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

[PATCH v3 0/2] Fixes for McASP and dmaengine_pcm
by Jai Luthra 12 Jun '24

12 Jun '24

This series fixes two patches: 1. Fix the dmaengine API usage by calling dmaengine_synchronize() after dmaengine_terminate_async() when xrun events occur in application 2. Use the McASP AFIFO property from DT to refine the period size, instead of hardcoding minimum to 64 samples Signed-off-by: Jai Luthra <j-luthra(a)ti.com> --- Changes in v3: - Use sync_stop() hook instead of a prepare() hook for the DMA channel synchronization - Link to v2: https://lore.kernel.org/r/20240610-asoc_next-v2-0-b52aaf5d67c4@ti.com Changes in v2: - Fix compiler warning for prepare callback by marking it static - Pass numevt directly to hw_rule_min_periodsize() - Link to v1: https://lore.kernel.org/r/20240604-asoc_next-v1-0-e895c88e744d@ti.com --- Jai Luthra (2): ALSA: dmaengine: Synchronize dma channel after drop() ASoC: ti: davinci-mcasp: Set min period size using FIFO config include/sound/dmaengine_pcm.h | 1 + sound/core/pcm_dmaengine.c | 10 ++++++++++ sound/soc/soc-generic-dmaengine-pcm.c | 8 ++++++++ sound/soc/ti/davinci-mcasp.c | 9 +++++++-- 4 files changed, 26 insertions(+), 2 deletions(-) --- base-commit: a957267fa7e9159d3d2ee1421359ebf228570c68 change-id: 20240604-asoc_next-c063fcc190c6 Best regards, -- Jai Luthra <j-luthra(a)ti.com>

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[PATCH v2] ASoC: ti: omap-hdmi: Fix too long driver name
by Primoz Fiser 12 Jun '24

12 Jun '24

Set driver name to "HDMI". This simplifies the code and gets rid of the following error messages: ASoC: driver name too long 'HDMI 58040000.encoder' -> 'HDMI_58040000_e' Signed-off-by: Primoz Fiser <primoz.fiser(a)norik.com> --- Changes in v2: - use "HDMI" instead of DRV_NAME (requested) sound/soc/ti/omap-hdmi.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c index 639bc83f4263..cf43ac19c4a6 100644 --- a/sound/soc/ti/omap-hdmi.c +++ b/sound/soc/ti/omap-hdmi.c @@ -354,11 +354,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) if (!card) return -ENOMEM; - card->name = devm_kasprintf(dev, GFP_KERNEL, - "HDMI %s", dev_name(ad->dssdev)); - if (!card->name) - return -ENOMEM; - + card->name = "HDMI"; card->owner = THIS_MODULE; card->dai_link = devm_kzalloc(dev, sizeof(*(card->dai_link)), GFP_KERNEL); -- 2.25.1

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[PATCH] ASoC: cs35l56: Disconnect ASP1 TX sources when ASP1 DAI is hooked up
by Richard Fitzgerald 12 Jun '24

12 Jun '24

If the ASP1 DAI is hooked up by the machine driver the ASP TX mixer sources should be initialized to disconnected. The silicon default is for the mixer source registers to default to a collection of monitoring sources. The problem with this is that it causes the DAPM graph to initialize with the capture path connected to a valid source widget, even though nothing setup a path. When the ASP DAI is connected as a codec-to-codec link this will cause the other codec to power-up even though nothing is using it. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: dfd2ffb37399 ("ASoC: cs35l56: Prevent overwriting firmware ASP config") --- sound/soc/codecs/cs35l56-shared.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index 8af89a263594..30497152e02a 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -215,6 +215,10 @@ static const struct reg_sequence cs35l56_asp1_defaults[] = { REG_SEQ0(CS35L56_ASP1_FRAME_CONTROL5, 0x00020100), REG_SEQ0(CS35L56_ASP1_DATA_CONTROL1, 0x00000018), REG_SEQ0(CS35L56_ASP1_DATA_CONTROL5, 0x00000018), + REG_SEQ0(CS35L56_ASP1TX1_INPUT, 0x00000000), + REG_SEQ0(CS35L56_ASP1TX2_INPUT, 0x00000000), + REG_SEQ0(CS35L56_ASP1TX3_INPUT, 0x00000000), + REG_SEQ0(CS35L56_ASP1TX4_INPUT, 0x00000000), }; /* -- 2.39.2

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[RESEND PATCH v4] ASoc: tas2781: Enable RCA-based playback without DSP firmware download
by Shenghao Ding 12 Jun '24

12 Jun '24

<<< No Message Collected >>>

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[RESEND PATCH v4] ASoc: tas2781: Enable RCA-based playback without DSP firmware download
by Shenghao Ding 12 Jun '24

12 Jun '24

In only RCA (Reconfigurable Architecture) binary case, no DSP program will be working inside tas2563/tas2781, that is dsp-bypass mode, do not support speaker protection, and audio acoustic algorithms in this mode. Fixes: ef3bcde75d06 ("ASoC: tas2781: Add tas2781 driver") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v4: - add a description of what the state machine looks like. - Remove TASDEVICE_DSP_FW_NONE because of unused. - Remove stray change. - Fix broken indentation. v3: - Add description on RCA is Reconfigurable Architecture. - Add the description on enabling - Reword the commit - Remove question mark in the comments. - Add spaces in comments. v2: - Correct comment. - Add Fixes. - Move header file to the first. v1: - Split out the different logical changes into different patches. - rename tasdevice_dsp_fw_state -> tasdevice_fw_state, the fw are not only DSP fw, but also RCA(Reconfigurable data, such as acoustic data and register setting, etc). - Add TASDEVICE_RCA_FW_OK in tasdevice_fw_state to identify the state that only RCA binary file has been download successfully, but DSP fw is not loaded or loading failure. - Add the this strategy into tasdevice_tuning_switch. - If one side of the if/else has a braces both should in tasdevice_tuning_switch. - Identify whehter both RCA and DSP have been loaded or only RCA has been loaded in tasdevice_fw_ready. - Add check fw load status in tasdevice_startup. - remove ret in tasdevice_startup to make the code neater. --- include/sound/tas2781-dsp.h | 11 ++++++++-- sound/soc/codecs/tas2781-fmwlib.c | 18 +++++++++++----- sound/soc/codecs/tas2781-i2c.c | 34 +++++++++++++++++++------------ 3 files changed, 43 insertions(+), 20 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 7fba7ea26a4b..3cda9da14f6d 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -117,10 +117,17 @@ struct tasdevice_fw { struct device *dev; }; -enum tasdevice_dsp_fw_state { - TASDEVICE_DSP_FW_NONE = 0, +enum tasdevice_fw_state { + /* Driver in startup mode, not load any firmware. */ TASDEVICE_DSP_FW_PENDING, + /* DSP firmware in the system, but parsing error. */ TASDEVICE_DSP_FW_FAIL, + /* + * Only RCA (Reconfigurable Architecture) firmware load + * successfully. + */ + TASDEVICE_RCA_FW_OK, + /* Both RCA and DSP firmware load successfully. */ TASDEVICE_DSP_FW_ALL_OK, }; diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 265a8ca25cbb..838d29fead96 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2324,14 +2324,21 @@ void tasdevice_tuning_switch(void *context, int state) struct tasdevice_fw *tas_fmw = tas_priv->fmw; int profile_cfg_id = tas_priv->rcabin.profile_cfg_id; - if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { - dev_err(tas_priv->dev, "DSP bin file not loaded\n"); + /* + * Only RCA-based Playback can still work with no dsp program running + * inside the chip. + */ + switch (tas_priv->fw_state) { + case TASDEVICE_RCA_FW_OK: + case TASDEVICE_DSP_FW_ALL_OK: + break; + default: return; } if (state == 0) { - if (tas_priv->cur_prog < tas_fmw->nr_programs) { - /*dsp mode or tuning mode*/ + if (tas_fmw && tas_priv->cur_prog < tas_fmw->nr_programs) { + /* dsp mode or tuning mode */ profile_cfg_id = tas_priv->rcabin.profile_cfg_id; tasdevice_select_tuningprm_cfg(tas_priv, tas_priv->cur_prog, tas_priv->cur_conf, @@ -2340,9 +2347,10 @@ void tasdevice_tuning_switch(void *context, int state) tasdevice_select_cfg_blk(tas_priv, profile_cfg_id, TASDEVICE_BIN_BLK_PRE_POWER_UP); - } else + } else { tasdevice_select_cfg_blk(tas_priv, profile_cfg_id, TASDEVICE_BIN_BLK_PRE_SHUTDOWN); + } } EXPORT_SYMBOL_NS_GPL(tasdevice_tuning_switch, SND_SOC_TAS2781_FMWLIB); diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9350972dfefe..9c3c89cb36de 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -380,23 +380,32 @@ static void tasdevice_fw_ready(const struct firmware *fmw, mutex_lock(&tas_priv->codec_lock); ret = tasdevice_rca_parser(tas_priv, fmw); - if (ret) + if (ret) { + tasdevice_config_info_remove(tas_priv); goto out; + } tasdevice_create_control(tas_priv); tasdevice_dsp_remove(tas_priv); tasdevice_calbin_remove(tas_priv); - tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + tas_priv->fw_state = TASDEVICE_RCA_FW_OK; scnprintf(tas_priv->coef_binaryname, 64, "%s_coef.bin", tas_priv->dev_name); ret = tasdevice_dsp_parser(tas_priv); if (ret) { dev_err(tas_priv->dev, "dspfw load %s error\n", tas_priv->coef_binaryname); - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; goto out; } - tasdevice_dsp_create_ctrls(tas_priv); + + /* + * If no dsp-related kcontrol created, the dsp resource will be freed. + */ + ret = tasdevice_dsp_create_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "dsp controls error\n"); + goto out; + } tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; @@ -417,9 +426,8 @@ static void tasdevice_fw_ready(const struct firmware *fmw, tasdevice_prmg_load(tas_priv, 0); tas_priv->cur_prog = 0; out: - if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { - /*If DSP FW fail, kcontrol won't be created */ - tasdevice_config_info_remove(tas_priv); + if (tas_priv->fw_state == TASDEVICE_RCA_FW_OK) { + /* If DSP FW fail, DSP kcontrol won't be created. */ tasdevice_dsp_remove(tas_priv); } mutex_unlock(&tas_priv->codec_lock); @@ -466,14 +474,14 @@ static int tasdevice_startup(struct snd_pcm_substream *substream, { struct snd_soc_component *codec = dai->component; struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); - int ret = 0; - if (tas_priv->fw_state != TASDEVICE_DSP_FW_ALL_OK) { - dev_err(tas_priv->dev, "DSP bin file not loaded\n"); - ret = -EINVAL; + switch (tas_priv->fw_state) { + case TASDEVICE_RCA_FW_OK: + case TASDEVICE_DSP_FW_ALL_OK: + return 0; + default: + return -EINVAL; } - - return ret; } static int tasdevice_hw_params(struct snd_pcm_substream *substream, -- 2.34.1

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[PATCH v5 0/7] ASoC: codecs: wcd937x: add wcd937x audio codec support
by Mohammad Rafi Shaik 11 Jun '24

11 Jun '24

This patchset adds support for Qualcomm WCD9370/WCD9375 codec. Qualcomm WCD9370/WCD9375 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 3 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 6 DMICs and MBHC. For codec driver to be functional it would need both tx and rx Soundwire devices to be up and this is taken care by using device component framework and device-links are used to ensure proper pm dependencies. Ex tx does not enter suspend before rx or codec is suspended. This patchset along with other SoundWire patches on the list have been tested on QCM6490 IDP device. Changes since v4: - Removed volatile/read-only registers from defaults list - Added wcd939x_volatile_register() with only volatile registers - Added a wcd939x_readable_register() with read-only and read-write registers, so cache does it's job - Fixed Spurious events for mixer controls and validated with mixer selftest tool - Used TLV instead of enum for ear_pa_gain mixer control - Used enum constraints instead of OneOf in dt-binding patch - Added vdd-px supply property as non optional in dt-binding patch - Reworked and done driver cleanup Changes since v3: - Fixed dt binding check errors. - Added constraints on values in v4-0001 binding patch as suggested by Krzysztof - Change the patch sequence soundwire driver first then codec driver - Added missing .remove soundwire driver function - Reworked and done driver cleanup Changes since v2: - Used common qcom,wcd93xx-common.yaml. removed duplicate properties. - Merged bindings patches "v2-0001" and "v2-0003" in single patch for easy review. - Fixed dt binding check errors. - Added missing "qcom,wcd9375-codec" in v3-0001 dt binding patch. - Added constraints on values in v3-0001 binding patch as suggested by Krzysztof - Fix the typo mistake in v2 cover letter Changes since v1: - Split the patch per driver for easier review as suggested by Krzysztof - Used devm_gpiod_get api to get reset gpio as suggested by Krzysztof Prasad Kumpatla (7): ASoC: dt-bindings: document wcd937x Audio Codec ASoC: codecs: wcd937x-sdw: add SoundWire driver ASoC: codecs: wcd937x: add wcd937x codec driver ASoC: codecs: wcd937x: add basic controls ASoC: codecs: wcd937x: add playback dapm widgets ASoC: codecs: wcd937x: add capture dapm widgets ASoC: codecs: wcd937x: add audio routing and Kconfig .../bindings/sound/qcom,wcd937x-sdw.yaml | 91 + .../bindings/sound/qcom,wcd937x.yaml | 82 + sound/soc/codecs/Kconfig | 20 + sound/soc/codecs/Makefile | 7 + sound/soc/codecs/wcd937x-sdw.c | 1139 +++++++ sound/soc/codecs/wcd937x.c | 3029 +++++++++++++++++ sound/soc/codecs/wcd937x.h | 653 ++++ 7 files changed, 5021 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd937x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml create mode 100644 sound/soc/codecs/wcd937x-sdw.c create mode 100644 sound/soc/codecs/wcd937x.c create mode 100644 sound/soc/codecs/wcd937x.h base-commit: 124cfbcd6d185d4f50be02d5f5afe61578916773 -- 2.25.1

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[PATCH] ASoC: samsung: midas_wm1811: Fix error code in probe()
by Dan Carpenter 11 Jun '24

11 Jun '24

This accidentally returns success instead of -EINVAL. Fixes: c91d0c2e198d ("ASoC: samsung: midas_wm1811: Add GPIO-based headset jack detection") Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/soc/samsung/midas_wm1811.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/midas_wm1811.c b/sound/soc/samsung/midas_wm1811.c index 3a269c7de169..bbfe5fef59af 100644 --- a/sound/soc/samsung/midas_wm1811.c +++ b/sound/soc/samsung/midas_wm1811.c @@ -623,7 +623,7 @@ static int midas_probe(struct platform_device *pdev) if (channel_type != IIO_VOLTAGE) { dev_err(dev, "ADC channel is not voltage\n"); - return ret; + return -EINVAL; } priv->gpio_headset_key = devm_gpiod_get(dev, "headset-key", -- 2.43.0

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[PATCH v2 0/2] Fixes for McASP and dmaengine_pcm
by Jai Luthra 11 Jun '24

11 Jun '24

This series fixes two patches: 1. Fix the dmaengine API usage by calling dmaengine_synchronize() after dmaengine_terminate_async() when xrun events occur in application 2. Use the McASP AFIFO property from DT to refine the period size, instead of hardcoding minimum to 64 samples Signed-off-by: Jai Luthra <j-luthra(a)ti.com> --- Changes in v2: - Fix compiler warning for prepare callback by marking it static - Pass numevt directly to hw_rule_min_periodsize() - Link to v1: https://lore.kernel.org/r/20240604-asoc_next-v1-0-e895c88e744d@ti.com --- Jai Luthra (2): ALSA: dmaengine: Synchronize dma channel in prepare() ASoC: ti: davinci-mcasp: Set min period size using FIFO config include/sound/dmaengine_pcm.h | 1 + sound/core/pcm_dmaengine.c | 10 ++++++++++ sound/soc/soc-generic-dmaengine-pcm.c | 8 ++++++++ sound/soc/ti/davinci-mcasp.c | 9 +++++++-- 4 files changed, 26 insertions(+), 2 deletions(-) --- base-commit: d97496ca23a2d4ee80b7302849404859d9058bcd change-id: 20240604-asoc_next-c063fcc190c6 Best regards, -- Jai Luthra <j-luthra(a)ti.com>

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[PATCH] bytcr_rt5640 : inverse jack detect for Archos 101 cesium
by Thomas GENTY 11 Jun '24

11 Jun '24

When headphones are plugged in, they appear absent; when they are removed, they appear present. Add a specific entry in bytcr_rt5640 for this device Signed-off-by: Thomas GENTY <tomlohave(a)gmail.com> Reviewed-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index b41a1147f1c3..a64d1989e28a 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -610,6 +610,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 101 CESIUM"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), -- 2.45.2

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[PATCH 0/3] ACPI/ALSA/soundwire: add acpi_get_local_u64_address() helper
by Pierre-Louis Bossart 11 Jun '24

11 Jun '24

The acpi_get_local_address() helper assumes a 32-bit ADR is used. If we want to use this helper for SoundWire/SDCA ASoC codecs, we need an extension where the native 64-bits are used. This patchset suggests a new helper, acpi_get_local_address() may be renamed if desired in a folow-up patch. The path of least resistance would be to merge this patchset in the ASoC tree, since I have additional changes for ASoC/SDCA (SoundWire Device Class) that depend on the new helper. Pierre-Louis Bossart (3): ACPI: utils: introduce acpi_get_local_u64_address() soundwire: slave: simplify code with acpi_get_local_u64_address() ALSA: hda: intel-sdw-acpi: use acpi_get_local_u64_address() drivers/acpi/utils.c | 22 ++++++++++++++++------ drivers/soundwire/slave.c | 13 ++++--------- include/linux/acpi.h | 1 + sound/hda/intel-sdw-acpi.c | 6 +++--- 4 files changed, 24 insertions(+), 18 deletions(-) -- 2.43.0

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