June 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: codecs: wsa883x: use existing define instead of raw value
by Krzysztof Kozlowski 07 Jun '23

07 Jun '23

Use existing define for WSA883X_GLOBAL_PA_ENABLE instead of hard-coded value, just like in other places in this driver. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/wsa883x.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c index c609cb63dae6..5c1cfceb2956 100644 --- a/sound/soc/codecs/wsa883x.c +++ b/sound/soc/codecs/wsa883x.c @@ -1334,7 +1334,8 @@ static int wsa883x_digital_mute(struct snd_soc_dai *dai, int mute, int stream) WSA883X_DRE_GAIN_EN_MASK, WSA883X_DRE_GAIN_FROM_CSR); snd_soc_component_write_field(component, WSA883X_PA_FSM_CTL, - WSA883X_GLOBAL_PA_EN_MASK, 1); + WSA883X_GLOBAL_PA_EN_MASK, + WSA883X_GLOBAL_PA_ENABLE); } -- 2.34.1

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[PATCH 0/3] ASoC: stm32: fix dtbs_check warnings
by Olivier Moysan 07 Jun '23

07 Jun '23

Fix dtbs_check warnings in STM32MP15 DK boards Devices Trees for STM32 I2S and Cirrus CS42L51 codec. - Add OF graph port property in I2S and CS42L51 DT bindings. Fixes warnings: audio-controller@4000b000: Unevaluated properties are not allowed ('port' was unexpected) cs42l51@4a: Unevaluated properties are not allowed ('port' was unexpected) - Correct OF graph DAI audio format property for STM32MP15x Dkx I2S node Olivier Moysan (3): ASoC: dt-bindings: stm32: document audio of graph port for i2s ASoC: dt-bindings: document audio of graph port for cs42l51 ARM: dts: stm32: fix i2s endpoint format property for stm32mp15xx-dkx Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml | 4 ++++ Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml | 4 ++++ arch/arm/boot/dts/stm32mp15xx-dkx.dtsi | 2 +- 3 files changed, 9 insertions(+), 1 deletion(-) -- 2.25.1

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[PATCH] ASoC: mediatek: mt8188-mt6359: add i2c dependency
by Arnd Bergmann 07 Jun '23

07 Jun '23

From: Arnd Bergmann <arnd(a)arndb.de> The newly added driver is missing this dependency, causing a possible build failure: WARNING: unmet direct dependencies detected for SND_SOC_MAX98390 WARNING: unmet direct dependencies detected for SND_SOC_NAU8825 Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=m] Selected by [y]: - SND_SOC_MT8188_MT6359 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_MT8188 [=y] && MTK_PMIC_WRAP [=y] aarch64-linux-ld: sound/soc/codecs/max98390.o: in function `max98390_i2c_probe': max98390.c:(.text+0x514): undefined reference to `__devm_regmap_init_i2c' Fixes: 9f08dcbddeb30 ("ASoC: mediatek: mt8188-mt6359: support new board with nau88255") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/mediatek/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 4ea012342b522..90db67e0ce4fb 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -223,6 +223,7 @@ config SND_SOC_MT8188 config SND_SOC_MT8188_MT6359 tristate "ASoC Audio driver for MT8188 with MT6359 and I2S codecs" depends on SND_SOC_MT8188 && MTK_PMIC_WRAP + depends on I2C select SND_SOC_MT6359 select SND_SOC_HDMI_CODEC select SND_SOC_DMIC -- 2.39.2

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[PATCH 0/2] ASoC: Intel: HDA: two cleanups
by Pierre-Louis Bossart 07 Jun '23

07 Jun '23

One removal of a non-existent Kconfig initialization and one correction to enable SoundWire link when no HDaudio external codec is detected. This will generate more information and help resolve cases where the ACPI information is incorrect. Pierre-Louis Bossart (2): ASoC: SOF: Intel: fix SoundWire/HDaudio mutual exclusion ASoC: SOF: Intel: hda-pcm: remove kernel parameter init sound/soc/sof/intel/hda-pcm.c | 2 +- sound/soc/sof/intel/hda.c | 12 +++++++++++- 2 files changed, 12 insertions(+), 2 deletions(-) -- 2.37.2

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[PATCH v2 0/9] Add support for IIO devices in ASoC
by Herve Codina 07 Jun '23

07 Jun '23

Several weeks ago, I sent a series [1] for adding a potentiometer as an auxiliary device in ASoC. The feedback was that the potentiometer should be directly handled in IIO (as other potentiometers) and something more generic should be present in ASoC in order to have a binding to import some IIO devices into sound cards. The series related to the IIO potentiometer device is already under review [2]. This series introduces audio-iio-aux. Its goal is to offer the binding between IIO and ASoC. It exposes attached IIO devices as ASoC auxiliary devices and allows to control them through mixer controls. On my system, the IIO device is a potentiometer and it is present in an amplifier design present in the audio path. Compare to the previous iteration https://lore.kernel.org/linux-kernel/20230421124122.324820-1-herve.codina@b… This v2 series mainly: - updates the binding using a simple-card subnode and handles this new subnode in the simple-card driver. - Improves existing IIO code and documentation. - Renames simple-iio-aux to audio-iio-aux and fixes the driver itself. Best regards, Hervé [1] https://lore.kernel.org/linux-kernel/20230203111422.142479-1-herve.codina@b… [2] https://lore.kernel.org/linux-kernel/20230421085245.302169-1-herve.codina@b… Changes v1 -> v2 - Patch 1 Rename simple-iio-aux to audio-iio-aux Rename invert to snd-control-invert-range Remove the /schemas/iio/iio-consumer.yaml reference Remove the unneeded '|' after description - Patch 2 (new in v2) Introduce the simple-audio-card additional-devs subnode - Patch 3 (new in v2) Check err before switch() in iio_channel_read_max() - Patch 4 (new in v2) Fix raw reads and raw writes documentation - Patch 5 (patch 2 in v1) Check err before switch() in iio_channel_read_min() Fix documentation - Patch 6 (path 3 in v1) No changes - Patch 7 (patch 4 in v1) Rename simple-iio-aux to audio-iio-aux Rename invert to snd-control-invert-range Remove the mask usage from audio_iio_aux_{get,put}_volsw helpers Use directly PTR_ERR() in dev_err_probe() parameter Remove the '!!' construction Remove of_match_ptr() - Patch 8 (new in v2) Add a missing of_node_put() in the simple-card driver - Patch 9 (new in v2) Handle additional-devs in the simple-card driver Herve Codina (9): ASoC: dt-bindings: Add audio-iio-aux ASoC: dt-bindings: simple-card: Add additional-devs subnode iio: inkern: Check error explicitly in iio_channel_read_max() iio: consumer.h: Fix raw values documentation notes iio: inkern: Add a helper to query an available minimum raw value ASoC: soc-dapm.h: Add a helper to build a DAPM widget dynamically ASoC: codecs: Add support for the generic IIO auxiliary devices ASoC: simple-card: Add missing of_node_put() in case of error ASoC: simple-card: Handle additional devices .../bindings/sound/audio-iio-aux.yaml | 64 ++++ .../bindings/sound/simple-card.yaml | 53 +++ drivers/iio/inkern.c | 75 ++++- include/linux/iio/consumer.h | 37 ++- include/sound/soc-dapm.h | 12 +- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/audio-iio-aux.c | 302 ++++++++++++++++++ sound/soc/generic/simple-card.c | 53 ++- 9 files changed, 596 insertions(+), 14 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-iio-aux.yaml create mode 100644 sound/soc/codecs/audio-iio-aux.c -- 2.40.1

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[PATCH v2 00/26] ASoC/soundwire: add support for ACE2.x
by Bard Liao 07 Jun '23

07 Jun '23

This series uses the abstraction added in past kernel cycles to provide support for the ACE2.x integration. The existing SHIM and Cadence registers are now split in 3 (SHIM, IP, SHIM vendor-specific), with some parts also moved to the HDaudio Extended Multi link structures. Nothing fundamentally different except for the register map. This series only provides the basic mechanisms to expose SoundWire-based DAIs. The PCI parts and DSP management will be contributed later, and the DAI ops are now empty as well. The change is mainly on SoundWire. It would be better to go through SoundWire tree. v2: - Some cleanup for free, trigger, hw_params callbacks before introducing ACE2.x callbacks. Pierre-Louis Bossart (26): ASoC: SOF: Intel: shim: add enum for ACE 2.0 IP used in LunarLake soundwire: intel: add ACE2.x SHIM definitions soundwire: intel_ace2x: add empty new ops for LunarLake soundwire/ASOC: Intel: update offsets for LunarLake soundwire: intel/cadence: set ip_offset at run-time ASoC/soundwire: intel: pass hdac_bus pointer for link management soundwire: intel: add eml_lock in the interface for new platforms ASoC: SOF: Intel: hda: retrieve SoundWire eml_lock and pass pointer soundwire: intel_init: use eml_lock parameter soundwire: intel_ace2x: add debugfs support soundwire: intel_ace2x: add link power-up/down helpers soundwire: intel_ace2x: set SYNCPRD before powering-up soundwire: intel_ace2x: configure link PHY soundwire: intel_ace2x: add DAI registration soundwire: intel_ace2x: add sync_arm/sync_go helpers soundwire: intel_ace2x: use common helpers for bus start/stop soundwire: intel_ace2x: enable wake support soundwire: intel_ace2x: add check_cmdsync_unlocked helper soundwire: bus: add new manager callback to deal with peripheral enumeration soundwire: intel_ace2x: add new_peripheral_assigned callback soundwire: intel_ace2x: add pre/post bank switch callbacks ASoC: SOF/soundwire: re-add substream in params_stream structure soundwire: intel: remove .trigger callback implementation soundwire: intel: use substream for .trigger callback soundwire: intel: remove .free callback implementation soundwire: intel: use substream for .free callback drivers/soundwire/Makefile | 3 +- drivers/soundwire/bus.c | 3 + drivers/soundwire/cadence_master.h | 2 + drivers/soundwire/intel.c | 54 +--- drivers/soundwire/intel.h | 16 + drivers/soundwire/intel_ace2x.c | 393 ++++++++++++++++++++++++ drivers/soundwire/intel_ace2x_debugfs.c | 147 +++++++++ drivers/soundwire/intel_auxdevice.c | 17 + drivers/soundwire/intel_init.c | 21 +- include/linux/soundwire/sdw.h | 3 +- include/linux/soundwire/sdw_intel.h | 94 +++++- sound/soc/sof/intel/hda.c | 33 +- sound/soc/sof/intel/shim.h | 1 + 13 files changed, 723 insertions(+), 64 deletions(-) create mode 100644 drivers/soundwire/intel_ace2x.c create mode 100644 drivers/soundwire/intel_ace2x_debugfs.c -- 2.25.1

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[PATCH v4 1/3] docs: sound: add 'pcmtest' driver documentation
by Ivan Orlov 07 Jun '23

07 Jun '23

Add documentation for the new Virtual PCM Test Driver. It covers all possible usage cases: errors and delay injections, random and pattern-based data generation, playback and ioctl redefinition functionalities testing. We have a lot of different virtual media drivers, which can be used for testing of the userspace applications and media subsystem middle layer. However, all of them are aimed at testing the video functionality and simulating the video devices. For audio devices we have only snd-dummy module, which is good in simulating the correct behavior of an ALSA device. I decided to write a tool, which would help to test the userspace ALSA programs (and the PCM middle layer as well) under unusual circumstances to figure out how they would behave. So I came up with this Virtual PCM Test Driver. This new Virtual PCM Test Driver has several features which can be useful during the userspace ALSA applications testing/fuzzing, or testing/fuzzing of the PCM middle layer. Not all of them can be implemented using the existing virtual drivers (like dummy or loopback). Here is what can this driver do: - Simulate both capture and playback processes - Check the playback stream for containing the looped pattern - Generate random or pattern-based capture data - Inject delays into the playback and capturing processes - Inject errors during the PCM callbacks Also, this driver can check the playback stream for containing the predefined pattern, which is used in the corresponding selftest to check the PCM middle layer data transferring functionality. Additionally, this driver redefines the default RESET ioctl, and the selftest covers this PCM API functionality as well. The driver supports both interleaved and non-interleaved access modes, and have separate pattern buffers for each channel. The driver supports up to 4 channels and up to 8 substreams. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- V1 -> V2: - Rename the driver from from 'valsa' to 'pcmtest'. - Implement support for interleaved and non-interleaved access modes - Add support for 8 substreams and 4 channels - Extend supported formats - Extend and rewrite in C the selftest for the driver V2 -> V3: - Add separate pattern buffers for each channel - Speed up the capture data generation when using interleaved access mode - Extend the corresponding selftest to cover the multiple channels capturing and playback functionalities when using interleaved access mode. - Fix documentation issues V3 -> V4: - Fix issue in the selftest: there was a typo in the fscanf argument. Documentation/sound/cards/index.rst | 1 + Documentation/sound/cards/pcmtest.rst | 120 ++++++++++++++++++++++++++ 2 files changed, 121 insertions(+) create mode 100644 Documentation/sound/cards/pcmtest.rst diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst index c016f8c3b88b..49c1f2f688f8 100644 --- a/Documentation/sound/cards/index.rst +++ b/Documentation/sound/cards/index.rst @@ -17,3 +17,4 @@ Card-Specific Information hdspm serial-u16550 img-spdif-in + pcmtest diff --git a/Documentation/sound/cards/pcmtest.rst b/Documentation/sound/cards/pcmtest.rst new file mode 100644 index 000000000000..e163522f3205 --- /dev/null +++ b/Documentation/sound/cards/pcmtest.rst @@ -0,0 +1,120 @@ +.. SPDX-License-Identifier: GPL-2.0 + +The Virtual PCM Test Driver +=========================== + +The Virtual PCM Test Driver emulates a generic PCM device, and can be used for +testing/fuzzing of the userspace ALSA applications, as well as for testing/fuzzing of +the PCM middle layer. Additionally, it can be used for simulating hard to reproduce +problems with PCM devices. + +What can this driver do? +~~~~~~~~~~~~~~~~~~~~~~~~ + +At this moment the driver can do the following things: + * Simulate both capture and playback processes + * Generate random or pattern-based capturing data + * Inject delays into the playback and capturing processes + * Inject errors during the PCM callbacks + +It supports up to 8 substreams and 4 channels. Also it supports both interleaved and +non-interleaved access modes. + +Also, this driver can check the playback stream for containing the predefined pattern, +which is used in the corresponding selftest (alsa/pcmtest-test.sh) to check the PCM middle +layer data transferring functionality. Additionally, this driver redefines the default +RESET ioctl, and the selftest covers this PCM API functionality as well. + +Configuration +------------- + +The driver has several parameters besides the common ALSA module parameters: + + * fill_mode (bool) - Buffer fill mode (see below) + * inject_delay (int) + * inject_hwpars_err (bool) + * inject_prepare_err (bool) + * inject_trigger_err (bool) + + +Capture Data Generation +----------------------- + +The driver has two modes of data generation: the first (0 in the fill_mode parameter) +means random data generation, the second (1 in the fill_mode) - pattern-based +data generation. Let's look at the second mode. + +First of all, you may want to specify the pattern for data generation. You can do it +by writing the pattern to the debugfs file. There are pattern buffer debugfs entries +for each channel, as well as entries which contain the pattern buffer length. + + * /sys/kernel/debug/pcmtest/fill_pattern[0-3] + * /sys/kernel/debug/pcmtest/fill_pattern[0-3]_len + +To set the pattern for the channel 0 you can execute the following command: + +.. code-block:: bash + + echo -n mycoolpattern > /sys/kernel/debug/pcmtest/fill_pattern0 + +Then, after every capture action performed on the 'pcmtest' device the buffer for the +channel 0 will contain 'mycoolpatternmycoolpatternmycoolpatternmy...'. + +The pattern itself can be up to 4096 bytes long. + +Delay injection +--------------- + +The driver has 'inject_delay' parameter, which has very self-descriptive name and +can be used for time delay/speedup simulations. The parameter has integer type, and +it means the delay added between module's internal timer ticks. + +If the 'inject_delay' value is positive, the buffer will be filled slower, if it is +negative - faster. You can try it yourself by starting a recording in any +audiorecording application (like Audacity) and selecting the 'pcmtest' device as a +source. + +This parameter can be also used for generating a huge amount of sound data in a very +short period of time (with the negative 'inject_delay' value). + +Errors injection +---------------- + +This module can be used for injecting errors into the PCM communication process. This +action can help you to figure out how the userspace ALSA program behaves under unusual +circumstances. + +For example, you can make all 'hw_params' PCM callback calls return EBUSY error by +writing '1' to the 'inject_hwpars_err' module parameter: + +.. code-block:: bash + + echo 1 > /sys/module/snd_pcmtest/parameters/inject_hwpars_err + +Errors can be injected into the following PCM callbacks: + + * hw_params (EBUSY) + * prepare (EINVAL) + * trigger (EINVAL) + +Playback test +------------- + +This driver can be also used for the playback functionality testing - every time you +write the playback data to the 'pcmtest' PCM device and close it, the driver checks the +buffer for containing the looped pattern (which is specified in the fill_pattern +debugfs file for each channel). If the playback buffer content represents the looped +pattern, 'pc_test' debugfs entry is set into '1'. Otherwise, the driver sets it to '0'. + +ioctl redefinition test +----------------------- + +The driver redefines the 'reset' ioctl, which is default for all PCM devices. To test +this functionality, we can trigger the reset ioctl and check the 'ioctl_test' debugfs +entry: + +.. code-block:: bash + + cat /sys/kernel/debug/pcmtest/ioctl_test + +If the ioctl is triggered successfully, this file will contain '1', and '0' otherwise. -- 2.34.1

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[PATCH] ASoC: soc.h: remove snd_soc_compr_ops :: trigger
by Kuninori Morimoto 07 Jun '23

07 Jun '23

ASoC framework is not using trigger call-back for snd_soc_compr_ops. This patch remove it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc.h | 1 - 1 file changed, 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index ed79b27b0392..10e4ea0664af 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -633,7 +633,6 @@ struct snd_soc_compr_ops { int (*startup)(struct snd_compr_stream *); void (*shutdown)(struct snd_compr_stream *); int (*set_params)(struct snd_compr_stream *); - int (*trigger)(struct snd_compr_stream *); }; struct snd_soc_component* -- 2.25.1

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[PATCH] ASoC: add snd_soc_get_stream_cpu()
by Kuninori Morimoto 07 Jun '23

07 Jun '23

We are using get_stream_cpu() to get CPU stream which cares Codec2Codec. But it is static function for now, and we want to use it from other files. This patch makes it global function. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 34 ++++++++++++++++++++++++++++++++++ sound/soc/soc-dapm.c | 35 +---------------------------------- sound/soc/soc-pcm.c | 6 ++---- 4 files changed, 38 insertions(+), 38 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 533e553a343f..ed79b27b0392 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1292,6 +1292,7 @@ unsigned int snd_soc_daifmt_parse_clock_provider_raw(struct device_node *np, snd_soc_daifmt_clock_provider_from_bitmap( \ snd_soc_daifmt_parse_clock_provider_as_bitmap(np, prefix)) +int snd_soc_get_stream_cpu(struct snd_soc_dai_link *dai_link, int stream); int snd_soc_get_dai_id(struct device_node *ep); int snd_soc_get_dai_name(const struct of_phandle_args *args, const char **dai_name); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b48efc3a08d2..e8308926bd98 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3196,6 +3196,40 @@ unsigned int snd_soc_daifmt_parse_clock_provider_raw(struct device_node *np, } EXPORT_SYMBOL_GPL(snd_soc_daifmt_parse_clock_provider_raw); +int snd_soc_get_stream_cpu(struct snd_soc_dai_link *dai_link, int stream) +{ + /* + * [Normal] + * + * Playback + * CPU : SNDRV_PCM_STREAM_PLAYBACK + * Codec: SNDRV_PCM_STREAM_PLAYBACK + * + * Capture + * CPU : SNDRV_PCM_STREAM_CAPTURE + * Codec: SNDRV_PCM_STREAM_CAPTURE + */ + if (!dai_link->c2c_params) + return stream; + + /* + * [Codec2Codec] + * + * Playback + * CPU : SNDRV_PCM_STREAM_CAPTURE + * Codec: SNDRV_PCM_STREAM_PLAYBACK + * + * Capture + * CPU : SNDRV_PCM_STREAM_PLAYBACK + * Codec: SNDRV_PCM_STREAM_CAPTURE + */ + if (stream == SNDRV_PCM_STREAM_CAPTURE) + return SNDRV_PCM_STREAM_PLAYBACK; + + return SNDRV_PCM_STREAM_CAPTURE; +} +EXPORT_SYMBOL_GPL(snd_soc_get_stream_cpu); + int snd_soc_get_dai_id(struct device_node *ep) { struct snd_soc_component *component; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f2f04ce693a1..09a197ad6c2c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4322,39 +4322,6 @@ static void dapm_connect_dai_routes(struct snd_soc_dapm_context *dapm, snd_soc_dapm_add_path(dapm, src, sink, NULL, NULL); } -static int get_stream_cpu(struct snd_soc_dai_link *dai_link, int stream) -{ - /* - * [Normal] - * - * Playback - * CPU : SNDRV_PCM_STREAM_PLAYBACK - * Codec: SNDRV_PCM_STREAM_PLAYBACK - * - * Playback - * CPU : SNDRV_PCM_STREAM_CAPTURE - * Codec: SNDRV_PCM_STREAM_CAPTURE - */ - if (!dai_link->c2c_params) - return stream; - - /* - * [Codec2Codec] - * - * Playback - * CPU : SNDRV_PCM_STREAM_CAPTURE - * Codec: SNDRV_PCM_STREAM_PLAYBACK - * - * Capture - * CPU : SNDRV_PCM_STREAM_PLAYBACK - * Codec: SNDRV_PCM_STREAM_CAPTURE - */ - if (stream == SNDRV_PCM_STREAM_CAPTURE) - return SNDRV_PCM_STREAM_PLAYBACK; - - return SNDRV_PCM_STREAM_CAPTURE; -} - static void dapm_connect_dai_pair(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *codec_dai, @@ -4372,7 +4339,7 @@ static void dapm_connect_dai_pair(struct snd_soc_card *card, for_each_pcm_streams(stream) { int stream_cpu, stream_codec; - stream_cpu = get_stream_cpu(dai_link, stream); + stream_cpu = snd_soc_get_stream_cpu(dai_link, stream); stream_codec = stream; /* connect BE DAI playback if widgets are valid */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2abeb6f6a4a8..f16402ee5a19 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2781,10 +2781,8 @@ static int soc_get_playback_capture(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *codec_dai; /* Adapt stream for codec2codec links */ - int cpu_capture = dai_link->c2c_params ? - SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; - int cpu_playback = dai_link->c2c_params ? - SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; + int cpu_capture = snd_soc_get_stream_cpu(dai_link, SNDRV_PCM_STREAM_CAPTURE); + int cpu_playback = snd_soc_get_stream_cpu(dai_link, SNDRV_PCM_STREAM_PLAYBACK); for_each_rtd_codec_dais(rtd, i, codec_dai) { if (dai_link->num_cpus == 1) { -- 2.25.1

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[PATCH] ASoC: mediatek: mt8188-mt6359: Fix error code in codec_init
by Dan Carpenter 07 Jun '23

07 Jun '23

Return -EINVAL on this error path instead of returning success. Fixes: 9f08dcbddeb3 ("ASoC: mediatek: mt8188-mt6359: support new board with nau88255") Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- This is based on static analysis and not tested. sound/soc/mediatek/mt8188/mt8188-mt6359.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c index bc4b74970a46..ae02bbcd5d4e 100644 --- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c +++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c @@ -594,7 +594,7 @@ static int mt8188_max98390_codec_init(struct snd_soc_pcm_runtime *rtd) } if (rtd->dai_link->num_codecs <= 2) - return ret; + return -EINVAL; /* add widgets/controls/dapm for rear speakers */ ret = snd_soc_dapm_new_controls(&card->dapm, mt8188_rear_spk_widgets, -- 2.39.2

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