July 2024 - Alsa-devel - mailman.alsa-project.org

ASUS RoG Strix Z790-F: SPDIF (PCM) has wrong channels | how to get passthrough working?
by GitHub issues - opened 29 Jul '24

29 Jul '24

alsa-project/alsa-ucm-conf issue #434 was opened from Nightwing0815: Hi there, finally, [after a long journey,](https://github.com/alsa-project/alsa-ucm-conf/issues/350) yesterday I did an update from linuxmint to newest version 22, based on ubuntu 24.04 (I'll attach alsa-info and a system info). Now it's no big problem any more, to get SPDIF working: I just selected the right device and muted all others in alsamixer, except the pcm signal. But hey, there's a new problem: With a speaker-test -c6, I'm not able to hear my rear channels and the lfe channel, where they should be. Rear coming on front speakers... So, what do I have to do? Further, I'm asking for help to configure a pcm signal passthru for dts and (e)ac3 soundfiles. My teufel receiver recognices only a 5.1 stereo source, not an DD5.1 or DTS signal... [alsa-info.txt](https://github.com/user-attachments/files/16417700/alsa-info… [sysinfo.txt](https://github.com/user-attachments/files/16417724/sysinfo.txt) Thanks in advance for getting this problems solved, ToM Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/434 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 0/7] ASoC: codecs: wsa88xx and wcd93xx: Soundwire port non-functional cleanup
by Krzysztof Kozlowski 29 Jul '24

29 Jul '24

Hi, Few cleanups to make the code more robust or readable. No functional impact (compiled objects stay the same). Best regards, Krzysztof --- Krzysztof Kozlowski (7): ASoC: codecs: wsa881x: Use designator array initializers for Soundwire ports ASoC: codecs: wsa883x: Use designator array initializers for Soundwire ports ASoC: codecs: wsa884x: Use designator array initializers for Soundwire ports ASoC: codecs: wcd938x: Drop unused defines and enums ASoC: codecs: wcd937x: Move max port number defines to enum ASoC: codecs: wcd938x: Move max port number defines to enum ASoC: codecs: wcd939x: Move max port number defines to enum sound/soc/codecs/wcd937x.h | 34 +++++++++++++++++----------------- sound/soc/codecs/wcd938x.c | 18 ------------------ sound/soc/codecs/wcd938x.h | 4 ++-- sound/soc/codecs/wcd939x.h | 6 +++--- sound/soc/codecs/wsa881x.c | 42 ++++++++++++++++++++++-------------------- sound/soc/codecs/wsa883x.c | 42 ++++++++++++++++++++++-------------------- sound/soc/codecs/wsa884x.c | 34 ++++++++++++++++++++++------------ 7 files changed, 88 insertions(+), 92 deletions(-) --- base-commit: 66df68b9536e7465aec703733ea91ca39b787ed8 change-id: 20240725-asoc-wsa88xx-port-arrays-c58c96cd888e Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

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[PATCH v2] ASoC: sti: add missing probe entry for player and reader
by Jerome Audu 29 Jul '24

29 Jul '24

This patch addresses a regression in the ASoC STI drivers that was introduced in Linux version 6.6.y. The issue originated from a series of patches (see https://lore.kernel.org/all/87wmy5b0wt.wl-kuninori.morimoto.gx@renesas.com/) that unintentionally omitted necessary probe functions for the player and reader components. Probe function in `sound/soc/sti/sti_uniperif.c:415` is being replaced by another probe function located at `sound/soc/sti/sti_uniperif.c:453`, which should instead be derived from the player and reader components. This patch correctly reinserts the missing probe entries, restoring the intended functionality. Fixes: 9f625f5e6cf9 ("ASoC: sti: merge DAI call back functions into ops") Signed-off-by: Jerome Audu <jau(a)free.fr> --- Changes in v2: - Update changelog - Link to v1: https://lore.kernel.org/r/20240721-sti-audio-fix-v1-1-a8b1ecf61cb4@free.fr --- sound/soc/sti/sti_uniperif.c | 2 +- sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 1 + sound/soc/sti/uniperif_reader.c | 1 + 4 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index ba824f14a39c..a7956e5a4ee5 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -352,7 +352,7 @@ static int sti_uniperiph_resume(struct snd_soc_component *component) return ret; } -static int sti_uniperiph_dai_probe(struct snd_soc_dai *dai) +int sti_uniperiph_dai_probe(struct snd_soc_dai *dai) { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct sti_uniperiph_dai *dai_data = &priv->dai_data; diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index 2a5de328501c..74e51f0ff85c 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1380,6 +1380,7 @@ int uni_reader_init(struct platform_device *pdev, struct uniperif *reader); /* common */ +int sti_uniperiph_dai_probe(struct snd_soc_dai *dai); int sti_uniperiph_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index dd9013c47664..6d1ce030963c 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -1038,6 +1038,7 @@ static const struct snd_soc_dai_ops uni_player_dai_ops = { .startup = uni_player_startup, .shutdown = uni_player_shutdown, .prepare = uni_player_prepare, + .probe = sti_uniperiph_dai_probe, .trigger = uni_player_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 065c5f0d1f5f..05ea2b794eb9 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -401,6 +401,7 @@ static const struct snd_soc_dai_ops uni_reader_dai_ops = { .startup = uni_reader_startup, .shutdown = uni_reader_shutdown, .prepare = uni_reader_prepare, + .probe = sti_uniperiph_dai_probe, .trigger = uni_reader_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, --- base-commit: 0c3836482481200ead7b416ca80c68a29cfdaabd change-id: 20240721-sti-audio-fix-1374fffb24e7 Best regards, -- Jerome Audu <jau(a)free.fr>

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[RESEND] ASoC: codecs: ES8326: suspend issue
by Zhang Yi 29 Jul '24

29 Jul '24

We find that we need to disable micbias for the codec to enter suspend So We modify the trigger conditions for enable_micbias and disable_micbias Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index b246694ebb4f..60877116c0ef 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -805,6 +805,7 @@ static void es8326_jack_button_handler(struct work_struct *work) SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); button_to_report = 0; } + es8326_disable_micbias(es8326->component); } mutex_unlock(&es8326->lock); } @@ -878,7 +879,6 @@ static void es8326_jack_detect_handler(struct work_struct *work) regmap_write(es8326->regmap, ES8326_INT_SOURCE, 0x00); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x00); - es8326_enable_micbias(es8326->component); usleep_range(50000, 70000); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x00); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x10); @@ -897,6 +897,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) dev_dbg(comp->dev, "button pressed\n"); regmap_write(es8326->regmap, ES8326_INT_SOURCE, (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON)); + es8326_enable_micbias(es8326->component); queue_delayed_work(system_wq, &es8326->button_press_work, 10); goto exit; } @@ -1067,6 +1068,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC_MUTE, 0x0f); regmap_write(es8326->regmap, ES8326_CLK_DIV_LRCK, 0xff); + es8326_disable_micbias(es8326->component); msleep(200); regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9); -- 2.17.1

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[PATCH] ASoC: codecs: lpass-wsa-macro: Do not hard-code dai in VI mixer
by Krzysztof Kozlowski 29 Jul '24

29 Jul '24

The wsa_macro_vi_feed_mixer_put() callback for setting VI feedback mixer value could be used for different DAIs (planned in the future CPS DAI), so make the code a bit more generic by using DAI ID from widget->shift, instead of hard-coding it. The get() callback already follows such convention. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/lpass-wsa-macro.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index 8351f0974e6a..4158657914fb 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -2296,36 +2296,37 @@ static int wsa_macro_vi_feed_mixer_put(struct snd_kcontrol *kcontrol, struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); u32 enable = ucontrol->value.integer.value[0]; u32 spk_tx_id = mixer->shift; + u32 dai_id = widget->shift; if (enable) { if (spk_tx_id == WSA_MACRO_TX0 && !test_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { set_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]++; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]++; } if (spk_tx_id == WSA_MACRO_TX1 && !test_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { set_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]++; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]++; } } else { if (spk_tx_id == WSA_MACRO_TX0 && test_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { clear_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]--; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]--; } if (spk_tx_id == WSA_MACRO_TX1 && test_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { clear_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]--; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]--; } } snd_soc_dapm_mixer_update_power(widget->dapm, kcontrol, enable, NULL); -- 2.43.0

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[PATCH] ASoC: cs42l42: Convert comma to semicolon
by Chen Ni 29 Jul '24

29 Jul '24

Replace a comma between expression statements by a semicolon. Fixes: 90f6a2a20bd2 ("ASoC: cs42l42: Add SoundWire support") Signed-off-by: Chen Ni <nichen(a)iscas.ac.cn> --- sound/soc/codecs/cs42l42-sdw.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs42l42-sdw.c b/sound/soc/codecs/cs42l42-sdw.c index 94a66a325303..29891c1f6bec 100644 --- a/sound/soc/codecs/cs42l42-sdw.c +++ b/sound/soc/codecs/cs42l42-sdw.c @@ -323,15 +323,15 @@ static int cs42l42_sdw_read_prop(struct sdw_slave *peripheral) prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; /* DP1 - capture */ - ports[0].num = CS42L42_SDW_CAPTURE_PORT, - ports[0].type = SDW_DPN_FULL, - ports[0].ch_prep_timeout = 10, + ports[0].num = CS42L42_SDW_CAPTURE_PORT; + ports[0].type = SDW_DPN_FULL; + ports[0].ch_prep_timeout = 10; prop->src_dpn_prop = &ports[0]; /* DP2 - playback */ - ports[1].num = CS42L42_SDW_PLAYBACK_PORT, - ports[1].type = SDW_DPN_FULL, - ports[1].ch_prep_timeout = 10, + ports[1].num = CS42L42_SDW_PLAYBACK_PORT; + ports[1].type = SDW_DPN_FULL; + ports[1].ch_prep_timeout = 10; prop->sink_dpn_prop = &ports[1]; return 0; -- 2.25.1

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[PATCH 0/4] ASoC: codecs: wsa88xx: Few cleanups
by Krzysztof Kozlowski 29 Jul '24

29 Jul '24

Hi, Few cleanups around wsa88xx codecs. Best regards, Krzysztof --- Krzysztof Kozlowski (4): ASoC: codecs: wsa881x: Drop unused version readout ASoC: codecs: wsa883x: Handle reading version failure ASoC: codecs: wsa883x: Simplify handling variant/version ASoC: codecs: wsa884x: Simplify handling variant sound/soc/codecs/wsa881x.c | 2 -- sound/soc/codecs/wsa883x.c | 33 ++++++++++++++++++--------------- sound/soc/codecs/wsa884x.c | 5 ++--- 3 files changed, 20 insertions(+), 20 deletions(-) --- base-commit: 0b58e108042b0ed28a71cd7edf5175999955b233 change-id: 20240710-asoc-wsa88xx-version-dd0039265245 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

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Re: [PATCH v2 0/2] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by edmund.raile 29 Jul '24

29 Jul '24

> Thank you for your sending the revised patches, it looks better than the > previous one. However, I have an additional request. Allright, patch v3 it is. > [1] https://git-scm.com/docs/git-revert Should have known git has something like that, how handy! > $ git revert -s b5b519965c4c Yes, 5b5 can be removed via revert, but what is the difference in effect? Just time saving? > $ git revert -s 7ba5ca32fe6e This one I'd like to ask you about: The original inline comment in amdtp-stream.c amdtp_domain_stream_pcm_pointer() ``` // This function is called in software IRQ context of // period_work or process context. // // When the software IRQ context was scheduled by software IRQ // context of IT contexts, queued packets were already handled. // Therefore, no need to flush the queue in buffer furthermore. // // When the process context reach here, some packets will be // already queued in the buffer. These packets should be handled // immediately to keep better granularity of PCM pointer. // // Later, the process context will sometimes schedules software // IRQ context of the period_work. Then, no need to flush the // queue by the same reason as described in the above ``` (let's call the above v1) was replaced with ``` // In software IRQ context, the call causes dead-lock to disable the tasklet // synchronously. ``` on occasion of 7ba5ca32fe6e (let's call this v2). I sought to replace it with ``` // use wq to prevent deadlock between process context spin_lock // of snd_pcm_stream_lock_irq() in snd_pcm_status64() and // softIRQ context spin_lock of snd_pcm_stream_lock_irqsave() // in snd_pcm_period_elapsed() ``` to prevent this issue from occurring again (let's call this v3). Should I include v1, v3 or a combination of v1 and v3 in my next patch? > Just for safe, it is preferable to execute 'scripts/checkpatch.pl' in > kernel tree to check the patchset generated by send-email subcommand[3]. Absolutely should have done so, sorry. Thank you for your patience and guidance, Edmund Raile.

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[PATCH AUTOSEL 5.15 09/13] ASoC: Intel: sof_sdw: fix jack detection on ADL-N variant RVP
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 65c90df918205bc84f5448550cde76a54dae5f52 ] Experimental tests show that JD2_100K is required, otherwise the jack is detected always even with nothing plugged-in. To avoid matching with other known quirks the SKU information is used. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-2-pierre-louis.bossart@linux.… 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 | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 089b6c7994f9a..cfa0c3eaffea7 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -258,6 +258,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_SKU, "0000000000070000"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2_100K), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

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[PATCH AUTOSEL 6.1 10/15] ASoC: Intel: sof_sdw: add quirk for Dell SKU 0B8C
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 92d5b5930e7d55ca07b483490d6298eee828bbe4 ] Jack detection needs to rely on JD2, as most other Dell AlderLake-based devices. Closes: https://github.com/thesofproject/linux/issues/5021 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-4-pierre-louis.bossart@linux.… 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 | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index cf501baf6f143..9cd85ab19c553 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -387,6 +387,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { RT711_JD2 | SOF_SDW_FOUR_SPK), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B8C"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

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