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

[PATCH v2] ASoC: dt-bindings: Document audio OF graph dai-tdm-slot-num dai-tdm-slot-width props
by Marek Vasut 27 Oct '22

27 Oct '22

Document dai-tdm-slot-num and dai-tdm-slot-width props as those are parsed by simple graph card and may therefore appear in audio OF graph node. Signed-off-by: Marek Vasut <marex(a)denx.de> --- Cc: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: devicetree(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org To: linux-arm-kernel(a)lists.infradead.org --- V2: Drop the definition refs and just fill in type and description --- .../devicetree/bindings/sound/audio-graph-port.yaml | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml index bc46a95ed8400..64654ceef2089 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -74,6 +74,12 @@ patternProperties: convert-sample-format: $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-format" + dai-tdm-slot-num: + description: Number of slots in use. + $ref: /schemas/types.yaml#/definitions/uint32 + dai-tdm-slot-width: + description: Width in bits for each slot. + $ref: /schemas/types.yaml#/definitions/uint32 dai-tdm-slot-width-map: description: Mapping of sample widths to slot widths. For hardware that cannot support a fixed slot width or a slot width always -- 2.35.1

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[PATCH 00/16] ASoC: SOF: Intel: HDaudio cleanups
by Pierre-Louis Bossart 27 Oct '22

27 Oct '22

This is the part1 of my HDaudio cleanups, before the addition of to-be-announced HDaudio extensions. The patchset includes more consistent use of read/write/update helpers, removal of useless waits, structure members and programming sequences, removal of confusing sharing of private_data between FE and BE. Additional patches are coming to split the controller, codec and multi-link management functionality in well-identified files. Pierre-Louis Bossart (16): ASoC: SOF: ops: fallback to mmio in helpers ASoC: SOF: Intel: use mmio fallback for all platforms ASoC: SOF: ops: add readb/writeb helpers ASoC: SOF: ops: add snd_sof_dsp_updateb() helper ASoC: SOF: Intel: hda-dsp: use SOF helpers for consistency ASoC: SOF: Intel: hda-dai: start removing the use of runtime->private_data in BE ASoC: SOF: Intel: hda-dai: use component_get_drvdata to find hdac_bus ASoC: SOF: Intel: hda-dai: remove useless members in hda_pipe_params ASoC: SOF: Intel: hda-ctrl: remove useless sleep ASoC: SOF: Intel: hda: always do a full reset ASoC: SOF: Intel: hda: remove useless check on GCTL ASoC: SOF: Intel: hda-stream: use SOF helpers for consistency ASoC: SOF: Intel: hda-stream: rename CL_SD_CTL registers as SD_CTL ASoC: SOF: Intel: hda: use SOF helper for consistency ASoC: SOF: Intel: hda-stream: use snd_sof_dsp_updateb() helper ASoC: SOF: Intel: hda-stream: use readb/writeb for stream registers sound/soc/sof/intel/bdw.c | 6 +--- sound/soc/sof/intel/byt.c | 12 ++----- sound/soc/sof/intel/hda-common-ops.c | 6 +--- sound/soc/sof/intel/hda-ctrl.c | 41 ++++++++-------------- sound/soc/sof/intel/hda-dai.c | 31 +++++++---------- sound/soc/sof/intel/hda-dsp.c | 12 ++++--- sound/soc/sof/intel/hda-loader-skl.c | 30 ++++++++-------- sound/soc/sof/intel/hda-loader.c | 4 +-- sound/soc/sof/intel/hda-stream.c | 47 ++++++++++++------------- sound/soc/sof/intel/hda.c | 5 ++- sound/soc/sof/intel/hda.h | 25 +++++++------- sound/soc/sof/intel/pci-tng.c | 6 +--- sound/soc/sof/ops.h | 51 ++++++++++++++++++++-------- sound/soc/sof/sof-priv.h | 4 +++ 14 files changed, 136 insertions(+), 144 deletions(-) -- 2.34.1

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Add Focusrite Scarlett 2i2 gen2
by GitHub pull_request - opened 27 Oct '22

27 Oct '22

alsa-project/alsa-ucm-conf pull request #234 was opened from superseed: This is basically #203 but for the 2i2. I don't really know what I'm doing, but with these changes, I get two separate mono inputs in Pipewire, which is what I was looking for. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/234 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/234.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[BUG][v6.1-rc2] [SOF] max98373 i2c-MX98373:00: Runtime PM usage count underflow!
by Lukasz Majczak 27 Oct '22

27 Oct '22

Hi, I have noticed Runtime PM usage count underflow on my Chromebook (12th Gen Intel(R) Core(TM) i3-1215U) running vanilla v6.1-rc2: [ 7.558178] max98373 i2c-MX98373:00: Runtime PM usage count underflow! [ 7.558188] max98373 i2c-MX98373:01: Runtime PM usage count underflow! [ 7.558206] snd-soc-dummy snd-soc-dummy: Runtime PM usage count underflow! [ 7.558421] snd-soc-dummy snd-soc-dummy: Runtime PM usage count underflow! Link to the full dmesg: https://gist.github.com/semihalf-majczak-lukasz/d6a6d61d5937f762a524ed50fed… Best regards, Lukasz

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[PATCH] ALSA: Use del_timer_sync() before freeing timer
by Steven Rostedt 27 Oct '22

27 Oct '22

From: "Steven Rostedt (Google)" <rostedt(a)goodmis.org> The current code for freeing the emux timer is extremely dangerous: CPU0 CPU1 ---- ---- snd_emux_timer_callback() snd_emux_free() spin_lock(&emu->voice_lock) del_timer(&emu->tlist); <-- returns immediately spin_unlock(&emu->voice_lock); [..] kfree(emu); spin_lock(&emu->voice_lock); [BOOM!] Instead just use del_timer_sync() which will wait for the timer to finish before continuing. No need to check if the timer is active or not when doing so. This doesn't fix the race of a possible re-arming of the timer, but at least it won't use the data that has just been freed. Cc: stable(a)vger.kernel.org Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2") Signed-off-by: Steven Rostedt (Google) <rostedt(a)goodmis.org> --- sound/synth/emux/emux.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c index 5ed8e36d2e04..a2ee78809cfb 100644 --- a/sound/synth/emux/emux.c +++ b/sound/synth/emux/emux.c @@ -131,10 +131,7 @@ int snd_emux_free(struct snd_emux *emu) if (! emu) return -EINVAL; - spin_lock_irqsave(&emu->voice_lock, flags); - if (emu->timer_active) - del_timer(&emu->tlist); - spin_unlock_irqrestore(&emu->voice_lock, flags); + del_timer_sync(&emu->tlist); snd_emux_proc_free(emu); snd_emux_delete_virmidi(emu); -- 2.35.1

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[PATCH 0/2] ASoC: Intel: sof_rt5682: quirk auto detection
by Brent Lu 27 Oct '22

27 Oct '22

A new feature to compose quirk value according to the result of amplifier and SoC type detection at runtime. Brent Lu (2): ASoC: Intel: add helpers to detect SoCs ASoC: Intel: sof_rt5682: quirk auto detection include/linux/platform_data/x86/soc.h | 24 +++ sound/soc/intel/boards/sof_rt5682.c | 173 +++++++----------- .../intel/common/soc-acpi-intel-adl-match.c | 12 +- .../intel/common/soc-acpi-intel-jsl-match.c | 6 +- .../intel/common/soc-acpi-intel-tgl-match.c | 6 +- 5 files changed, 101 insertions(+), 120 deletions(-) -- 2.25.1

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Re: [PATCH 6/6] ASoC: Intel: boards: add missing dependency from realtek-common
by Mark Brown 27 Oct '22

27 Oct '22

On Mon, Oct 17, 2022 at 08:17:12AM +0200, Adam Borowski wrote: > These boards use code from sound/soc/sof/sof-audio.c > > Fixes: e1d5e13324020c4b405e63cae34560c7992bec2e > Signed-off-by: Adam Borowski <kilobyte(a)angband.pl> Please include human readable descriptions of things like commits and issues being discussed in e-mail in your mails, this makes them much easier for humans to read especially when they have no internet access. I do frequently catch up on my mail on flights or while otherwise travelling so this is even more pressing for me than just being about making things a bit easier to read. You've not incldued a cover letter for this and only copied me on some patches, are there any dependency issues?

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[PATCH v3 0/9] ASoC: Intel: avs: PCM power management
by Cezary Rojewski 27 Oct '22

27 Oct '22

Goal of the series is implementation of suspend/resume operations for a PCM stream along with all the collaterals connected to the subject. Start with splitting avs_dai_fe_hw_free() as ideally we would like to reuse as much of existing code as possible but snd_pcm_lib_free_pages() is not desired part of the function when speaking of suspend operation. The actual implementation of suspend/resume() for component drivers follows. For most scenarios, the PM flow is similar to standard streaming one, except for the part where the position register are being saved and the lack of PCM pages freeing. To reduce code duplication, all avs_dai_suspend_XXX() and avs_dai_resume_XXX() functions reuse their non-PM equivalents. Order of operations is affected by the fact that path binding/unbinding happens only in FE part of the stream. Above essentially unlocks SX+streaming scenarios i.e.: power transitions with an ongoing stream. As some streams are allowed to run in low power state, support is provided for S0iX state. The handlers check ACPI capabilities and the number of active low-power paths before deciding between SX and S0iX flows. The last portion of the patchset is addition of power/clock gating overrides. There is no single set of registers that ensures AudioDSP firmware loads 100% of time on every single configuration. By having them exposed, user can have the loading procedure behavior adjusted for their configuration without having to recompile the kernel. Changes in v3: - fixed unused-but-set-variable warnings reported by the test robot Changes in v2: - moved DRSM reg polling to separate function, snd_hdac_stream_wait_drsm() as suggested by Pierre - moved the acpi_gbl_FADT under AVS_S0IX_SUPPORTED macro and relocated the checks from pcm.c to topology.c so that the low-power streams are filtered as early as possible - fixed compilation when CONFIG_ACPI is disabled as reported by the test robot - simplified all the avs_dai_resume/suspend_xxx() by moving ->ignore_suspend check one level up, to avs_component_pm_op() (patch 0002) Amadeusz Sławiński (1): ASoC: Intel: avs: Handle SUSPEND and RESUME triggers Cezary Rojewski (7): ASoC: Intel: avs: Split pcm pages freeing operation from hw_free() ASoC: Intel: avs: Introduce PCM power management routines ALSA: hda: Introduce snd_hdac_stream_wait_drsm() ASoC: Intel: avs: Restart instead of resuming HDA capture streams ASoC: Intel: avs: Count low power streams ASoC: Intel: avs: Power and clock gating policy overriding ASoC: Intel: avs: Enact power gating policy Piotr Maziarz (1): ASoC: Intel: avs: Standby power-state support include/sound/hdaudio.h | 1 + include/sound/hdaudio_ext.h | 5 + sound/hda/hdac_stream.c | 21 ++ sound/soc/intel/avs/avs.h | 8 + sound/soc/intel/avs/core.c | 97 +++++++-- sound/soc/intel/avs/loader.c | 10 + sound/soc/intel/avs/pcm.c | 368 +++++++++++++++++++++++++++++---- sound/soc/intel/avs/topology.c | 10 + 8 files changed, 460 insertions(+), 60 deletions(-) -- 2.25.1

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[PATCH] ASoC: sof_es8336: reduce pop noise on speaker
by Zhu Ning 27 Oct '22

27 Oct '22

The Speaker GPIO needs to be turned on slightly behind the codec turned on. It also need to be turned off slightly before the codec turned down. Current code uses delay in DAPM_EVENT to do it but the mdelay delays the DAPM itself and thus has no effect. A delayed_work is added to turn on the speaker. The Speaker is turned off in .trigger since trigger is called slightly before the DAPM events. Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> --- sound/soc/intel/boards/sof_es8336.c | 58 ++++++++++++++++++++++------- 1 file changed, 44 insertions(+), 14 deletions(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index fbb42e54947a..7fde36a02d11 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -63,6 +63,7 @@ struct sof_es8336_private { struct snd_soc_jack jack; struct list_head hdmi_pcm_list; bool speaker_en; + struct delayed_work pcm_pop_work; }; struct sof_hdmi_pcm { @@ -111,6 +112,44 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk headset at mic1 port enabled\n"); } +static void pcm_pop_work_events(struct work_struct *work) +{ + struct sof_es8336_private *priv = + container_of(work, struct sof_es8336_private, pcm_pop_work.work); + + gpiod_set_value_cansleep(priv->gpio_speakers, priv->speaker_en); + + if (quirk & SOF_ES8336_HEADPHONE_GPIO) + gpiod_set_value_cansleep(priv->gpio_headphone, priv->speaker_en); + +} + +static int sof_8336_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct sof_es8336_private *priv = snd_soc_card_get_drvdata(card); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + break; + + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + if (priv->speaker_en == false) + if (substream->stream == 0) + gpiod_set_value_cansleep(priv->gpio_speakers, true); + break; + default: + return -EINVAL; + } + + return 0; +} + static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -122,19 +161,7 @@ static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, priv->speaker_en = !SND_SOC_DAPM_EVENT_ON(event); - if (SND_SOC_DAPM_EVENT_ON(event)) - msleep(70); - - gpiod_set_value_cansleep(priv->gpio_speakers, priv->speaker_en); - - if (!(quirk & SOF_ES8336_HEADPHONE_GPIO)) - return 0; - - if (SND_SOC_DAPM_EVENT_ON(event)) - msleep(70); - - gpiod_set_value_cansleep(priv->gpio_headphone, priv->speaker_en); - + queue_delayed_work(system_wq, &priv->pcm_pop_work, msecs_to_jiffies(70)); return 0; } @@ -344,6 +371,7 @@ static int sof_es8336_hw_params(struct snd_pcm_substream *substream, /* machine stream operations */ static struct snd_soc_ops sof_es8336_ops = { .hw_params = sof_es8336_hw_params, + .trigger = sof_8336_trigger, }; static struct snd_soc_dai_link_component platform_component[] = { @@ -723,7 +751,8 @@ static int sof_es8336_probe(struct platform_device *pdev) } INIT_LIST_HEAD(&priv->hdmi_pcm_list); - + INIT_DELAYED_WORK(&priv->pcm_pop_work, + pcm_pop_work_events); snd_soc_card_set_drvdata(card, priv); if (mach->mach_params.dmic_num > 0) { @@ -752,6 +781,7 @@ static int sof_es8336_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct sof_es8336_private *priv = snd_soc_card_get_drvdata(card); + cancel_delayed_work(&priv->pcm_pop_work); gpiod_put(priv->gpio_speakers); device_remove_software_node(priv->codec_dev); put_device(priv->codec_dev); -- 2.37.2

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[PATCH] ALSA: aoa: i2sbus: fix possible memory leak in i2sbus_add_dev()
by Yang Yingliang 27 Oct '22

27 Oct '22

dev_set_name() in soundbus_add_one() allocates memory for name, it need be freed when of_device_register() fails, call soundbus_dev_put() to give up the reference that hold in device_initialize(), so that it can be freed in kobject_cleanup() when the refcount hit to 0. And other resources are also freed in i2sbus_release_dev(), so it can return 0 directly. Fixes: f3d9478b2ce4 ("[ALSA] snd-aoa: add snd-aoa") Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/aoa/soundbus/i2sbus/core.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index faf6b03131ee..f6841daf9e3b 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -302,6 +302,10 @@ static int i2sbus_add_dev(struct macio_dev *macio, if (soundbus_add_one(&dev->sound)) { printk(KERN_DEBUG "i2sbus: device registration error!\n"); + if (dev->sound.ofdev.dev.kobj.state_initialized) { + soundbus_dev_put(&dev->sound); + return 0; + } goto err; } -- 2.25.1

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