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[PATCH 00/17] ASoC: remove snd_soc_dai_set_tristate()
by Kuninori Morimoto 10 Dec '24

10 Dec '24

Hi Mark snd_soc_dai_set_tristate() has added in v2.6.27 (2008/Oct) by this commit 8c6529dbf881303920a415c2d14a500218661949 ("ALSA: asoc: core - add Digital Audio Interface (DAI) control functions.") But it has never been used during this 16 years. Let's remove it. Kuninori Morimoto (17): ASoC: pxa: remove snd_soc_dai_set_tristate() ASoC: codecs: cs35l32: remove snd_soc_dai_set_tristate() ASoC: codecs: cs35l33: remove snd_soc_dai_set_tristate() ASoC: codecs: cs35l34: remove snd_soc_dai_set_tristate() ASoC: codecs: cs42l73: remove snd_soc_dai_set_tristate() ASoC: codecs: cs53l30: remove snd_soc_dai_set_tristate() ASoC: codecs: wm8961: remove snd_soc_dai_set_tristate() ASoC: codecs: wm8994: remove snd_soc_dai_set_tristate() ASoC: codecs: wm8995: remove snd_soc_dai_set_tristate() ASoC: codecs: wm9713: remove snd_soc_dai_set_tristate() ASoC: codecs: adau1372: remove snd_soc_dai_set_tristate() ASoC: codecs: adau1977: remove snd_soc_dai_set_tristate() ASoC: codecs: adau7118: remove snd_soc_dai_set_tristate() ASoC: codecs: madera: remove snd_soc_dai_set_tristate() ASoC: codecs: arizona: remove snd_soc_dai_set_tristate() ASoC: codecs: twl4030: remove snd_soc_dai_set_tristate() ASoC: remove snd_soc_dai_set_tristate() include/sound/soc-dai.h | 3 --- sound/soc/codecs/adau1372.c | 14 -------------- sound/soc/codecs/adau1977.c | 15 --------------- sound/soc/codecs/adau7118.c | 19 ------------------- sound/soc/codecs/arizona.c | 17 ----------------- sound/soc/codecs/cs35l32.c | 9 --------- sound/soc/codecs/cs35l33.c | 21 --------------------- sound/soc/codecs/cs35l34.c | 15 --------------- sound/soc/codecs/cs42l73.c | 10 ---------- sound/soc/codecs/cs53l30.c | 10 ---------- sound/soc/codecs/madera.c | 22 ---------------------- sound/soc/codecs/twl4030.c | 28 ---------------------------- sound/soc/codecs/wm8961.c | 14 -------------- sound/soc/codecs/wm8994.c | 28 ---------------------------- sound/soc/codecs/wm8995.c | 33 --------------------------------- sound/soc/codecs/wm9713.c | 17 ----------------- sound/soc/pxa/pxa-ssp.c | 21 --------------------- sound/soc/soc-dai.c | 19 ------------------- 18 files changed, 315 deletions(-) -- 2.43.0

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Re: [vdr] Identifying MPEG-4 HE-AAC (LATM, LAOS) audio formats
by schorpp 09 Dec '24

09 Dec '24

Am 06.12.24 um 21:32 schrieb Marko Mäkelä: > A couple of weeks ago, Finland took the penultimate step to get rid of > DVB-T. Since years, we had DVB-T and DVB-T2 running in parallel. The > DVB-T transponders should be shut down in 2025. same here but h.265 video and aac audio. > > The step was to introduce a second free-to-air DVB-T2 bouquet to carry > some channels that were previously only available with DVB-T and MPEG-2. > The VDR channel search failed to find this transponder; I ran a patched > version of https://github.com/mighty-p/t2scan/pull/15/ to generate a new > channels.conf. (Is this a bug of VDR, or a feature? I remember that when > I originally set up VDR, I had to get a channels.conf from tscan or > t2scan.) have You tried the latest git version of scan in dvb-utils @linuxtx.org repo? > > Unlike our first free-to-air DVB-T2 bouquet, which uses MPEG-4 H.264 > video with AC3 audio, the second one uses H.264 video and HE-AAC audio, > also known as LATM or LOAS. This had to be explicitly enabled for my > output device: https://github.com/reufer/rpihddevice/issues/20 > > The HE-AAC support in rpihddevice is experimental, as it has been in ffmpeg until today and rpihddevive is based on it? > maybe because the > sampling rate is hard-coded as 48 kHz and the number of channels as 2. > Unlike AC3, it does not look like these parameters would be available in > any audio frame header. As far I understood from a NorDig.org > specification, the sampling rate would always be 48 kHz, but the number > of channels could be identified by something in the container, > specifically, Table 12.28 and Table 12.29 at page 164 of > https://nordig.org/wp-content/uploads/2016/03/NorDig-Unified- > Requirements-ver.-3.2.1.pdf Have You tried the latest git ffprobe version, does it detect the aac flavor of Your dvb-t2 DVB channels correctly? Have You tried latest git mediainfo? > > What would have to be changed in VDR or rpihddevice or both to correctly > switch between 2-channel and 5.1-channel HE-AAC audio? This is mostly an > academic question and exercise, because I only have stereo audio > hardware and don't know if any programming is actually available with > multichannel audio. Not at all academic. Since AVRs do not implement Fraunhofer codecs we need a HDNI PCM Multichannel driver for the raspberry pi to playback 5.1 channel aac without transcoding it to ac3 which can be done with mplayer on the fly. I've made a HDMI PCM Multichannel output to AVR driver patch for the Allwinner SoC alsa audio driver, look for it on the alsa-dev list archives. > > Best regards, > > Marko y tom

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[PATCH treewide 00/11] ASoC: Clean up {hp,mic}-det-gpio handling
by Geert Uytterhoeven 09 Dec '24

09 Dec '24

Hi all, The "gpio" suffix for GPIO consumers was deprecated a while ago, in favor of the "gpios" suffix. However, there are still several users of the "hp-det-gpio" and "mic-det-gpio" properties, in DT bindings, drivers, and DT source files. Drivers that use gpiolib handle both the deprecated and new properties transparently, but that is not the case when using of_property_*() calls directly. - The first patch fixes a bug, as the Freescale Generic ASoC Sound driver handles only the deprecated properties, while there is already a user of the new variants, - The second patch updates the Audio Graph and Simple Audio Card DT bindings, - Patches 3-9 converts various DTS files to use the new properties, - The last 2 patches convert Freescale sound device nodes to use the new properties. All patches can be applied independently, except for the last two, which depend on the first patch to avoid regressions. This series was compile-tested only. Thanks for your comments! Geert Uytterhoeven (11): ASoC: fsl-asoc-card: Add missing handling of {hp,mic}-dt-gpios ASoC: dt-bindings: Deprecate {hp,mic}-det-gpio ARM: dts: marvell: mmp2-olpc-xo-1-75: Switch to {hp,mic}-det-gpios arm64: dts: freescale: imx: Switch to simple-audio-card,hp-det-gpios arm64: dts: renesas: beacon-renesom: Switch to mic-det-gpios arm64: dts: rockchip: Switch to simple-audio-card,hp-det-gpios arm64: dts: rockchip: Switch to hp-det-gpios arm64: dts: uniphier: Switch to hp-det-gpios mips: dts: ingenic: Switch to simple-audio-card,hp-det-gpios ARM: dts: nxp: imx: Switch to {hp,mic}-det-gpios arm64: dts: freescale: imx: Switch to hp-det-gpios .../devicetree/bindings/sound/audio-graph.yaml | 6 ++++++ .../devicetree/bindings/sound/simple-card.yaml | 12 ++++++++++++ arch/arm/boot/dts/marvell/mmp2-olpc-xo-1-75.dts | 4 ++-- arch/arm/boot/dts/nxp/imx/imx6qdl-sabresd.dtsi | 4 ++-- arch/arm/boot/dts/nxp/imx/imx6sl-evk.dts | 2 +- arch/arm/boot/dts/nxp/imx/imx6sll-evk.dts | 2 +- arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi | 2 +- arch/arm/boot/dts/nxp/imx/imx6ul-14x14-evk.dtsi | 2 +- arch/arm/boot/dts/nxp/imx/imx7d-sdb.dts | 2 +- .../boot/dts/freescale/imx8mq-librem5-devkit.dts | 2 +- arch/arm64/boot/dts/freescale/imx8mq-librem5.dtsi | 2 +- arch/arm64/boot/dts/freescale/imx8qm-mek.dts | 2 +- arch/arm64/boot/dts/freescale/imx8qxp-mek.dts | 2 +- arch/arm64/boot/dts/freescale/imx95-19x19-evk.dts | 2 +- .../boot/dts/renesas/beacon-renesom-baseboard.dtsi | 2 +- .../boot/dts/rockchip/rk3326-anbernic-rg351m.dtsi | 2 +- .../arm64/boot/dts/rockchip/rk3326-gameforce-chi.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3326-odroid-go.dtsi | 2 +- arch/arm64/boot/dts/rockchip/rk3368-lba3368.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3399-eaidk-610.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3399-pinebook-pro.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3399-roc-pc-plus.dts | 2 +- .../boot/dts/rockchip/rk3399-rock-pi-4a-plus.dts | 2 +- .../boot/dts/rockchip/rk3399-rock-pi-4b-plus.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3399-rock-pi-4c.dts | 2 +- .../boot/dts/rockchip/rk3399-sapphire-excavator.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg-arc.dtsi | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg353p.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg353ps.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg353v.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg353vs.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg503.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3566-pinetab2.dtsi | 2 +- arch/arm64/boot/dts/rockchip/rk3566-powkiddy-x55.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3568-odroid-m1.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-armsom-sige7.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-evb1-v10.dts | 2 +- .../dts/rockchip/rk3588-friendlyelec-cm3588-nas.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-nanopc-t6.dtsi | 2 +- arch/arm64/boot/dts/rockchip/rk3588-ok3588-c.dts | 2 +- .../boot/dts/rockchip/rk3588-orangepi-5-plus.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-quartzpro64.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-rock-5-itx.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-rock-5b.dts | 2 +- .../boot/dts/rockchip/rk3588s-gameforce-ace.dts | 2 +- .../boot/dts/socionext/uniphier-ld11-global.dts | 2 +- .../boot/dts/socionext/uniphier-ld20-global.dts | 2 +- arch/mips/boot/dts/ingenic/gcw0.dts | 2 +- arch/mips/boot/dts/ingenic/rs90.dts | 2 +- sound/soc/fsl/fsl-asoc-card.c | 8 +++++--- 50 files changed, 72 insertions(+), 52 deletions(-) -- 2.34.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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Audio Crackling with some Steam games
by orbea 06 Dec '24

06 Dec '24

Hi, OS: Gentoo Kernel: 6.12.1 alsa-lib: 1.2.12 Product Name: X670E PG Lightning GPU: [AMD/ATI] Navi 10 [Radeon RX 5600 OEM/5600 XT / 5700/5700 XT] (rev c1) When using HDMI audio I am having an issue where some Steam games as run using Proton will have audio crackling or static after a short period of time after starting the game. The games that I know do this are 'Tales of Symphonia', 'Ys VIII: Lacrimosa of DANA' and 'Metaphor: Refantazio'. While many other games do not have this problem. I am not sure if this is a problem specific to the Steam ecosystem, the games, alsa, the kernel or something else? At least in the case of 'Ys VIII' I can only reproduce the problem when running the game in Steam where the same game files when run using the system wine or self compiled proton there are no audio issues. In the Proton logs it shows many underruns as soon as the issue starts: ALSA lib pcm.c:8545:(snd_pcm_recover) underrun occurred The problem also appears when using a simple /etc/asound.conf that contains only: defaults.pcm.card 0 defaults.pcm.device 9 defaults.ctl.card 0 I have also tried asking about this issue on the Proton Github issue tracker and the FAudio issue tracker. https://github.com/FNA-XNA/FAudio/issues/367 https://github.com/ValveSoftware/Proton/issues/3820#issuecomment-2477846139 https://github.com/ValveSoftware/Proton/issues/822#issuecomment-2326999328 In the FAudio issue it was suggested it was related to protonmediaconverter where the system wine and proton use ffmpeg, but I am not sure how to confirm this. I attached the alsa-info.txt. Is there anything I can do to further diagnose, resolve or workaround this? Any further info I neglected to provide that can help this? Thanks!

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[PATCH v2 0/3] ASoC: mediatek:mt8186: fix both the speaker and hdmi
by Jiaxin Yu 06 Dec '24

06 Dec '24

The speaker and hdmi of mt8186 platform are shared the same port of I2S, when connect the external display, use build-in speakers to play audio, they both playback at the same time. So we want to manage the playback device through DAPM events. Change since v1: Support audio function for it6505 bridge to hdmi-codec Jiaxin Yu (3): ASoC: hdmi-codec: Add event handler for hdmi TX ASoC: mediatek: mt8186: correct the HDMI widgets drm/bridge: it6505: Add audio support drivers/gpu/drm/bridge/ite-it6505.c | 81 +++++++++++++++++-- include/sound/hdmi-codec.h | 6 ++ sound/soc/codecs/hdmi-codec.c | 37 ++++++++- .../mt8186/mt8186-mt6366-da7219-max98357.c | 2 +- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 2 +- 5 files changed, 116 insertions(+), 12 deletions(-) -- 2.18.0

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[PATCH v5 RESEND 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 06 Dec '24

06 Dec '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20241002093904.180979… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v5: - Drop Jaroslav Kysela's patch as it has been merged. - Add Jaroslav Kysela's Acked-by tag, received in v3. changes in v4: - remove the RFC tag, no comments receive in v3 - Add Jaroslav Kysela's patch in this patch set. because it may be better for reviewing in a full patch set. - Fix the list_for_each_entry_reverse to list_for_each_entry_safe_reverse - Fix some coding style issues in Jaroslav Kysela's patch changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_params.h | 23 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 +++ sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 +++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1260 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

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[GIT PULL] ASoC fixes for v6.13-rc1
by Mark Brown 05 Dec '24

05 Dec '24

The following changes since commit 40384c840ea1944d7c5a392e8975ed088ecf0b37: Linux 6.13-rc1 (2024-12-01 14:28:56 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.13-rc1 for you to fetch changes up to ec16a3cdf37e507013062f9c4a2067eacdd12b62: ASoC: mediatek: mt8188-mt6359: Remove hardcoded dmic codec (2024-12-05 13:39:58 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.13 A few small fixes for v6.13, all system specific - the biggest thing is the fix for jack handling over suspend on some Intel laptops. ---------------------------------------------------------------- Dan Carpenter (1): ASoC: SOF: ipc3-topology: fix resource leaks in sof_ipc3_widget_setup_comp_dai() Marek Maslanka (1): ASoC: Intel: avs: da7219: Remove suspend_pre() and resume_post() Nícolas F. R. A. Prado (1): ASoC: mediatek: mt8188-mt6359: Remove hardcoded dmic codec sound/soc/intel/avs/boards/da7219.c | 17 ----------------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 4 +--- sound/soc/sof/ipc3-topology.c | 7 ++++--- 3 files changed, 5 insertions(+), 23 deletions(-)

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With SOF enabled in sof-hda-generic.tplg device (no DMIC) the control 'Capture Source' is not controlled by UCM
by GitHub issues - edited 05 Dec '24

05 Dec '24

alsa-project/alsa-ucm-conf issue #484 was edited from singalsu: The configuration is not usual. If there is no DMIC in the system usually the legacy mode is selected (as with `options snd_intel_dspcfg dsp_driver=1`). In such PCs SOF can be enabled with `options snd_intel_dspcfg dsp_driver=3`. When a headset is plugged/unplugged the playback correctly switches between speaker and headphone/headset earbuds. But the microphone capture remains muted. I noticed with legacy mode where internal microphones work that control name='Capture Source' changes to 1 when headset is removed: ``` singalsu@asteri:~$ amixer -c0 cget name='Capture Source' numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Headset Mic' ; Item #1 'Internal Mic' : values=1 singalsu@asteri:~$ amixer -c0 cget name='Capture Source' numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Headset Mic' ; Item #1 'Internal Mic' : values=0 ``` This is not happening when SOF is enabled. But I could make internal microphones capture work with `amixer -c0 cset name='Capture Source' 1`. Is there a way to fix this in ucm2? I'm not absolutely sure about this issue in Fedora 41 but this happens with my ucm2 development code with plain git master branch (commit 0782bf435b8332833e7b3fc91501d78473dd6edf). Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/484 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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With SOF enabled in sof-hda-generic.tplg device (no DMIC) the control 'Capture Source' is not controlled by UCM
by GitHub issues - opened 05 Dec '24

05 Dec '24

alsa-project/alsa-ucm-conf issue #484 was opened from singalsu: The configuration is not usual. If there is no DMIC in the system usually the legacy mode is selected (as with `options snd_intel_dspcfg dsp_driver=1`). In such PCs SOF can be enabled with `options snd_intel_dspcfg dsp_driver=3`. When a headset is plugged/unplugged the playback correctly switches between speaker and headphone/headset earbuds. But the microphone capture remains muted. I noticed with legacy mode where internal microphones work that control name='Capture Source' changes to 1 when headset is removed: ``` singalsu@asteri:~$ amixer -c0 cget name='Capture Source' numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Headset Mic' ; Item #1 'Internal Mic' : values=1 singalsu@asteri:~$ amixer -c0 cget name='Capture Source' numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Headset Mic' ; Item #1 'Internal Mic' : values=0 ``` This is not happening when SOF is enabled. But I could make internal microphones capture work with `amixer -c0 cset name='Capture Source' 1`. Is there a way to fix this in ucm2? I'm not absolutely sure about this issue in Fedora 41 but this happens with my modified ucm2 development code with git master branch. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/484 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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scsi: aic7xxx: Fix build 'aicasm' warning
by wangdicheng 05 Dec '24

05 Dec '24

1 0