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

[PATCH v2] ALSA: hda/realtek: Enable headset mic on IdeaPad 330-17IKB 81DM
by Ajrat Makhmutov 25 Jun '24

25 Jun '24

Headset microphone do not work out of the box with this laptop. This quirk fixes it. Zihao Wang specified the wrong subsystem id in his patch. Link: https://lore.kernel.org/all/20220424084120.74125-1-wzhd@ustc.edu/ Fixes: 3b79954fd00d ("ALSA: hda/realtek: Add quirk for Yoga Duet 7 13ITL6 speakers") Signed-off-by: Ajrat Makhmutov <rauty(a)altlinux.org> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index aa76d1c88589..5ec95a7903b8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10502,7 +10502,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940 / Yoga Duet 7", ALC298_FIXUP_LENOVO_C940_DUET7), SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), - SND_PCI_QUIRK(0x17aa, 0x3820, "Yoga Duet 7 13ITL6", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3820, "IdeaPad 330-17IKB 81DM", ALC269_FIXUP_ASPIRE_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3824, "Legion Y9000X 2020", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3834, "Lenovo IdeaPad Slim 9i 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), -- 2.33.8

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[PATCH 0/4] ASoC: Intel: boards: updates for 6.11 - part2
by Pierre-Louis Bossart 25 Jun '24

25 Jun '24

Minor additions and cleanups this time. Bard Liao (1): ASoC: Intel: sof_sdw: select PINCTRL_CS42L43 and SPI_CS42L43 Brent Lu (1): ASoC: Intel: maxim-common: add max_98373_get_tx_mask function Pierre-Louis Bossart (2): ASoC: Intel: sof_sdw: fix jack detection on ADL-N variant RVP ASoC: Intel: sof_sdw: add quirk for Dell SKU 0B8C sound/soc/intel/boards/Kconfig | 2 + sound/soc/intel/boards/sof_maxim_common.c | 56 +++++++++++++++++------ sound/soc/intel/boards/sof_sdw.c | 18 ++++++++ 3 files changed, 63 insertions(+), 13 deletions(-) -- 2.43.0

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[PATCH v1 2/4] ASoC: codecs: ES8326: codec can't enter suspend issue
by Zhang Yi 25 Jun '24

25 Jun '24

> > > Should mark the widgets with snd_soc_dapm_ignore_suspend() to allow the device to suspend with the widgets enabled. > > Thanks for your advice. But unfortunately codec didn't enter suspend after marking > > the widgets with snd_soc_dapm_ignore_suspend(). We will modify the trigger conditions > > for enable_micbias and disable_micbias to solve this issue > Really? We should fix that - that's the use case for this feature. > What goes wrong? We mark "MICBIAS1" and "MICBIAS2" with snd_soc_dapm_ignore_suspend() at es8326_enable_micbias(), and make "MICBIAS1" "MICBIAS2" active with snd_soc_dapm_force_enable_pin_unlocked(). But when the device enter suspend, the codec can't enter suspend.

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[PATCH 2/4] ASoC: codecs: ES8326: codec can't enter suspend issue
by Zhang Yi 25 Jun '24

25 Jun '24

> > When widgets "MICBIAS1" and "MICBIAS2" are active, the codec cannot > > enter suspend mode. So we removed these two widgets.We replaced > > enable_micbias and disable_micbias with regmap_update_bits to make > > sure the codec can enter suspend mode. > > Should mark the widgets with snd_soc_dapm_ignore_suspend() to allow the device to suspend with the widgets enabled. Thanks for your advice. But unfortunately codec didn't enter suspend after marking the widgets with snd_soc_dapm_ignore_suspend(). We will modify the trigger conditions for enable_micbias and disable_micbias to solve this issue

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[PATCH 0/3] ASoC: SOF: topology/intel improvements
by Pierre-Louis Bossart 25 Jun '24

25 Jun '24

Two updates for corner cases seen with new topologies, and one correction for the number of periods reported by Chrome stress tests. Peter Ujfalusi (3): ASoC: SOF: ipc4-topology: Use single token list for the copiers ASoC: SOF: ipc4-topology: Use correct queue_id for requesting input pin format ASoC: SOF: Intel: hda-pcm: Limit the maximum number of periods by MAX_BDL_ENTRIES sound/soc/sof/intel/hda-pcm.c | 6 ++++ sound/soc/sof/ipc4-topology.c | 57 ++++++++++++++++------------------- 2 files changed, 32 insertions(+), 31 deletions(-) -- 2.43.0

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[PATCH v1] ASoc: tas2781: Add name_prefix as the prefix name of firmwares and kcontrol to support corresponding TAS2563/TAS2781s
by Shenghao Ding 25 Jun '24

25 Jun '24

Add name_prefix as the prefix name of firmwares and kcontrol to support corresponding TAS2563/TAS2781s. name_prefix is not mandatory. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Changed the copyright year to 2024 in tas2781-comlib.c. - Correct the filename in the header comments of tas2781-comlib.c, remove the wrong file name. - Add name_prefix as name of firmwares and kcontrol. --- include/sound/tas2781.h | 1 + sound/soc/codecs/tas2781-comlib.c | 13 +++++++++---- sound/soc/codecs/tas2781-i2c.c | 1 + 3 files changed, 11 insertions(+), 4 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 99ca3e401fd1..cd8ce522b78e 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -108,6 +108,7 @@ struct tasdevice_priv { unsigned char coef_binaryname[64]; unsigned char rca_binaryname[64]; unsigned char dev_name[32]; + const char *name_prefix; unsigned char ndev; unsigned int magic_num; unsigned int chip_id; diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 3aa81514dad7..6db1a260da82 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -1,8 +1,8 @@ // SPDX-License-Identifier: GPL-2.0 // -// tas2781-lib.c -- TAS2781 Common functions for HDA and ASoC Audio drivers +// TAS2781 Common functions for HDA and ASoC Audio drivers // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> @@ -277,8 +277,13 @@ int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, */ mutex_lock(&tas_priv->codec_lock); - scnprintf(tas_priv->rca_binaryname, 64, "%sRCA%d.bin", - tas_priv->dev_name, tas_priv->ndev); + if (tas_priv->name_prefix) + scnprintf(tas_priv->rca_binaryname, 64, "%s-%sRCA%d.bin", + tas_priv->name_prefix, tas_priv->dev_name, + tas_priv->ndev); + else + scnprintf(tas_priv->rca_binaryname, 64, "%sRCA%d.bin", + tas_priv->dev_name, tas_priv->ndev); crc8_populate_msb(tas_priv->crc8_lkp_tbl, TASDEVICE_CRC8_POLYNOMIAL); tas_priv->codec = codec; ret = request_firmware_nowait(module, FW_ACTION_UEVENT, diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index c64d458e524e..4d1a0d836e77 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -579,6 +579,7 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) { struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + tas_priv->name_prefix = codec->name_prefix; return tascodec_init(tas_priv, codec, THIS_MODULE, tasdevice_fw_ready); } -- 2.34.1

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[RESEND PATCH] ALSA: dmaengine_pcm: terminate dmaengine before synchronize
by Shengjiu Wang 25 Jun '24

25 Jun '24

When dmaengine supports pause function, in suspend state, dmaengine_pause() is called instead of dmaengine_terminate_async(), In end of playback stream, the runtime->state will go to SNDRV_PCM_STATE_DRAINING, if system suspend & resume happen at this time, application will not resume playback stream, the stream will be closed directly, the dmaengine_terminate_async() will not be called before the dmaengine_synchronize(), which violates the call sequence for dmaengine_synchronize(). This behavior also happens for capture streams, but there is no SNDRV_PCM_STATE_DRAINING state for capture. So use dmaengine_tx_status() to check the DMA status if the status is DMA_PAUSED, then call dmaengine_terminate_async() to terminate dmaengine before dmaengine_synchronize(). Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in resend: - update the commit header sound/core/pcm_dmaengine.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index ed07fa5693d2..cc5db93b9132 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -368,6 +368,12 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_sync_stop); int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct dma_tx_state state; + enum dma_status status; + + status = dmaengine_tx_status(prtd->dma_chan, prtd->cookie, &state); + if (status == DMA_PAUSED) + dmaengine_terminate_async(prtd->dma_chan); dmaengine_synchronize(prtd->dma_chan); kfree(prtd); @@ -388,6 +394,12 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close); int snd_dmaengine_pcm_close_release_chan(struct snd_pcm_substream *substream) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct dma_tx_state state; + enum dma_status status; + + status = dmaengine_tx_status(prtd->dma_chan, prtd->cookie, &state); + if (status == DMA_PAUSED) + dmaengine_terminate_async(prtd->dma_chan); dmaengine_synchronize(prtd->dma_chan); dma_release_channel(prtd->dma_chan); -- 2.34.1

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[PATCH v1 0/4] ASoC: codecs: ES8326: Solving headphone detection and
by Zhang Yi 25 Jun '24

25 Jun '24

We propose four patches to solve headphone detection and suspend issues. And there are several registers that should be read-only registers. So we create es8326_writeable_register, and set these registers to false. Zhang Yi (4): ASoC: codecs: ES8326: Slove headphone detection issue ASoC: codecs: ES8326: codec can't enter suspend issue ASoC: codecs: ES8326: Minimize the pop noise ASoC: codecs: ES8326: regcache_sync error issue sound/soc/codecs/es8326.c | 62 +++++++++++++++++++++++++++++++-------- 1 file changed, 50 insertions(+), 12 deletions(-) -- 2.17.1

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[PATCH alsa-ucm-conf 1/2] sof-soundwire: Add missing match for cs42l43 speakers
by Charles Keepax 24 Jun '24

24 Jun '24

The cs42l43 reports as cs42l43-spk in the speaker case, the regex in sof-soundwire needs updated to recognise that. Fixes: 035d9206cffd ("sof-soundwire: Add basic support for cs42l43's speaker") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- ucm2/sof-soundwire/sof-soundwire.conf | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ucm2/sof-soundwire/sof-soundwire.conf b/ucm2/sof-soundwire/sof-soundwire.conf index 31492e3..2f61d11 100644 --- a/ucm2/sof-soundwire/sof-soundwire.conf +++ b/ucm2/sof-soundwire/sof-soundwire.conf @@ -21,7 +21,7 @@ Define { DefineRegex { SpeakerCodec { - Regex " spk:([a-z0-9]+(-sdca)?)" + Regex " spk:([a-z0-9]+((-sdca)|(-spk))?)" String "${CardComponents}" } SpeakerChannels { -- 2.39.2

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[PATCH v2 0/2] ASoC: codecs: lpass-wsa: fix vi capture setup
by Srinivas Kandagatla 24 Jun '24

24 Jun '24

This two patches fixes issues with VI capture rate and path setup. Changes since v1: - added missing break statements in switch case. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Srinivas Kandagatla (2): ASoC: codecs:lpass-wsa-macro: Fix vi feedback rate ASoC: codecs:lpass-wsa-macro: Fix logic of enabling vi channels sound/soc/codecs/lpass-wsa-macro.c | 103 ++++++++++++++++++++++++++++--------- 1 file changed, 80 insertions(+), 23 deletions(-) --- base-commit: 8aae76179d6b4587368d50614e114b391cf33ec9 change-id: 20240619-lpass-wsa-vi-0336f21092be Best regards, -- Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>

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