Fix duplicate 'allOf' entries
by Rob Herring 20 Nov '20

20 Nov '20

Commit e52f3f291152 ("ASoC: audio-graph-card: Refactor schema") added an 'allOf' entry, but one is already present in the schema. Multiple keys is not valid and results in an error: ruamel.yaml.constructor.DuplicateKeyError: while constructing a mapping in "<unicode string>", line 4, column 1 found duplicate key "allOf" with value "[]" (original value: "[]") in "<unicode string>", line 262, column 1 Fixes: e52f3f291152 ("ASoC: audio-graph-card: Refactor schema") Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Sameer Pujar <spujar(a)nvidia.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index cbfd5914b432..0fd37aa84947 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -9,10 +9,6 @@ title: Renesas R-Car Sound Driver Device Tree Bindings maintainers: - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> -allOf: - - $ref: audio-graph.yaml# - - $ref: audio-graph-port.yaml# - properties: compatible: @@ -260,6 +256,8 @@ required: - "#sound-dai-cells" allOf: + - $ref: audio-graph.yaml# + - $ref: audio-graph-port.yaml# - if: properties: compatible: -- 2.25.1

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[PATCH] ASoC: SOF: control: override volume info callback
by Kai Vehmanen 20 Nov '20

20 Nov '20

From: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> ASoC dapm controls currently don't support more than 2 channels. This is a problem for SOF-based devices where individual volume control cannot be provided on the 4 DMIC input path. If we want to provide controls for more than 2 channels, this patch suggests a simple solution based on an override of the info callback. For example, in the case with 4 channel DMIC PGAs, a sof_info callback would be used. Mono and stereo cases will keep using the existing dapm info callback. A longer-term solution would be to remove the limits to 2 channels in ASoC/DAPM/topology. This is a topic Intel is currently looking into, e.g. by removing the use of 'reg' and 'rreg' fields and use arrays instead. Such changes will be rather intrusive and touch multiple codec and platform drivers. Removing restrictions is the right thing to do, but this will need to be done in steps with lots of validation. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/control.c | 22 ++++++++++++++++++++++ sound/soc/sof/sof-audio.h | 2 ++ sound/soc/sof/topology.c | 9 +++++++++ 3 files changed, 33 insertions(+) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index 056c86ad5a47..a5dd728c580a 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -114,6 +114,28 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol, return change; } +int snd_sof_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *sm = (struct soc_mixer_control *)kcontrol->private_value; + struct snd_sof_control *scontrol = sm->dobj.private; + unsigned int channels = scontrol->num_channels; + int platform_max; + + if (!sm->platform_max) + sm->platform_max = sm->max; + platform_max = sm->platform_max; + + if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + + uinfo->count = channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = platform_max - sm->min; + return 0; +} + int snd_sof_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 9f645a2e5a6c..ddcfd46617f8 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -124,6 +124,8 @@ int snd_sof_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_sof_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_sof_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *ucontrol); int snd_sof_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_sof_switch_put(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 44fddeda6043..b0132fa56fe2 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1041,6 +1041,15 @@ static int sof_control_load_volume(struct snd_soc_component *scomp, goto out; } + /* + * If control has more than 2 channels we need to override the info. This is because even if + * ASoC layer has defined topology's max channel count to SND_SOC_TPLG_MAX_CHAN = 8, the + * pre-defined dapm control types (and related functions) creating the actual control + * restrict the channels only to mono or stereo. + */ + if (le32_to_cpu(mc->num_channels) > 2) + kc->info = snd_sof_volume_info; + /* init the volume get/put data */ scontrol->size = struct_size(scontrol->control_data, chanv, le32_to_cpu(mc->num_channels)); -- 2.28.0

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[PATCH] ASoC: SOF: nocodec: modify DAI link definitions
by Kai Vehmanen 20 Nov '20

20 Nov '20

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> The ignore_machine field in the component driver is used to ignore the FE DAI links defined in the machine driver, override BE fixups and set the stream names for the DAI links defined in the machine driver. This is required to make SOF compatible with the legacy machine drivers. In the case of the nocodec machine driver in SOF, there is no need to rely upon this ignore_machine logic in the core. Modify the machine driver to set DAI link stream names and the BE hw_params_fixup callback appropriately. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- include/sound/sof.h | 6 ++++-- sound/soc/sof/nocodec.c | 18 ++++++++++++++---- sound/soc/sof/pcm.c | 3 +-- sound/soc/sof/sof-audio.c | 2 +- sound/soc/sof/sof-audio.h | 3 +++ 5 files changed, 23 insertions(+), 9 deletions(-) diff --git a/include/sound/sof.h b/include/sound/sof.h index 9aa055289dcc..646a655c3c6b 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -100,6 +100,8 @@ struct sof_dev_desc { const struct snd_sof_dsp_ops *ops; }; -int sof_nocodec_setup(struct device *dev, - const struct snd_sof_dsp_ops *ops); +int sof_nocodec_setup(struct device *dev, const struct snd_sof_dsp_ops *ops, + int (*pcm_dai_link_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params)); + #endif diff --git a/sound/soc/sof/nocodec.c b/sound/soc/sof/nocodec.c index 9e922df6a710..3b9bb2e83a86 100644 --- a/sound/soc/sof/nocodec.c +++ b/sound/soc/sof/nocodec.c @@ -10,17 +10,21 @@ #include <linux/module.h> #include <sound/sof.h> +#include "sof-audio.h" #include "sof-priv.h" static struct snd_soc_card sof_nocodec_card = { .name = "nocodec", /* the sof- prefix is added by the core */ + .topology_shortname = "sof-nocodec", .owner = THIS_MODULE }; static int sof_nocodec_bes_setup(struct device *dev, const struct snd_sof_dsp_ops *ops, struct snd_soc_dai_link *links, - int link_num, struct snd_soc_card *card) + int link_num, struct snd_soc_card *card, + int (*pcm_dai_link_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params)) { struct snd_soc_dai_link_component *dlc; int i; @@ -39,6 +43,8 @@ static int sof_nocodec_bes_setup(struct device *dev, if (!links[i].name) return -ENOMEM; + links[i].stream_name = links[i].name; + links[i].cpus = &dlc[0]; links[i].codecs = &dlc[1]; links[i].platforms = &dlc[2]; @@ -57,6 +63,8 @@ static int sof_nocodec_bes_setup(struct device *dev, links[i].dpcm_playback = 1; if (ops->drv[i].capture.channels_min) links[i].dpcm_capture = 1; + + links[i].be_hw_params_fixup = pcm_dai_link_fixup; } card->dai_link = links; @@ -65,8 +73,9 @@ static int sof_nocodec_bes_setup(struct device *dev, return 0; } -int sof_nocodec_setup(struct device *dev, - const struct snd_sof_dsp_ops *ops) +int sof_nocodec_setup(struct device *dev, const struct snd_sof_dsp_ops *ops, + int (*pcm_dai_link_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params)) { struct snd_soc_dai_link *links; @@ -77,7 +86,7 @@ int sof_nocodec_setup(struct device *dev, return -ENOMEM; return sof_nocodec_bes_setup(dev, ops, links, ops->num_drv, - &sof_nocodec_card); + &sof_nocodec_card, pcm_dai_link_fixup); } EXPORT_SYMBOL(sof_nocodec_setup); @@ -86,6 +95,7 @@ static int sof_nocodec_probe(struct platform_device *pdev) struct snd_soc_card *card = &sof_nocodec_card; card->dev = &pdev->dev; + card->topology_shortname_created = true; return devm_snd_soc_register_card(&pdev->dev, card); } diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 37d12162e448..0dc39fbcd81d 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -620,8 +620,7 @@ static int sof_pcm_new(struct snd_soc_component *component, } /* fixup the BE DAI link to match any values from topology */ -static int sof_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) +int sof_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 9a8ce25a8fc8..3277489fee5e 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -468,7 +468,7 @@ int sof_machine_check(struct snd_sof_dev *sdev) mach->drv_name = "sof-nocodec"; sof_pdata->tplg_filename = desc->nocodec_tplg_filename; - ret = sof_nocodec_setup(sdev->dev, desc->ops); + ret = sof_nocodec_setup(sdev->dev, desc->ops, sof_pcm_dai_link_fixup); if (ret < 0) return ret; diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 9f645a2e5a6c..5b016f00956b 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -212,6 +212,9 @@ int snd_sof_ipc_set_get_comp_data(struct snd_sof_control *scontrol, enum sof_ipc_ctrl_cmd ctrl_cmd, bool send); +/* DAI link fixup */ +int sof_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params); + /* PM */ int sof_restore_pipelines(struct device *dev); int sof_set_hw_params_upon_resume(struct device *dev); -- 2.28.0

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ucm2: add support to for Qualcomm RB5 Platform
by GitHub pull_request - opened 20 Nov '20

20 Nov '20

alsa-project/alsa-ucm-conf pull request #63 was opened from Srinivas-Kandagatla: This patchset adds support to The Qualcomm Robotics RB5 Platform Audio ports. RB5 has 2 WSA881X smart speakers attached via Soundwire, HDMI and a DMIC0 on the board. All the audio related patches are merged in ASoC tree. Also to note that the driver name and long card names are fixed properly on all new and old machine drivers so that we can organise the ucm2 directories correctly. That is the reason for moving sm8250 out of Qualcomm folder. Thanks, srini Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/63 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/63.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Sm8250 v1
by GitHub pull_request - opened 20 Nov '20

20 Nov '20

alsa-project/alsa-ucm-conf pull request #62 was opened from Srinivas-Kandagatla: This patchset adds support to The Qualcomm Robotics RB5 Platform Audio ports. RB5 has 2 WSA881X smart speakers attached via Soundwire, HDMI and a DMIC0 on the board. All the audio related patches are merged in ASoC tree. Also to note that the driver name and long card names are fixed properly on all new and old machine drivers so that we can organise the ucm2 directories correctly. That is the reason for moving sm8250 out of Qualcomm folder. Thanks, srini Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/62 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/62.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Dell AIO wrong sound tone
by Kailang 20 Nov '20

20 Nov '20

Hi Takashi, Dell AIO only had one audio jack. It will get wrong sound tone with replug speaker or HP or headset. This patch will solve it. Many thanks. BR, Kailang

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[RFC PATCH] ASoC: ak4458: use reset control instead of reset gpio
by Viorel Suman (OSS) 20 Nov '20

20 Nov '20

From: Viorel Suman <viorel.suman(a)nxp.com> Using GPIO API seems not a way to go for the case when the same reset GPIO is used to control several codecs. For a such case we can use the "gpio-reset" driver and the "shared" reset API to manage the reset GPIO - to deassert the reset when the first codec is powered up and assert it when there is no codec in use. DTS is supposed to look as follows: / { ak4458_reset: gpio-reset { compatible = "gpio-reset"; reset-gpios = <&pca6416 4 GPIO_ACTIVE_LOW>; #reset-cells = <0>; initially-in-reset; }; }; &i2c3 { pca6416: gpio@20 { compatible = "ti,tca6416"; reg = <0x20>; gpio-controller; #gpio-cells = <2>; }; ak4458_1: ak4458@10 { compatible = "asahi-kasei,ak4458"; reg = <0x10>; resets = <&ak4458_reset>; }; ak4458_2: ak4458@11 { compatible = "asahi-kasei,ak4458"; reg = <0x11>; resets = <&ak4458_reset>; }; ak4458_3: ak4458@12 { compatible = "asahi-kasei,ak4458"; reg = <0x12>; resets = <&ak4458_reset>; }; }; Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> --- sound/soc/codecs/ak4458.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 1010c9ee2e83..f27727cb1382 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -13,6 +13,7 @@ #include <linux/of_gpio.h> #include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> +#include <linux/reset.h> #include <linux/slab.h> #include <sound/initval.h> #include <sound/pcm_params.h> @@ -45,7 +46,7 @@ struct ak4458_priv { const struct ak4458_drvdata *drvdata; struct device *dev; struct regmap *regmap; - struct gpio_desc *reset_gpiod; + struct reset_control *reset; struct gpio_desc *mute_gpiod; int digfil; /* SSLOW, SD, SLOW bits */ int fs; /* sampling rate */ @@ -597,17 +598,17 @@ static struct snd_soc_dai_driver ak4497_dai = { static void ak4458_power_off(struct ak4458_priv *ak4458) { - if (ak4458->reset_gpiod) { - gpiod_set_value_cansleep(ak4458->reset_gpiod, 0); - usleep_range(1000, 2000); + if (ak4458->reset) { + reset_control_assert(ak4458->reset); + msleep(20); } } static void ak4458_power_on(struct ak4458_priv *ak4458) { - if (ak4458->reset_gpiod) { - gpiod_set_value_cansleep(ak4458->reset_gpiod, 1); - usleep_range(1000, 2000); + if (ak4458->reset) { + reset_control_deassert(ak4458->reset); + msleep(20); } } @@ -685,7 +686,6 @@ static int __maybe_unused ak4458_runtime_resume(struct device *dev) if (ak4458->mute_gpiod) gpiod_set_value_cansleep(ak4458->mute_gpiod, 1); - ak4458_power_off(ak4458); ak4458_power_on(ak4458); regcache_cache_only(ak4458->regmap, false); @@ -771,10 +771,9 @@ static int ak4458_i2c_probe(struct i2c_client *i2c) ak4458->drvdata = of_device_get_match_data(&i2c->dev); - ak4458->reset_gpiod = devm_gpiod_get_optional(ak4458->dev, "reset", - GPIOD_OUT_LOW); - if (IS_ERR(ak4458->reset_gpiod)) - return PTR_ERR(ak4458->reset_gpiod); + ak4458->reset = devm_reset_control_get_optional_shared(ak4458->dev, NULL); + if (IS_ERR(ak4458->reset)) + return PTR_ERR(ak4458->reset); ak4458->mute_gpiod = devm_gpiod_get_optional(ak4458->dev, "mute", GPIOD_OUT_LOW); -- 2.26.2

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Lenovo t14s amd only Dummy Device as output in pulseaudio
by GitHub issues - opened 20 Nov '20

20 Nov '20

alsa-project/alsa-ucm-conf issue #61 was opened from J-Bu: Hi, since upgrading to alsa-ucm 1.2.4 I only have a "Dummy Device" instead of the internal speaker. When using alsa-ucm 1.2.3 all is fine. I'm not quite sure if this is a pulseaudio or alsa issue maybe you can guide me in the right direction Output of lspci: ``` 06:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] Device 1637 Subsystem: Lenovo Device 5082 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 32 bytes Interrupt: pin B routed to IRQ 85 Region 0: Memory at fd3c8000 (32-bit, non-prefetchable) [size=16K] Capabilities: [48] Vendor Specific Information: Len=08 <?> Capabilities: [50] Power Management version 3 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1+,D2+,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable+ DSel=0 DScale=0 PME+ Capabilities: [64] Express (v2) Legacy Endpoint, MSI 00 DevCap: MaxPayload 256 bytes, PhantFunc 0, Latency L0s <4us, L1 unlimited ExtTag+ AttnBtn- AttnInd- PwrInd- RBE+ FLReset- DevCtl: CorrErr- NonFatalErr- FatalErr- UnsupReq- RlxdOrd+ ExtTag+ PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 512 bytes DevSta: CorrErr- NonFatalErr- FatalErr- UnsupReq- AuxPwr- TransPend- LnkCap: Port #0, Speed 16GT/s, Width x16, ASPM L0s L1, Exit Latency L0s <64ns, L1 <1us ClockPM- Surprise- LLActRep- BwNot- ASPMOptComp+ LnkCtl: ASPM Disabled; RCB 64 bytes, Disabled- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 16GT/s (ok), Width x16 (ok) TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- DevCap2: Completion Timeout: Range ABCD, TimeoutDis+ NROPrPrP- LTR- 10BitTagComp+ 10BitTagReq- OBFF Not Supported, ExtFmt+ EETLPPrefix+, MaxEETLPPrefixes 1 EmergencyPowerReduction Not Supported, EmergencyPowerReductionInit- FRS- AtomicOpsCap: 32bit- 64bit- 128bitCAS- DevCtl2: Completion Timeout: 50us to 50ms, TimeoutDis- LTR- OBFF Disabled, AtomicOpsCtl: ReqEn- LnkSta2: Current De-emphasis Level: -3.5dB, EqualizationComplete- EqualizationPhase1- EqualizationPhase2- EqualizationPhase3- LinkEqualizationRequest- Retimer- 2Retimers- CrosslinkRes: unsupported Capabilities: [a0] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee09000 Data: 4024 Capabilities: [100 v1] Vendor Specific Information: ID=0001 Rev=1 Len=010 <?> Kernel driver in use: snd_hda_intel ``` I bisected the issue: ``` git bisect start # good: [dbac4c3e7ad05b4eb17964688fd9f07c61b8d336] ucm2: fix more indentation issues git bisect good dbac4c3e7ad05b4eb17964688fd9f07c61b8d336 # bad: [334b12e9a796cddf70779b68418ee7ad52ba1d14] sof-soundwire: use the codecs/hda/hdmi.conf macro git bisect bad 334b12e9a796cddf70779b68418ee7ad52ba1d14 # bad: [8a9609cc694cb1de501a67d7c091edaa9a338932] cht-bsw-rt5672: Add Lenovo ThinkPad 10 specific configuration git bisect bad 8a9609cc694cb1de501a67d7c091edaa9a338932 # good: [9f2dae6a560c60593c24aa4b85aed07418f9dd8e] sof-soundwire: initial UCM2 version git bisect good 9f2dae6a560c60593c24aa4b85aed07418f9dd8e # bad: [73c105b4707b601de63d1f24641e65e5ac23f3a6] DB820c/DB845c: move to Qualcomm/ tree git bisect bad 73c105b4707b601de63d1f24641e65e5ac23f3a6 # good: [38e5906cd1b18b6f4df5a1d35ca535325e7b65a0] sof-hda-dsp: fix the device order (Hdmi devices) git bisect good 38e5906cd1b18b6f4df5a1d35ca535325e7b65a0 # bad: [6b8518c431cac18a825bd7342875f54b44cb7afc] DAISY-I2S: move to Samsung/snow/snow.conf git bisect bad 6b8518c431cac18a825bd7342875f54b44cb7afc # bad: [dcef48f13d4f5db79b006755074940b94730a883] HDA-Intel: add support for AMD acp microphone devices git bisect bad dcef48f13d4f5db79b006755074940b94730a883 # first bad commit: [dcef48f13d4f5db79b006755074940b94730a883] HDA-Intel: add support for AMD acp microphone devices ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/61 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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