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

[bug report] ASoC: codecs: lpass-macro: add helpers to get codec version
by Dan Carpenter 20 Jun '24

20 Jun '24

Hello Srinivas Kandagatla, Commit 378918d59181 ("ASoC: codecs: lpass-macro: add helpers to get codec version") from Jun 13, 2024 (linux-next), leads to the following Smatch static checker warning: sound/soc/codecs/lpass-va-macro.c:1485 va_macro_set_lpass_codec_version() error: uninitialized symbol 'version'. sound/soc/codecs/lpass-va-macro.c 1464 static void va_macro_set_lpass_codec_version(struct va_macro *va) 1465 { 1466 int core_id_0 = 0, core_id_1 = 0, core_id_2 = 0, version; We could just initialize this to zero to make the checker happy? The compiler is generally going to zero it anyway so it's free. 1467 1468 regmap_read(va->regmap, CDC_VA_TOP_CSR_CORE_ID_0, &core_id_0); 1469 regmap_read(va->regmap, CDC_VA_TOP_CSR_CORE_ID_1, &core_id_1); 1470 regmap_read(va->regmap, CDC_VA_TOP_CSR_CORE_ID_2, &core_id_2); 1471 1472 if ((core_id_0 == 0x01) && (core_id_1 == 0x0F)) 1473 version = LPASS_CODEC_VERSION_2_0; 1474 if ((core_id_0 == 0x02) && (core_id_1 == 0x0E)) 1475 version = LPASS_CODEC_VERSION_2_1; 1476 if ((core_id_0 == 0x02) && (core_id_1 == 0x0F) && (core_id_2 == 0x50 || core_id_2 == 0x51)) 1477 version = LPASS_CODEC_VERSION_2_5; 1478 if ((core_id_0 == 0x02) && (core_id_1 == 0x0F) && (core_id_2 == 0x60 || core_id_2 == 0x61)) 1479 version = LPASS_CODEC_VERSION_2_6; 1480 if ((core_id_0 == 0x02) && (core_id_1 == 0x0F) && (core_id_2 == 0x70 || core_id_2 == 0x71)) 1481 version = LPASS_CODEC_VERSION_2_7; 1482 if ((core_id_0 == 0x02) && (core_id_1 == 0x0F) && (core_id_2 == 0x80 || core_id_2 == 0x81)) 1483 version = LPASS_CODEC_VERSION_2_8; 1484 --> 1485 lpass_macro_set_codec_version(version); 1486 1487 dev_dbg(va->dev, "LPASS Codec Version %s\n", lpass_macro_get_codec_version_string(version)); 1488 } regards, dan carpenter

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[PATCH] ALSA: hda: cs35l56: Select SERIAL_MULTI_INSTANTIATE
by Simon Trimmer 20 Jun '24

20 Jun '24

The ACPI IDs used in the CS35L56 HDA drivers are all handled by the serial multi-instantiate driver which starts multiple Linux device instances from a single ACPI Device() node. As serial multi-instantiate is not an optional part of the system add it as a dependency in Kconfig so that it is not overlooked. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/pci/hda/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 1252632f0ffa..731f4e99bc75 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -162,6 +162,7 @@ config SND_HDA_SCODEC_CS35L56_I2C depends on ACPI || COMPILE_TEST depends on SND_SOC select FW_CS_DSP + select SERIAL_MULTI_INSTANTIATE select SND_HDA_GENERIC select SND_SOC_CS35L56_SHARED select SND_HDA_SCODEC_CS35L56 @@ -178,6 +179,7 @@ config SND_HDA_SCODEC_CS35L56_SPI depends on ACPI || COMPILE_TEST depends on SND_SOC select FW_CS_DSP + select SERIAL_MULTI_INSTANTIATE select SND_HDA_GENERIC select SND_SOC_CS35L56_SHARED select SND_HDA_SCODEC_CS35L56 -- 2.34.1

2 1

ALSA volume control ignored for "4c4a:4155 Jieli Technology USB Composite Device"
by GitHub issues - opened 20 Jun '24

20 Jun '24

alsa-project/alsa-lib issue #401 was opened from davidmcnabnz: I just purchased a generic USB audio interface, which shows up in `lsusb` as: ``` 4c4a:4155 Jieli Technology USB Composite Device ``` The device on the whole seems to _simply just work_. But the level control for the stereo capture seems to be completely ignored, both in `alsamixer` and `pavucontrol`. My setup is Debian GNU/Linux 12 (bookworm), standard apt including contrib. General stack is wireplumber and pipewire, implementing JACK via pw-jack and pipewire-pulse. This is working very well for my music work. Lastly, below is a paste of my currently installed packages relevant to ALSA and JACK. Grateful if anyone can advise. Cheers David ``` ii a2jmidid 9-3 amd64 Daemon for exposing legacy ALSA MIDI in JACK MIDI systems ii aj-snapshot 0.9.9-4 amd64 make snapshots of JACK connections ii alsa-tools 1.2.5-3 amd64 Console based ALSA utilities for specific hardware ii alsa-tools-gui 1.2.5-3 amd64 GUI based ALSA utilities for specific hardware ii alsa-topology-conf 1.2.5.1-2 all ALSA topology configuration files ii alsa-ucm-conf 1.2.8-1 all ALSA Use Case Manager configuration files ii alsa-utils 1.2.8-1 amd64 Utilities for configuring and using ALSA ii drumstick-data 2.7.2-1 all Qt5/C++ wrapper for ALSA Sequencer (data files) ii gir1.2-cvc-1.0:amd64 5.6.1-1 amd64 Introspection data for Cinnamon pulseaudio abstraction ii gnome-remote-desktop 43.3-1 amd64 Remote desktop daemon for GNOME using PipeWire ii gstreamer1.0-alsa:amd64 1.22.0-3+deb12u1 amd64 GStreamer plugin for ALSA ii gstreamer1.0-pipewire:amd64 0.3.65-3+deb12u1 amd64 GStreamer 1.0 plugin for the PipeWire multimedia server ii jack-midi-clock 0.4.3-1+b1 amd64 JACK client that sends MIDI beat clock pulses ii jack-mixer 17-2+b2 amd64 GTK+ JACK audio mixer application (Python 3) ii jack-tools 20131226-1.1 amd64 various JACK tools: dl, record, scope, osc, plumbing, udp, play, transport ii jackd 5+nmu1 all JACK Audio Connection Kit (default server package) ii jackd2 1.9.21~dfsg-3 amd64 JACK Audio Connection Kit (server and example clients) ii jackd2-firewire 1.9.21~dfsg-3 amd64 JACK Audio Connection Kit (FFADO backend) ii jackmeter 0.4-1+b2 amd64 a basic command line meter for the JACK audio system ii jacktrip 1.7.0+ds-1 amd64 high-quality system for audio network performances (cmdline) ii klick 0.12.2-4.1 amd64 advanced metronome for JACK ii kmetronome 1.3.1-1 amd64 ALSA MIDI Metronome ii libasound2:amd64 1.2.8-1+b1 amd64 shared library for ALSA applications ii libasound2:i386 1.2.8-1+b1 i386 shared library for ALSA applications ii libasound2-data 1.2.8-1 all Configuration files and profiles for ALSA drivers ii libasound2-dev:amd64 1.2.8-1+b1 amd64 shared library for ALSA applications -- development files ii libasound2-plugins:amd64 1.2.7.1-1 amd64 ALSA library additional plugins ii libasound2-plugins:i386 1.2.7.1-1 i386 ALSA library additional plugins ii libatopology2:amd64 1.2.8-1+b1 amd64 shared library for handling ALSA topology definitions ii libcanberra-pulse:amd64 0.30-10 amd64 PulseAudio backend for libcanberra ii libclalsadrv2 2.0.0-3.1 amd64 ALSA driver C++ access library ii libcvc0:amd64 5.6.1-1 amd64 Cinnamon pulseaudio abstraction library ii libdrumstick-alsa2:amd64 2.7.2-1 amd64 Qt5/C++ wrapper for ALSA Sequencer (shared library) ii libdrumstick-widgets2:amd64 2.7.2-1 amd64 Qt5/C++ wrapper for ALSA Sequencer (widgets) ii libjack-jackd2-0:amd64 1.9.21~dfsg-3 amd64 JACK Audio Connection Kit (libraries) ii libjack-jackd2-0:i386 1.9.21~dfsg-3 i386 JACK Audio Connection Kit (libraries) ii libjack-jackd2-dev:amd64 1.9.21~dfsg-3 amd64 JACK Audio Connection Kit (development files) ii libpipewire-0.3-0:amd64 0.3.65-3+deb12u1 amd64 libraries for the PipeWire multimedia server ii libpipewire-0.3-common 0.3.65-3+deb12u1 all libraries for the PipeWire multimedia server - common files ii libpipewire-0.3-modules:amd64 0.3.65-3+deb12u1 amd64 libraries for the PipeWire multimedia server - modules ii libpulse-mainloop-glib0:amd64 16.1+dfsg1-2+b1 amd64 PulseAudio client libraries (glib support) ii libpulse0:amd64 16.1+dfsg1-2+b1 amd64 PulseAudio client libraries ii libpulse0:i386 16.1+dfsg1-2+b1 i386 PulseAudio client libraries ii libpulsedsp:amd64 16.1+dfsg1-2+b1 amd64 PulseAudio OSS pre-load library ii libspa-0.2-bluetooth:amd64 0.3.65-3+deb12u1 amd64 libraries for the PipeWire multimedia server - bluetooth plugins ii libspa-0.2-modules:amd64 0.3.65-3+deb12u1 amd64 libraries for the PipeWire multimedia server Simple Plugin API - modules ii libzita-alsa-pcmi0:amd64 0.6.1-1 amd64 C++ wrapper around the ALSA API ii lsp-plugins-jack 1.2.5-1 amd64 LSP (Linux Studio Plugins) audio-plugins (standalone JACK clients) ii mudita24 1.0.3+svn13-7 amd64 ALSA GUI control tool for Envy24 (ice1712) soundcards ii multimedia-jack 0.10 all JACK (Jack Audio Connection Kit) and ALSA related packages ii pavucontrol 5.0-2 amd64 PulseAudio Volume Control ii pipewire:amd64 0.3.65-3+deb12u1 amd64 audio and video processing engine multimedia server ii pipewire-alsa:amd64 0.3.65-3+deb12u1 amd64 PipeWire ALSA plugin ii pipewire-audio 0.3.65-3+deb12u1 all recommended set of PipeWire packages for a standard audio desktop use ii pipewire-bin 0.3.65-3+deb12u1 amd64 PipeWire multimedia server - programs ii pipewire-jack:amd64 0.3.65-3+deb12u1 amd64 PipeWire JACK plugin ii pipewire-pulse 0.3.65-3+deb12u1 amd64 PipeWire PulseAudio daemon ii pulseaudio-utils 16.1+dfsg1-2+b1 amd64 Command line tools for the PulseAudio sound server ii qasconfig 0.23.0-1 amd64 ALSA configuration browser ii qashctl 0.23.0-1 amd64 mixer for ALSA's High level Control Interface ii qasmixer 0.23.0-1 amd64 ALSA mixer for the desktop ii qjackctl 0.9.9-1 amd64 User interface for controlling the JACK sound server ii silentjack 0.3-2+b2 amd64 silence detector for the JACK audio system ii wireplumber 0.4.13-1 amd64 modular session / policy manager for PipeWire ii xjadeo 0.8.12-1 amd64 Video player with JACK sync ii zita-ajbridge 0.8.4-1 amd64 alsa to jack bridge ii zita-alsa-pcmi-utils 0.6.1-1 amd64 Two simple demo programs zita-alsa-pcmi library ii zita-lrx 0.1.2-1 amd64 Command line jack application providing crossover filters ii zita-mu1 0.3.3-2 amd64 organise stereo monitoring for Jack Audio Connection Kit ii zita-njbridge 0.4.8-1 amd64 Jack clients to transmit multichannel audio over a local IP network ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/401 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v6 0/7] ASoC: codecs: wcd937x: add wcd937x audio codec support
by Mohammad Rafi Shaik 19 Jun '24

19 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 v5: - Remove the string compare in MIC BIAS widget settings as suggested by Srinivas Kandagatla - Fixed Unbalanced pm_runtime_enable! in wcd937x-sdw soundwire slave. 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 | 1136 +++++++ sound/soc/codecs/wcd937x.c | 3011 +++++++++++++++++ sound/soc/codecs/wcd937x.h | 653 ++++ 7 files changed, 5000 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: d35b2284e966c0bef3e2182a5c5ea02177dd32e4 -- 2.25.1

4 17

[PATCH] ASoC: cs35l56: Accept values greater than 0 as IRQ numbers
by Simon Trimmer 19 Jun '24

19 Jun '24

IRQ lookup functions such as those in ACPI can return error values when an IRQ is not defined. The i2c core driver converts the error codes to a value of 0 and the SPI bus driver passes them unaltered to client device drivers. The cs35l56 driver should only accept positive non-zero values as IRQ numbers. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Fixes: 8a731fd37f8b ("ASoC: cs35l56: Move utility functions to shared file") --- sound/soc/codecs/cs35l56-shared.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index 27869e14e9c8..880228f89baf 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -397,7 +397,7 @@ int cs35l56_irq_request(struct cs35l56_base *cs35l56_base, int irq) { int ret; - if (!irq) + if (irq < 1) return 0; ret = devm_request_threaded_irq(cs35l56_base->dev, irq, NULL, cs35l56_irq, -- 2.34.1

3 13

[PATCH v5] ASoc: tas2781: Enable RCA-based playback without DSP firmware download
by Shenghao Ding 19 Jun '24

19 Jun '24

In only loading RCA (Reconfigurable Architecture) binary case, no DSP program will be working inside tas2563/tas2781, that is dsp-bypass mode, do not support speaker protection, or audio acoustic algorithms in this mode. Fixes: ef3bcde75d06 ("ASoC: tas2781: Add tas2781 driver") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v5: - Add reviewed-by information. - Add comments on it is not an error in case of RCA-Only, ie, failure to load DSP firmware. 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 | 39 ++++++++++++++++++++----------- 3 files changed, 48 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..c64d458e524e 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -380,23 +380,37 @@ 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; + /* + * The baseline is the RCA-only case, and then the code attempts to + * load DSP firmware but in case of failures just keep going, i.e. + * failing to load DSP firmware is NOT an error. + */ + 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 +431,8 @@ 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 +479,14 @@ 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

2 1

[PATCH] ASoC: dt-bindings: audio-graph-card2: add support for aux devices
by Kuninori Morimoto 19 Jun '24

19 Jun '24

Add device tree property to define auxiliary devices to be added to Audio Graph Card which is already supported on Simle Card. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../devicetree/bindings/sound/audio-graph-card2.yaml | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml b/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml index d3ce4de449d5..f943f90d8b15 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml @@ -23,6 +23,11 @@ properties: Each entry is a pair of strings, the first being the connection's sink, the second being the connection's source. $ref: /schemas/types.yaml#/definitions/non-unique-string-array + aux-devs: + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + $ref: /schemas/types.yaml#/definitions/phandle-array multi: type: object description: Multi-CPU/Codec node -- 2.43.0

2 1

[PATCH 0/5] ASoC: dt-bindings: rename akXXXX to asahi-kasei,akXXXX
by Kuninori Morimoto 19 Jun '24

19 Jun '24

Hi Mark Cc Rob, Krzysztof. These patches adds vendor prefix to asahi-kasei Kuninori Morimoto (5): ASoC: dt-bindings: rename ak4118.txt to asahi-kasei,ak4118.txt ASoC: dt-bindings: rename ak5386.txt to asahi-kasei,ak5386.txt ASoC: dt-bindings: rename ak4375.yaml to asahi-kasei,ak4375.yaml ASoC: dt-bindings: rename ak4613.yaml to asahi-kasei,ak4613.yaml ASoC: dt-bindings: rename ak4642.yaml to asahi-kasei,ak4642.yaml .../bindings/sound/{ak4118.txt => asahi-kasei,ak4118.txt} | 0 .../bindings/sound/{ak4375.yaml => asahi-kasei,ak4375.yaml} | 2 +- .../bindings/sound/{ak4613.yaml => asahi-kasei,ak4613.yaml} | 2 +- .../bindings/sound/{ak4642.yaml => asahi-kasei,ak4642.yaml} | 2 +- .../bindings/sound/{ak5386.txt => asahi-kasei,ak5386.txt} | 0 5 files changed, 3 insertions(+), 3 deletions(-) rename Documentation/devicetree/bindings/sound/{ak4118.txt => asahi-kasei,ak4118.txt} (100%) rename Documentation/devicetree/bindings/sound/{ak4375.yaml => asahi-kasei,ak4375.yaml} (94%) rename Documentation/devicetree/bindings/sound/{ak4613.yaml => asahi-kasei,ak4613.yaml} (94%) rename Documentation/devicetree/bindings/sound/{ak4642.yaml => asahi-kasei,ak4642.yaml} (94%) rename Documentation/devicetree/bindings/sound/{ak5386.txt => asahi-kasei,ak5386.txt} (100%) -- 2.43.0

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[PATCH 0/2] ASoC: codecs: lpass-wsa: fix vi capture setup
by Srinivas Kandagatla 19 Jun '24

19 Jun '24

This two patches fixes issues with VI capture rate and path setup. 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 | 111 +++++++++++++++++++++++++++++-------- 1 file changed, 88 insertions(+), 23 deletions(-) --- base-commit: 1613e604df0cd359cf2a7fbd9be7a0bcfacfabd0 change-id: 20240618-lpass-wsa-vi-b63375f21d63 Best regards, -- Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>

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aloop driver introducing glitches
by noman pouigt 18 Jun '24

18 Jun '24

Hello Folks, I wanted to use aloop driver to dump playback audio data to a file in userspace for debugging without data being sent to hardware (just as a debug feature for future use). Command flow is this: 1. aloop capture device is started using below command first: "arec -D hw:0,1 /data/rec.wav &" 2. using aplay, userspace sends the playback data. "aplay -D hw:0,0 /data/test.wav" Using jiffies. Wondering if changing rate_shift (don't know what is that for) will help? However, even in this simple setup I can see aloop driver is corrupting the data and sinetone sent from playback has glitches. I have tried different period sizes as well but nothing worked. Wondering if this is by design i.e. if the aloop driver shouldn't be used for this purpose? Any debug steps will be highly appreciated. Thanks for your help,

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