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

[PATCH v2 0/3] Multiple headphone codec driver support
by Brent Lu 07 Oct '21

07 Oct '21

Support multiple headphone drivers in same machine driver. In this case, both rt5682 and rt5682s are supported and enumerated by different ACPI HID "10EC5682" and "RTL5682". V2 Changes: - remove useless 'NULL', 'false' in if-condition - can use 'comp_ids' field alone to enumerate driver - add comma to the end of entry in structure initialization - keep the table of byt/cht/cml/icl untouched Brent Lu (3): ASoC: soc-acpi: add comp_ids field for machine driver matching ASoC: Intel: sof_rt5682: detect codec variant in probe function ASoC: Intel: sof_rt5682: use comp_ids to enumerate rt5682s include/sound/soc-acpi.h | 2 ++ sound/soc/intel/boards/sof_rt5682.c | 34 +++--------------- .../intel/common/soc-acpi-intel-adl-match.c | 11 ++++-- .../intel/common/soc-acpi-intel-jsl-match.c | 35 +++++-------------- .../intel/common/soc-acpi-intel-tgl-match.c | 11 ++++-- sound/soc/soc-acpi.c | 24 +++++++++++-- 6 files changed, 52 insertions(+), 65 deletions(-) -- 2.25.1

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[PATCH v2 00/13] Extend AHUB audio support for Tegra210 and later
by Sameer Pujar 07 Oct '21

07 Oct '21

Earlier as part of series [0], support for ADMAIF and I/O modules (such as I2S, DMIC and DSPK) was added. This series aims at exposing some of the AHUB internal modules (listed below), which can be used for audio pre or post processing. * SFC (Sampling Frequency Converter) * MVC (Master Volume Control) * AMX (Audio Multiplexer) * ADX (Audio Demultiplexer) * Mixer These modules can be plugged into audio paths and relevant processing can be done. The MUX routes are extended to allow add or remove above modules in the path via mixer controls. This is similar to how specific ADMAIF channels are connected to relevant I/O module instances at the moment. Some of these modules can alter PCM parameters. Consider example of resampler (44.1 -> 48 kHz) in the path. aplay(44.1 kHz) -> ADMAIF -> SFC -> (48 kHz) I2S -> (48kHz) Codec The modules following SFC should be using converted sample rate and DAIs need to be configured accordingly. The audio-graph driver provides a mechanism to fixup the new parameters which can be specified in DT for a given DAI. Then core uses these new values via fixup callback and then pass it to respective DAIs hw_param() callback. The "convert-rate", described in [1], property can be used when there is rate conversion in the audio path. Similarly "convert-channels" can be used when there is channel conversion in the path. There is no "convert-xxx" property for sample size conversions. It can be added if necessary. [0] https://www.lkml.org/lkml/2020/7/21/1357 [1] Documentation/devicetree/bindings/sound/audio-graph-port.yaml Changelog ========= v1 -> v2 -------- * Put comments for soft reset application in the drivers. * Split out mute/volume control logic in put() calls of MVC driver and use separate callbacks for the respective kcontrols. * Update kcontrol put() callback in MVC driver to return 1 whenever there is change. Similar change is done in other drivers too. * Use name-prefix.yaml reference for the driver documentation now. * Add sound-name-prefix pattern for MIXER driver and use prefix accordingly in DT. Sameer Pujar (13): ASoC: soc-pcm: Don't reconnect an already active BE ASoC: simple-card-utils: Increase maximum DAI links limit to 512 ASoC: audio-graph: Fixup CPU endpoint hw_params in a BE<->BE link ASoC: dt-bindings: tegra: Few more Tegra210 AHUB modules ASoC: tegra: Add routes for few AHUB modules ASoC: tegra: Add Tegra210 based MVC driver ASoC: tegra: Add Tegra210 based SFC driver ASoC: tegra: Add Tegra210 based AMX driver ASoC: tegra: Add Tegra210 based ADX driver ASoC: tegra: Add Tegra210 based Mixer driver arm64: defconfig: Enable few Tegra210 based AHUB drivers arm64: tegra: Add few AHUB devices for Tegra210 and later arm64: tegra: Extend APE audio support on Jetson platforms .../bindings/sound/nvidia,tegra210-adx.yaml | 76 + .../bindings/sound/nvidia,tegra210-ahub.yaml | 20 + .../bindings/sound/nvidia,tegra210-amx.yaml | 76 + .../bindings/sound/nvidia,tegra210-mixer.yaml | 74 + .../bindings/sound/nvidia,tegra210-mvc.yaml | 76 + .../bindings/sound/nvidia,tegra210-sfc.yaml | 73 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 1554 ++++++++- arch/arm64/boot/dts/nvidia/tegra186.dtsi | 120 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 1493 +++++++- .../arm64/boot/dts/nvidia/tegra194-p3509-0000.dtsi | 1520 ++++++++- arch/arm64/boot/dts/nvidia/tegra194.dtsi | 116 + arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 876 +++++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 876 +++++ arch/arm64/boot/dts/nvidia/tegra210.dtsi | 77 + arch/arm64/configs/defconfig | 5 + include/sound/simple_card_utils.h | 2 +- sound/soc/generic/audio-graph-card.c | 4 +- sound/soc/soc-pcm.c | 4 + sound/soc/tegra/Kconfig | 48 + sound/soc/tegra/Makefile | 10 + sound/soc/tegra/tegra210_adx.c | 531 +++ sound/soc/tegra/tegra210_adx.h | 72 + sound/soc/tegra/tegra210_ahub.c | 511 ++- sound/soc/tegra/tegra210_amx.c | 600 ++++ sound/soc/tegra/tegra210_amx.h | 93 + sound/soc/tegra/tegra210_mixer.c | 674 ++++ sound/soc/tegra/tegra210_mixer.h | 100 + sound/soc/tegra/tegra210_mvc.c | 645 ++++ sound/soc/tegra/tegra210_mvc.h | 117 + sound/soc/tegra/tegra210_sfc.c | 3549 ++++++++++++++++++++ sound/soc/tegra/tegra210_sfc.h | 78 + 31 files changed, 13647 insertions(+), 423 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-amx.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml create mode 100644 sound/soc/tegra/tegra210_adx.c create mode 100644 sound/soc/tegra/tegra210_adx.h create mode 100644 sound/soc/tegra/tegra210_amx.c create mode 100644 sound/soc/tegra/tegra210_amx.h create mode 100644 sound/soc/tegra/tegra210_mixer.c create mode 100644 sound/soc/tegra/tegra210_mixer.h create mode 100644 sound/soc/tegra/tegra210_mvc.c create mode 100644 sound/soc/tegra/tegra210_mvc.h create mode 100644 sound/soc/tegra/tegra210_sfc.c create mode 100644 sound/soc/tegra/tegra210_sfc.h -- 2.7.4

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[PATCH 0/3] Multiple headphone codec driver support
by Brent Lu 07 Oct '21

07 Oct '21

Support multiple headphone drivers in same machine driver. In this case, both rt5682 and rt5682s are supported and enumerated by different ACPI HID "10EC5682" and "RTL5682". Brent Lu (3): ASoC: soc-acpi: add alternative id field for machine driver matching ASoC: Intel: sof_rt5682: detect codec variant in probe function ASoC: Intel: sof_rt5682: use id_alt to enumerate rt5682s include/sound/soc-acpi.h | 2 ++ sound/soc/intel/boards/sof_rt5682.c | 34 +++---------------- .../intel/common/soc-acpi-intel-adl-match.c | 8 +++++ .../intel/common/soc-acpi-intel-byt-match.c | 6 ++++ .../intel/common/soc-acpi-intel-cht-match.c | 6 ++++ .../intel/common/soc-acpi-intel-cml-match.c | 8 +++++ .../intel/common/soc-acpi-intel-icl-match.c | 6 ++++ .../intel/common/soc-acpi-intel-jsl-match.c | 32 +++++------------ .../intel/common/soc-acpi-intel-tgl-match.c | 8 +++++ sound/soc/soc-acpi.c | 21 +++++++++++- 10 files changed, 76 insertions(+), 55 deletions(-) -- 2.25.1

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DAPM PIN switches do not update in alsamixer when changed through UCM profile
by Hans de Goede 07 Oct '21

07 Oct '21

Hi All, I notice that DAPM PIN switches, such as e.g. the "Headphone" SOC_DAPM_PIN_SWITCH defined in: sound/soc/intel/boards/cht_bsw_nau8824.c: static const struct snd_kcontrol_new cht_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Int Mic"), SOC_DAPM_PIN_SWITCH("Ext Spk"), }; Do not get updated to reflect state-changes when the output is switched between e.g. Headphone / "Ext Spk" by pulseaudio/pipewire through the UCM profile mechanism. If I exit alsa-mixer after changing the output and start it again then the control does show the expect value. So it seems that we are failing to send a change event about this somewhere? Regards, Hans

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dlmisc.c: Avoid double unlock in snd_dlobj_cache_get0()
by GitHub pull_request - opened 07 Oct '21

07 Oct '21

alsa-project/alsa-lib pull request #182 was opened from johado: Remove call to snd_dlobj_unlock() in snd_dlobj_cache_get0() All lock/unlock is done by callers of the function. Fixes: https://github.com/alsa-project/alsa-lib/issues/181 Request URL : https://github.com/alsa-project/alsa-lib/pull/182 Patch URL : https://github.com/alsa-project/alsa-lib/pull/182.patch Repository URL: https://github.com/alsa-project/alsa-lib

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Double snd_dlobj_unlock()
by GitHub issues - edited 07 Oct '21

07 Oct '21

alsa-project/alsa-lib issue #181 was edited from johado: In dlmisc.c:snd_dlobj_cache_get0() there is a call to snd_dlobj_unlock() in the __err: path which should not be there. All lock/unlock are done by the callers to this function. ``` --- a/src/dlmisc.c +++ b/src/dlmisc.c @@ -359,7 +359,6 @@ snd_dlobj_cache_get0(const char *lib, const char *name, free(c); __err: snd_dlclose(dlobj); - snd_dlobj_unlock(); return NULL; } c->dlobj = dlobj; ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/181 Repository URL: https://github.com/alsa-project/alsa-lib

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Double snd_dlobj_unlock()
by GitHub issues - opened 07 Oct '21

07 Oct '21

alsa-project/alsa-lib issue #181 was opened from johado: In dlmisc.c:snd_dlobj_cache_get0() there is a call to snd_dlobj_unlock() in the __err: path which should not be there. All lock/unlock are done by the callers to this function. --- a/src/dlmisc.c +++ b/src/dlmisc.c @@ -359,7 +359,6 @@ snd_dlobj_cache_get0(const char *lib, const char *name, free(c); __err: snd_dlclose(dlobj); - snd_dlobj_unlock(); return NULL; } c->dlobj = dlobj; Issue URL : https://github.com/alsa-project/alsa-lib/issues/181 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ALSA: usb-audio: Pass JOINT_DUPLEX info flag for implicit fb streams
by Takashi Iwai 07 Oct '21

07 Oct '21

When a stream is in the implicit feedback mode, it's more or less tied with a capture stream. Passing SNDRV_PCM_INFO_JOINT_DUPLEX may help for user-space to understand the situation. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index f09c7380a923..3bb095a3c9b6 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1080,6 +1080,13 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre if (err < 0) return err; + list_for_each_entry(fp, &subs->fmt_list, list) { + if (fp->implicit_fb) { + runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; + break; + } + } + return 0; } -- 2.31.1

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ALC236 recording issue
by Kailang 07 Oct '21

07 Oct '21

Hi Takashi, Attach patch will solve alc236 recording issue for HP platform. Many Thanks. BR, Kailang

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[PATCH v3 0/4] ASoC: Intel: bytcht_es8316: few cleanups
by Andy Shevchenko 06 Oct '21

06 Oct '21

The small set of cleanups against bytcht_es8316 board file. In v3: - actually added a Pierre's tag (Mark) In v2: - added tag (Pierre) - added commit message to the patch 2 (Joe) Andy Shevchenko (4): ASoC: Intel: bytcht_es8316: Get platform data via dev_get_platdata() ASoC: Intel: bytcht_es8316: Use temporary variable for struct device ASoC: Intel: bytcht_es8316: Switch to use gpiod_get_optional() ASoC: Intel: bytcht_es8316: Utilize dev_err_probe() to avoid log saturation sound/soc/intel/boards/bytcht_es8316.c | 37 +++++++++----------------- 1 file changed, 12 insertions(+), 25 deletions(-) -- 2.33.0

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