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

[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

1 0

[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

2 7

[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

2 1

[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

3 4

[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

2 5

[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

2 3

[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>

2 6

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

24 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 | 92 ++++++++++++++++++++++----------------- 1 file changed, 52 insertions(+), 40 deletions(-) -- 2.17.1

2 5

[PATCHv5 7/9] ASoC: dt-bindings: imx-audio-spdif: remove binding
by Elinor Montmasson 24 Jun '24

24 Jun '24

imx-audio-spdif was merged into the fsl-asoc-card driver, and therefore removed. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- .../bindings/sound/fsl,imx-audio-spdif.yaml | 66 ------------------- 1 file changed, 66 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml deleted file mode 100644 index 5fc543d02ecb..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml +++ /dev/null @@ -1,66 +0,0 @@ -# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) -%YAML 1.2 ---- -$id: http://devicetree.org/schemas/sound/fsl,imx-audio-spdif.yaml# -$schema: http://devicetree.org/meta-schemas/core.yaml# - -title: Freescale i.MX audio complex with S/PDIF transceiver - -maintainers: - - Shengjiu Wang <shengjiu.wang(a)nxp.com> - -properties: - compatible: - oneOf: - - items: - - enum: - - fsl,imx-sabreauto-spdif - - fsl,imx6sx-sdb-spdif - - const: fsl,imx-audio-spdif - - enum: - - fsl,imx-audio-spdif - - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - - spdif-controller: - $ref: /schemas/types.yaml#/definitions/phandle - description: The phandle of the i.MX S/PDIF controller - - spdif-out: - type: boolean - description: - If present, the transmitting function of S/PDIF will be enabled, - indicating there's a physical S/PDIF out connector or jack on the - board or it's connecting to some other IP block, such as an HDMI - encoder or display-controller. - - spdif-in: - type: boolean - description: - If present, the receiving function of S/PDIF will be enabled, - indicating there is a physical S/PDIF in connector/jack on the board. - -required: - - compatible - - model - - spdif-controller - -anyOf: - - required: - - spdif-in - - required: - - spdif-out - -additionalProperties: false - -examples: - - | - sound-spdif { - compatible = "fsl,imx-audio-spdif"; - model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; - spdif-in; - }; -- 2.34.1

2 5

[PATCHv5 6/9] ASoC: dt-bindings: fsl-asoc-card: add compatible string for spdif
by Elinor Montmasson 24 Jun '24

24 Jun '24

The S/PDIF audio card support was merged from imx-spdif into the fsl-asoc-card driver, making it possible to use an S/PDIF with an ASRC. Add the new compatible and update properties. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- .../bindings/sound/fsl-asoc-card.yaml | 30 ++++++++++++++++--- 1 file changed, 26 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml index 9922664d5ccc..f2e28b32808e 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml @@ -33,6 +33,7 @@ properties: - items: - enum: - fsl,imx-sgtl5000 + - fsl,imx-sabreauto-spdif - fsl,imx25-pdk-sgtl5000 - fsl,imx53-cpuvo-sgtl5000 - fsl,imx51-babbage-sgtl5000 @@ -54,6 +55,7 @@ properties: - fsl,imx6q-ventana-sgtl5000 - fsl,imx6sl-evk-wm8962 - fsl,imx6sx-sdb-mqs + - fsl,imx6sx-sdb-spdif - fsl,imx6sx-sdb-wm8962 - fsl,imx7d-evk-wm8960 - karo,tx53-audio-sgtl5000 @@ -65,6 +67,7 @@ properties: - fsl,imx-audio-sgtl5000 - fsl,imx-audio-wm8960 - fsl,imx-audio-wm8962 + - fsl,imx-audio-spdif - items: - enum: - fsl,imx-audio-ac97 @@ -81,6 +84,7 @@ properties: - fsl,imx-audio-wm8960 - fsl,imx-audio-wm8962 - fsl,imx-audio-wm8958 + - fsl,imx-audio-spdif model: $ref: /schemas/types.yaml#/definitions/string @@ -93,8 +97,15 @@ properties: need to add ASRC support via DPCM. audio-codec: - $ref: /schemas/types.yaml#/definitions/phandle - description: The phandle of an audio codec + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + The phandle of an audio codec. + With "fsl,imx-audio-spdif", either SPDIF audio codec spdif_transmitter, + spdif_receiver or both. + minItems: 1 + maxItems: 2 + items: + maxItems: 1 audio-cpu: $ref: /schemas/types.yaml#/definitions/phandle @@ -150,8 +161,10 @@ properties: description: dai-link uses bit clock inversion. mclk-id: - $ref: /schemas/types.yaml#/definitions/uint32 - description: main clock id, specific for each card configuration. + $ref: /schemas/types.yaml#/definitions/uint32-array + description: Main clock id for each codec, specific for each card configuration. + minItems: 1 + maxItems: 2 mux-int-port: $ref: /schemas/types.yaml#/definitions/uint32 @@ -195,3 +208,12 @@ examples: "AIN2L", "Line In Jack", "AIN2R", "Line In Jack"; }; + + - | + sound-spdif-asrc { + compatible = "fsl,imx-audio-spdif"; + model = "spdif-asrc-audio"; + audio-cpu = <&spdif>; + audio-asrc = <&easrc>; + audio-codec = <&spdifdit>, <&spdifdir>; + }; -- 2.34.1

2 3