August 2022 - Alsa-devel - mailman.alsa-project.org

Support MSI MPG Z690 EDGE WIFI DDR4 (ALC4080)
by GitHub issues - edited 31 Aug '22

31 Aug '22

alsa-project/alsa-ucm-conf issue #210 was edited from tekstryder: Per @perexg request, filing a new issue after testing git master branch changes for the ALC4080 and finding any profile(s) to be missing/nonfunctional. USB ID: `0db0:1feb` In version 1.2.7.2 of .../ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf I need to change: - Define.SPDIFIndex "3" to Define.SPDIFIndex "2" For proper playback via S/PDIF. After that change the release version is working great as far as I could test. Please let me know if I can provide any other useful information. Attaching the output of `alsaucm -c hw:2 dump text` Attaching the output of `pactl list sinks` [ucm-dump.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9463734/u… [sinks.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9463784/sink… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/210 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Support MSI MPG Z690 EDGE WIFI DDR4 (ALC4080)
by GitHub issues - opened 31 Aug '22

31 Aug '22

alsa-project/alsa-ucm-conf issue #210 was opened from tekstryder: Per @perexg request, filing a new issue after testing git master branch changes for the ALC4080 and finding any profile(s) to be missing/nonfunctional. USB ID: `0db0:1feb` In version 1.2.7.2 of .../ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf I need to change: - Define.SPDIFIndex "3" to Define.SPDIFIndex "Define.SPDIFIndex "2"" For proper playback via S/PDIF. After that change the release version is working great as far as I could test. Please let me know if I can provide any other useful information. Attaching the output of `alsaucm -c hw:2 dump text` Attaching the output of `pactl list sinks` [ucm-dump.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9463734/u… [sinks.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9463784/sink… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/210 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v2 0/2] libfs: Introduce tokenize_user_input()
by Cezary Rojewski 31 Aug '22

31 Aug '22

Continuation of recent upstream discussion [1] regarding user string tokenization. First, tokenize_user_input() is introduced to allow for splitting specified user string into a sequence of integers. Makes use of get_options() internally so the parsing logic is not duplicated. With that done, redundant parts of the sound driver are removed. Originally similar functionality was added for the SOF sound driver. As more users are on the horizon, it is desirable to update existing fs code and provide a unified solution. Changes in v2: - reused get_options() so no parsing logic is duplicated - simplified __user variant with help of memdup_user_nul() Both suggested by Andy, thanks for thourough review [1]: https://lore.kernel.org/alsa-devel/20220707091301.1282291-1-cezary.rojewski… Cezary Rojewski (2): libfs: Introduce tokenize_user_input() ASoC: SOF: Remove strsplit_u32() and tokenize_input() fs/libfs.c | 45 +++++++++++++++ include/linux/fs.h | 1 + sound/soc/sof/sof-client-probes.c | 92 ++++--------------------------- 3 files changed, 57 insertions(+), 81 deletions(-) -- 2.25.1

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[PATCH] ALSA: usb-audio: Drop superfluous interface setup at parsing
by Takashi Iwai 31 Aug '22

31 Aug '22

We reset each interface that is being parsed for each stream, but this is superfluous and even can lead to spurious errors. Since the interface is set up properly at opening the endpoint for each actual stream operation, let's drop the superfluous one. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/stream.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index ceb93d798182..99578e9a8af0 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1221,12 +1221,6 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, if (err < 0) return err; } - - /* try to set the interface... */ - usb_set_interface(chip->dev, iface_no, 0); - snd_usb_init_pitch(chip, fp); - snd_usb_init_sample_rate(chip, fp, fp->rate_max); - usb_set_interface(chip->dev, iface_no, altno); } return 0; } -- 2.35.3

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[PATCH 1/2] ALSA: usb-audio: Inform the delayed registration more properly
by Takashi Iwai 31 Aug '22

31 Aug '22

The info message that was added in the commit a4aad5636c72 ("ALSA: usb-audio: Inform devices that need delayed registration") is actually useful to know the need for the delayed registration. However, it turned out that this doesn't catch the all cases; namely, this warned only when a PCM stream is attached onto the existing PCM instance, but it doesn't count for a newly created PCM instance. This made confusion as if there were no further delayed registration. This patch moves the check to the code path for either adding a stream or creating a PCM instance. Also, make it simpler by checking the card->registered flag instead of querying each snd_device state. Fixes: a4aad5636c72 ("ALSA: usb-audio: Inform devices that need delayed registration") Link: https://bugzilla.kernel.org/show_bug.cgi?id=216082 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/stream.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index ceb93d798182..40b7821c6c99 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -495,6 +495,10 @@ static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip, return 0; } } + + if (chip->card->registered) + chip->need_delayed_register = true; + /* look for an empty stream */ list_for_each_entry(as, &chip->pcm_list, list) { if (as->fmt_type != fp->fmt_type) @@ -502,9 +506,6 @@ static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip, subs = &as->substream[stream]; if (subs->ep_num) continue; - if (snd_device_get_state(chip->card, as->pcm) != - SNDRV_DEV_BUILD) - chip->need_delayed_register = true; err = snd_pcm_new_stream(as->pcm, stream, 1); if (err < 0) return err; -- 2.35.3

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[PATCH v3 1/2] ASoC: dt-bindings: max98396: Document data monitor properties
by Daniel Mack 31 Aug '22

31 Aug '22

This device features a data monitor that puts the device in software reset upon a configurable set of events. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- v1 -> v2: fix a typo and remove a stray blank line v2 -> v3: Rename device tree properties, drop $ref for standard unit suffix properties .../bindings/sound/adi,max98396.yaml | 34 +++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/adi,max98396.yaml b/Documentation/devicetree/bindings/sound/adi,max98396.yaml index 8d2ef991db40..7985b1410017 100644 --- a/Documentation/devicetree/bindings/sound/adi,max98396.yaml +++ b/Documentation/devicetree/bindings/sound/adi,max98396.yaml @@ -78,6 +78,40 @@ properties: interleaved on a single output channel. type: boolean + adi,dmon-stuck-enable: + description: + Enables the "data monitor stuck" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-stuck-threshold-bits: + description: + Sets the threshold for the "data monitor stuck" feature, in bits. + enum: [9, 11, 13, 15] + default: 15 + + adi,dmon-magnitude-enable: + description: + Enables the "data monitor magnitude" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-magnitude-threshold-bits: + description: + Sets the threshold for the "data monitor magnitude" feature, in bits. + enum: [2, 3, 4, 5] + default: 5 + + adi,dmon-duration-ms: + description: + Sets the duration for the "data monitor" feature, in milliseconds. + enum: [64, 256, 1024, 4096] + default: 64 + reset-gpios: maxItems: 1 -- 2.37.2

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[PATCH 0/5] ASoC: codecs: minor cppcheck cleanups
by Pierre-Louis Bossart 31 Aug '22

31 Aug '22

Small number of cleanups that were either missed in previous versions or detected by new cppcheck version. Pierre-Louis Bossart (5): ASoC: hdmi-codec: remove unused definitions ASoC: wcd-mbhc-v2: remove always-true condition ASoC: wcd9335: remove always-true condition ASoC: fsl: fsl-utils: remove useless assignment ASoC: ti: omap-mcbsp: remove useless assignment sound/soc/codecs/hdmi-codec.c | 4 ---- sound/soc/codecs/wcd-mbhc-v2.c | 2 +- sound/soc/codecs/wcd9335.c | 4 +--- sound/soc/fsl/fsl_utils.c | 2 +- sound/soc/ti/omap-mcbsp.c | 2 +- 5 files changed, 4 insertions(+), 10 deletions(-) -- 2.34.1

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[PATCH 0/2] ASoC: SOF: compress: Add support for timestamp on capture
by Laurentiu Mihalcea 31 Aug '22

31 Aug '22

The purpose of this patch series is to add support for timestamping on capture direction using the compress API. This is simply done by splitting sof_compr_copy into 2 functions: sof_compr_copy_playback and sof_compr_copy_capture. Each of these functions handles one of the possible directions: capture or playback and is called in sof_compr_copy based on the stream's direction. The only difference between sof_compr_copy_playback and sof_compr_copy_capture is the fact that on playback case we need to copy data from user space and on capture we need to copy data to user space. Laurentiu Mihalcea (2): ASoC: SOF: compress: Move sof_compr_copy functionality ASoC: SOF: compress: Add copy function for capture case sound/soc/sof/compress.c | 47 +++++++++++++++++++++++++++++++++------- 1 file changed, 39 insertions(+), 8 deletions(-) -- 2.34.1

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[PATCH v2] ASoC: amd: acp: Modify dai_id macros to be more generic
by Venkata Prasad Potturu 31 Aug '22

31 Aug '22

Change dai_id macros to make I2S instances in order. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- Changes since v1: -- Change commit description. sound/soc/amd/acp/amd.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index af9603724a68..60a7c95f134f 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -21,9 +21,9 @@ #define ACP3X_DEV 3 #define ACP6X_DEV 6 -#define I2S_SP_INSTANCE 0x00 -#define I2S_BT_INSTANCE 0x01 -#define DMIC_INSTANCE 0x02 +#define DMIC_INSTANCE 0x00 +#define I2S_SP_INSTANCE 0x01 +#define I2S_BT_INSTANCE 0x02 #define I2S_HS_INSTANCE 0x03 #define MEM_WINDOW_START 0x4080000 -- 2.25.1

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[PATCH 0/6] ASoC: qcom: add support for sc8280xp machine
by Srinivas Kandagatla 31 Aug '22

31 Aug '22

THis patchset adds support for SC8280XP SoC machine driver. First patch moves some of the commonly used code to common from sm8250 machine driver and the follow on code adds minimal support for sc8280xp. Currently this driver is only tested with SmartSpeakers and Headset on Lenovo Thinkpad X13s. Support for sm8450 is tested and I will post the patches soon. Thanks, Srini Srinivas Kandagatla (3): ASoC: dt-bindings: qcom,sm8250: add compatibles for sm8450 and sm8250 ASoC: qcom: sm8250: move some code to common ASoC: qcom: add machine driver for sc8280xp .../bindings/sound/qcom,sm8250.yaml | 2 + sound/soc/qcom/Kconfig | 11 ++ sound/soc/qcom/Makefile | 2 + sound/soc/qcom/common.c | 169 ++++++++++++++++++ sound/soc/qcom/common.h | 12 ++ sound/soc/qcom/sc8280xp.c | 157 ++++++++++++++++ sound/soc/qcom/sm8250.c | 152 +--------------- 7 files changed, 360 insertions(+), 145 deletions(-) create mode 100644 sound/soc/qcom/sc8280xp.c -- 2.21.0

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