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[PATCH] ASoC: codecs: lpass-rx-macro: add missing handling of v2.1 codec
by Krzysztof Kozlowski 25 Jun '24

25 Jun '24

We have also v2.1 version of the codec (see 'enum lpass_codec_version'), so handle it as well in all switch cases. Fixes: dbacef05898d ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/lpass-rx-macro.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 00ecc470ba8b..1c3429f004ed 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -1601,6 +1601,7 @@ static bool rx_is_rw_register(struct device *dev, unsigned int reg) case LPASS_CODEC_VERSION_1_1: case LPASS_CODEC_VERSION_1_2: case LPASS_CODEC_VERSION_2_0: + case LPASS_CODEC_VERSION_2_1: return rx_pre_2_5_is_rw_register(dev, reg); case LPASS_CODEC_VERSION_2_5: case LPASS_CODEC_VERSION_2_6: @@ -3639,6 +3640,7 @@ static int rx_macro_component_probe(struct snd_soc_component *component) case LPASS_CODEC_VERSION_1_1: case LPASS_CODEC_VERSION_1_2: case LPASS_CODEC_VERSION_2_0: + case LPASS_CODEC_VERSION_2_1: controls = rx_macro_def_snd_controls; num_controls = ARRAY_SIZE(rx_macro_def_snd_controls); widgets = rx_macro_def_dapm_widgets; @@ -3812,6 +3814,7 @@ static int rx_macro_probe(struct platform_device *pdev) case LPASS_CODEC_VERSION_1_1: case LPASS_CODEC_VERSION_1_2: case LPASS_CODEC_VERSION_2_0: + case LPASS_CODEC_VERSION_2_1: rx->rxn_reg_stride = 0x80; def_count = ARRAY_SIZE(rx_defaults) + ARRAY_SIZE(rx_pre_2_5_defaults); reg_defaults = kmalloc_array(def_count, sizeof(struct reg_default), GFP_KERNEL); -- 2.43.0

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Virtual nodes created by UCM rules have the exact same nick, making them indistinguishable in KDE's audio applet
by GitHub issues - opened 25 Jun '24

25 Jun '24

alsa-project/alsa-ucm-conf issue #426 was opened from z411: Hello, Recently KDE Plasma's applet started using the node's nick to show the name of the device. The problem is that for audio interfaces that separate their inputs into virtual interfaces, they end up with something like this: ``` alsa.card_name = "Volt 2" device.profile.description = "Mono Input 1" node.description = "Volt 2 Mono Input 1" node.nick = "Volt 2" ``` ``` alsa.card_name = "Volt 2" device.profile.description = "Mono Input 2" node.description = "Volt 2 Mono Input 2" node.nick = "Volt 2" ``` Since they use the same nick, the end result is that we end up with an input list that looks like this: ![image](https://github.com/alsa-project/alsa-ucm-conf/assets/1988512/34e794ce-07ce-45b8-8be4-073dad24d339) Could the node nick be changed to reflect that this is a virtual device that refers to a different input from the same device? We already have an issue about this open in PipeWire and KDE issue trackers but we're not sure where this should be really fixed. Thanks. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/426 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[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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