March 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH v2] ALSA: hda/realtek: Fix audio regression on Mi Notebook Pro 2020
by Kai-Heng Feng 30 Mar '22

30 Mar '22

Commit 5aec98913095 ("ALSA: hda/realtek - ALC236 headset MIC recording issue") is to solve recording issue met on AL236, by matching codec variant ALC269_TYPE_ALC257 and ALC269_TYPE_ALC256. This match can be too broad and Mi Notebook Pro 2020 is broken by the patch. Instead, use codec ID to be narrow down the scope, in order to make ALC256 unaffected. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=215484 Fixes: 5aec98913095 ("ALSA: hda/realtek - ALC236 headset MIC recording issue") Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v2: Fix wrong comparison, || -> && sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4c33cb57963db..aace474a899de 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3617,8 +3617,8 @@ static void alc256_shutup(struct hda_codec *codec) /* If disable 3k pulldown control for alc257, the Mic detection will not work correctly * when booting with headset plugged. So skip setting it for the codec alc257 */ - if (spec->codec_variant != ALC269_TYPE_ALC257 && - spec->codec_variant != ALC269_TYPE_ALC256) + if (codec->core.vendor_id != 0x10ec0236 && + codec->core.vendor_id != 0x10ec0257) alc_update_coef_idx(codec, 0x46, 0, 3 << 12); if (!spec->no_shutup_pins) -- 2.34.1

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[PATCH 0/2] rawmidi/meson: fix miscellaneous bugs
by GitHub pull_request - opened 30 Mar '22

30 Mar '22

alsa-project/alsa-gobject pull request #65 was opened from takaswie: This patchset fixes unintended two items below. * argument name against convention in this project * line break inner identifier in meson.build ``` Takashi Sakamoto (2): rawmidi: fix argument names with suffix '_id' meson: fix unintended line break for identifier src/rawmidi/query.c | 12 ++++++------ src/rawmidi/query.h | 2 +- src/seq/meson.build | 3 +-- 3 files changed, 8 insertions(+), 9 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/65 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/65.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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Fix support for sof-essx8336
by GitHub pull_request - opened 30 Mar '22

30 Mar '22

alsa-project/alsa-ucm-conf pull request #144 was opened from mchehab: There are a couple of issues at the support for sof-essx8336, causing it to fail working. This PR fix them. Tested on Huawei Matebook D15. Before the fix, the device was appearing as "sysdefault:0" instead of "hw:0", and there were several other problems. After the fix: - the device is now identified as hw:0; - calling alsamixer without `-Dsysdefault:0` (or `-Dhw:0`) now shows a mixer (instead of showing nothing); - The HDMI volume controls now appear; - Using `alsaucm load` now works fine. There are still a couple of things to be fixed, but I suspect that at least some of them would require firmware (or driver?) changes: - the internal microphone is not working yet; - the headphone/headset doesn't work. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/144 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/144.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Nhlt plugin and simple plugin interface
by GitHub pull_request - edited 29 Mar '22

29 Mar '22

alsa-project/alsa-utils pull request #129 was edited from juimonen: Initial something... I need to figure out: - where to specify which plugins to load/run (nhlt now hard coded) - where to install the plugins - how to get the to-be-loaded plugin names and where - is the "hidden-visibility" gcc flag ok to hide other than .so entrypoint Request URL : https://github.com/alsa-project/alsa-utils/pull/129 Patch URL : https://github.com/alsa-project/alsa-utils/pull/129.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH] ASoC: SOF: topology: use new sound control LED layer
by Jaroslav Kysela 29 Mar '22

29 Mar '22

Use the new sound control LED layer instead the direct ledtrig_audio_set() call - see 22d8de62f11b ("ALSA: control - add generic LED trigger module as the new control layer"). Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/control.c | 33 --------------------------------- sound/soc/sof/topology.c | 10 ++++++++++ 2 files changed, 10 insertions(+), 33 deletions(-) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index de1778c4002b..3b5718a3516d 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -15,36 +15,6 @@ #include "sof-priv.h" #include "sof-audio.h" -static void update_mute_led(struct snd_sof_control *scontrol, - struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - int temp = 0; - int mask; - int i; - - mask = 1U << snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - - for (i = 0; i < scontrol->num_channels; i++) { - if (ucontrol->value.integer.value[i]) { - temp |= mask; - break; - } - } - - if (temp == scontrol->led_ctl.led_value) - return; - - scontrol->led_ctl.led_value = temp; - -#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) - if (!scontrol->led_ctl.direction) - ledtrig_audio_set(LED_AUDIO_MUTE, temp ? LED_OFF : LED_ON); - else - ledtrig_audio_set(LED_AUDIO_MICMUTE, temp ? LED_OFF : LED_ON); -#endif -} - int snd_sof_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -121,9 +91,6 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); const struct sof_ipc_tplg_ops *tplg_ops = sdev->ipc->ops->tplg; - if (scontrol->led_ctl.use_led) - update_mute_led(scontrol, kcontrol, ucontrol); - if (tplg_ops->control->switch_put) return tplg_ops->control->switch_put(scontrol, ucontrol); diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 9b11e9795a7a..267a781494da 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -773,6 +773,7 @@ static int sof_control_load_volume(struct snd_soc_component *scomp, struct snd_soc_tplg_mixer_control *mc = container_of(hdr, struct snd_soc_tplg_mixer_control, hdr); int tlv[TLV_ITEMS]; + unsigned int mask; int ret; /* validate topology data */ @@ -821,6 +822,15 @@ static int sof_control_load_volume(struct snd_soc_component *scomp, goto err; } + if (scontrol->led_ctl.use_led) { + mask = scontrol->led_ctl.direction ? SNDRV_CTL_ELEM_ACCESS_MIC_LED : + SNDRV_CTL_ELEM_ACCESS_SPK_LED; + scontrol->access &= ~SNDRV_CTL_ELEM_ACCESS_LED_MASK; + scontrol->access |= mask; + kc->access &= ~SNDRV_CTL_ELEM_ACCESS_LED_MASK; + kc->access |= mask; + } + dev_dbg(scomp->dev, "tplg: load kcontrol index %d chans %d\n", scontrol->comp_id, scontrol->num_channels); -- 2.34.1

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[PATCH 0/6] ASRC support on Tegra186 and later
by Sameer Pujar 29 Mar '22

29 Mar '22

This series adds support for Asynchronous Sample Rate Converter (ASRC) module on Tegra186 and later generations of SoCs. ASRC is a client of AHUB. The driver and DT support is added to make it work with Tegra audio graph card. The module can be plugged into audio path using ALSA mixer controls. ASRC supports two modes of operation, where it gets the ratio info from SW and ratio detector module. Presently the support is added for SW mode. Sameer Pujar (6): ASoC: tegra: Add binding doc for ASRC module ASoC: tegra: Add Tegra186 based ASRC driver ASoC: tegra: AHUB routes for ASRC module arm64: defconfig: Build Tegra ASRC module arm64: tegra: Add ASRC device on Tegra186 and later arm64: tegra: Enable ASRC on various platforms .../bindings/sound/nvidia,tegra186-asrc.yaml | 81 ++ .../bindings/sound/nvidia,tegra210-ahub.yaml | 4 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 223 +++++ arch/arm64/boot/dts/nvidia/tegra186.dtsi | 7 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 223 +++++ .../arm64/boot/dts/nvidia/tegra194-p3509-0000.dtsi | 223 +++++ arch/arm64/boot/dts/nvidia/tegra194.dtsi | 8 + .../dts/nvidia/tegra234-p3737-0000+p3701-0000.dts | 223 +++++ arch/arm64/boot/dts/nvidia/tegra234.dtsi | 8 + arch/arm64/configs/defconfig | 1 + sound/soc/tegra/Kconfig | 12 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra186_asrc.c | 1045 ++++++++++++++++++++ sound/soc/tegra/tegra186_asrc.h | 113 +++ sound/soc/tegra/tegra210_ahub.c | 82 +- 15 files changed, 2254 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-asrc.yaml create mode 100644 sound/soc/tegra/tegra186_asrc.c create mode 100644 sound/soc/tegra/tegra186_asrc.h -- 2.7.4

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[PATCH] ALSA: hda/realtek: Fix audio regression on Mi Notebook Pro 2020
by Kai-Heng Feng 29 Mar '22

29 Mar '22

Commit 5aec98913095 ("ALSA: hda/realtek - ALC236 headset MIC recording issue") is to solve recording issue met on AL236, by matching codec variant ALC269_TYPE_ALC257 and ALC269_TYPE_ALC256. This match can be too broad and Mi Notebook Pro 2020 is broken by the patch. Instead, use codec ID to be narrow down the scope, in order to make ALC256 unaffected. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=215484 Fixes: 5aec98913095 ("ALSA: hda/realtek - ALC236 headset MIC recording issue") Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4c33cb57963db..0d5e6cfa44888 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3617,8 +3617,8 @@ static void alc256_shutup(struct hda_codec *codec) /* If disable 3k pulldown control for alc257, the Mic detection will not work correctly * when booting with headset plugged. So skip setting it for the codec alc257 */ - if (spec->codec_variant != ALC269_TYPE_ALC257 && - spec->codec_variant != ALC269_TYPE_ALC256) + if (codec->core.vendor_id != 0x10ec0236 || + codec->core.vendor_id != 0x10ec0257) alc_update_coef_idx(codec, 0x46, 0, 3 << 12); if (!spec->no_shutup_pins) -- 2.34.1

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USB-Audio: Add profile for MSI MPG X570S Carbon Max Wifi.
by GitHub pull_request - opened 29 Mar '22

29 Mar '22

alsa-project/alsa-ucm-conf pull request #143 was opened from lordhoto: This patch depends on Linux 5.17. This board features an Realtek ALC4080 chip. The various inputs and outputs are distributed over multiple hardware devices: * Input - hw:$card,0 loop back from speaker (called "Analog In") - hw:$card,1 line in at back panel - hw:$card,2 microphone from front AND back panel * Output - hw:$card,0 speaker output at back panel (up to 7.1) - hw:$card,1 headerphone output at front panel - hw:$card,2 could not figure this one out - hw:$card,3 guessed S/PDIF output (sadly no way to test) By default Mic and Line in are disabled. This confuses applications, thus I made the profile enable them by default. Without an UCM profile PulseAudio is not able to create output on the front panel nor record any microphone input. Signed-off-by: Johannes Schickel <lordhoto(a)gmail.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/143 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/143.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[RFC PATCH v2 0/6] Flexible codec clock configuration
by Sameer Pujar 29 Mar '22

29 Mar '22

Typically the codec drivers require setting up of Sysclk. Sometimes presence of internal PLL can provide more options of Sysclk configuration. Presently ASoC provides callbacks set_sysclk() and set_pll() in such cases. However it comes with following limitations considering generic machine drivers (simple-card or audio-graph-card): 1. The Sysclk source needs to be passed to set_sysclk() callback. Presently simple-card or audio-graph-card card rely on default source value (which is 0). If any other source needs to be used, it is currently not possible. 2. The same would be true for codec PLL configuration as well, though simple-card or audio-graph-card don't have support yet for the PLL configuration. Earlier attempt[0] to address above was not felt suitable. The suggestion was to use standard clock based bindings instead. This RFC series takes RT5659 as a reference and exposes clock relationships via DT. **This is not in the final shape yet**, but I wanted to get some valuable feedback to understand if the idea is right. If this appears fine, this can be extended to other codecs (wherever necessary). This does not completely remove the need of set_sysclk() callback because the clock requirement (MCLK * fs) would come from the machine driver. But machine driver need not worry about Sysclk source. It would be internally managed by Codec via DT clock relationships. [0] https://patchwork.kernel.org/project/alsa-devel/list/?series=438531&archive… Changelog: ========== v1 -> v2: --------- * New patch added to convert rt5659 binding doc to YAML format * New patch added to document audio-graph-port binding for rt5659 * New patch added to document clock binding enhancement for rt5659 Sameer Pujar (6): ASoC: dt-bindings: Convert rt5659 bindings to YAML schema ASoC: dt-bindings: Add audio-graph-port bindings to rt5659 ASoC: dt-bindings: Extend clock bindings of rt5659 ASoC: soc-pcm: tweak DPCM BE hw_param() call order ASoC: rt5659: Expose internal clock relationships ASoC: tegra: Get clock rate in consumer mode .../devicetree/bindings/sound/realtek,rt5659.yaml | 179 +++++++++++++++ Documentation/devicetree/bindings/sound/rt5659.txt | 89 -------- sound/soc/codecs/rt5659.c | 248 ++++++++++++++++++++- sound/soc/codecs/rt5659.h | 9 + sound/soc/soc-pcm.c | 60 ++++- sound/soc/tegra/tegra210_i2s.c | 25 ++- 6 files changed, 503 insertions(+), 107 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5659.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rt5659.txt -- 2.7.4

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[PATCH] mediatek: mt8195: fix a missing check on list iterator
by Xiaomeng Tong 29 Mar '22

29 Mar '22

The bug is here: mt8195_etdm_hw_params_fixup(runtime, params); For the for_each_card_rtds(), just like list_for_each_entry(), the list iterator 'runtime' will point to a bogus position containing HEAD if the list is empty or no element is found. This case must be checked before any use of the iterator, otherwise it will lead to a invalid memory access. To fix the bug, use a new variable 'iter' as the list iterator, while use the original variable 'runtime' as a dedicated pointer to point to the found element. Cc: stable(a)vger.kernel.org Fixes: 3d00d2c07f04f ("ASoC: mediatek: mt8195: add sof support on mt8195-mt6359-rt1019-rt5682") Signed-off-by: Xiaomeng Tong <xiam0nd.tong(a)gmail.com> --- .../mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index 29c2d3407cc7..dc91877e4c3c 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -814,7 +814,7 @@ static int mt8195_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_soc_card *card = rtd->card; struct snd_soc_dai_link *sof_dai_link = NULL; - struct snd_soc_pcm_runtime *runtime; + struct snd_soc_pcm_runtime *runtime = NULL, *iter; struct snd_soc_dai *cpu_dai; int i, j, ret = 0; @@ -824,16 +824,17 @@ static int mt8195_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, if (strcmp(rtd->dai_link->name, conn->normal_link)) continue; - for_each_card_rtds(card, runtime) { - if (strcmp(runtime->dai_link->name, conn->sof_link)) + for_each_card_rtds(card, iter) { + if (strcmp(iter->dai_link->name, conn->sof_link)) continue; - for_each_rtd_cpu_dais(runtime, j, cpu_dai) { + for_each_rtd_cpu_dais(iter, j, cpu_dai) { if (cpu_dai->stream_active[conn->stream_dir] > 0) { - sof_dai_link = runtime->dai_link; + sof_dai_link = iter->dai_link; break; } } + runtime = iter; break; } @@ -845,7 +846,8 @@ static int mt8195_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, if (!strcmp(rtd->dai_link->name, "ETDM2_IN_BE") || !strcmp(rtd->dai_link->name, "ETDM1_OUT_BE")) { - mt8195_etdm_hw_params_fixup(runtime, params); + if (runtime) + mt8195_etdm_hw_params_fixup(runtime, params); } return ret; -- 2.17.1

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