August 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/4] ASoC: Intel: boards: quirks and DMIC corrections
by Pierre-Louis Bossart 12 Aug '19

12 Aug '19

Extend quirk for GeminiLake-based Chromebooks and correct missing dependencies for DMIC support (align with all other machine drivers). Keyon Jie (3): ASoC: Intel: sof-rt5682: add dmic dapm widget to support dmic PCM ASoC: Intel: skl-hda-dsp-generic: add dependency to dmic driver ASoC: Intel: skl-hda-dsp-generic: add dmic dapm widget and route Sathya Prakash M R (1): ASoC: Intel: boards: Match Product Family instead of product sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 4 ++++ sound/soc/intel/boards/sof_rt5682.c | 7 +++++-- 3 files changed, 10 insertions(+), 2 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH 0/2] ASoC: pcm3168a: DAC/ADC independence and right_j fix
by Peter Ujfalusi 12 Aug '19

12 Aug '19

Hi, The first patch makes sure that the DAC and ADC side of the codec can be used and configured independently from each other. The second patch allows the reconfiguration of tdm_slots and slot_width. It is needed in case the codec is configured as slave and right_j whisch is the only mode when it can support 16 and 24 bit audio, but 16 bit requires 16 slot_width while 24 bit can use 48 or 32 slot_width. The codec driver must allow runtime reconfiguration of the tdm_slots and slot_width for this purpose. Regards, Peter --- Peter Ujfalusi (2): ASoC: pcm3168a: Retain the independence of DAC and ADC side of the codec ASoC: pcm3168a: Allow reconfiguration of tdm_slots and slot_width sound/soc/codecs/pcm3168a.c | 129 ++++++++++++++---------------------- 1 file changed, 50 insertions(+), 79 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH] ASoC: ti: davinci-mcasp: Protect hw_params callback against race
by Peter Ujfalusi 12 Aug '19

12 Aug '19

If the playback and capture of the same McASP is connected to different dai link (non duplex PCM links, like with pcm3168a codec) then there is a high probability of race between the two direction leaving McASP in a confused state. Protect the hw_params() with a mutex to make sure that this is not happening. The concurrent execution of hw_params for capture and playback can be easily triggered with custom .asoundrc file (for pcm3168a codec): pcm.dmixed8 { type dmix ipc_key 2048 ipc_perm 0666 slave { pcm "hw:0,0" format S24_LE channels 8 rate 96000 } bindings { 0 0 1 1 2 2 3 3 4 4 5 5 6 6 7 7 } } pcm.cpb-headset-1 { type plug slave.pcm dmixed8 hint { show on description "Headset 1 jack" } ttable.0.0 1 ttable.1.4 1 } pcm.dsnooped6 { type dsnoop ipc_key 2049 ipc_perm 0666 slave { pcm "hw:0,1" format S24_LE channels 6 rate 96000 } bindings { 0 0 1 1 2 2 3 3 4 4 5 5 } } pcm.cpb-mic-1 { type plug slave.pcm "dsnooped6" hint { show on description "Microphone 1 jack" } ttable.0.0 1 ttable.1.3 1 } Then running: arecord -D cpb-mic-1 -f S24_LE -c2 -r48000 | aplay -D cpb-headset-1 -f S24_LE -c2 -r48000 Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/ti/davinci-mcasp.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 7aa3c32e4a49..fe7a0b3572e2 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -111,6 +111,7 @@ struct davinci_mcasp { u32 channels; int max_format_width; u8 active_serializers[2]; + struct mutex mutex; #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; @@ -1169,6 +1170,8 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int period_size = params_period_size(params); int ret; + mutex_lock(&mcasp->mutex); + switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: case SNDRV_PCM_FORMAT_S8: @@ -1197,12 +1200,13 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, default: printk(KERN_WARNING "davinci-mcasp: unsupported PCM format"); - return -EINVAL; + ret = -EINVAL; + goto out; } ret = davinci_mcasp_set_dai_fmt(cpu_dai, mcasp->dai_fmt); if (ret) - return ret; + goto out; /* * If mcasp is BCLK master, and a BCLK divider was not provided by @@ -1223,7 +1227,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, ret = mcasp_common_hw_param(mcasp, substream->stream, period_size * channels, channels); if (ret) - return ret; + goto out; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) ret = mcasp_dit_hw_param(mcasp, params_rate(params)); @@ -1232,7 +1236,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, channels); if (ret) - return ret; + goto out; davinci_config_channel_size(mcasp, word_length); @@ -1242,7 +1246,10 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, mcasp->max_format_width = word_length; } - return 0; +out: + mutex_unlock(&mcasp->mutex); + + return ret; } static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, @@ -2335,6 +2342,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (ret) return -EINVAL; + mutex_init(&mcasp->mutex); + ret = devm_snd_soc_register_component(&pdev->dev, &davinci_mcasp_component, &davinci_mcasp_dai[pdata->op_mode], 1); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH v4 0/7] Allwinner H6 SPDIF support
by Clément Péron 12 Aug '19

12 Aug '19

*H6 DMA support IS REQUIRED* Allwinner H6 SoC has a SPDIF controller called One Wire Audio (OWA) which is different from the previous H3 generation and not compatible. Difference are an increase of fifo sizes, some memory mapping are different and there is now the possibility to output the master clock on a pin. Actually all these features are unused and only a bit for flushing the TX fifo is required. Also this series requires the DMA working on H6, a first version has been submitted by Jernej Škrabec[1] but has not been accepted yet. [1] https://patchwork.kernel.org/project/linux-arm-kernel/list/?series=89011 Changes since v3: - rename reg_fctl_ftx to val_fctl_ftx - rebase this series on sound-next - fix dt-bindings due to change in sound-next - change node name sound_spdif to sound-spdif Changes since v2: - Split quirks and H6 support patch - Add specific section for quirks comment Changes since v1: - Remove H3 compatible - Add TX fifo bit flush quirks - Add H6 bindings in SPDIF driver Clément Péron (7): dt-bindings: sound: sun4i-spdif: Add Allwinner H6 compatible ASoC: sun4i-spdif: Move quirks to the top ASoC: sun4i-spdif: Add TX fifo bit flush quirks ASoC: sun4i-spdif: Add support for H6 SoC arm64: dts: allwinner: Add SPDIF node for Allwinner H6 arm64: dts: allwinner: h6: Enable SPDIF for Beelink GS1 arm64: defconfig: Enable Sun4i SPDIF module .../sound/allwinner,sun4i-a10-spdif.yaml | 1 + .../dts/allwinner/sun50i-h6-beelink-gs1.dts | 4 ++ arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 38 ++++++++++++++ arch/arm64/configs/defconfig | 1 + sound/soc/sunxi/sun4i-spdif.c | 49 ++++++++++++++++--- 5 files changed, 87 insertions(+), 6 deletions(-) -- 2.20.1

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[alsa-devel] Is period_size bytes per period or frames per period?
by GitHub issues - edited 12 Aug '19

12 Aug '19

alsa-project/alsa-lib issue #8 was edited from crocket: According to https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html period_size in dmix is bytes per period. According to aplay --dump-hw-params -D dmix ... period_size is frames per period. Which one is right? Issue URL : https://github.com/alsa-project/alsa-lib/issues/8 Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] Is period_size bytes per period or frames per period?
by GitHub issues - opened 12 Aug '19

12 Aug '19

alsa-project/alsa-lib issue #8 was opened from crocket: According to https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html period_size in dmix is bytes per period. According to aplay --dump-hw-params period_size is frames per period. Which one is right? Issue URL : https://github.com/alsa-project/alsa-lib/issues/8 Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] [PATCH v4 0/9]ASoC: sun4i-i2s: Updates to the driver
by codekipper＠gmail.com 12 Aug '19

12 Aug '19

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, here is a patch series to add some improvements to the sun4i-i2s driver found whilst getting slave clocking and hdmi audio working on the newer SoCs. As the LibreELEC project is progressing extremely well then there has been some activity getting surround sound working and this is included. The functionality included with the new patch set has been extended to cover more sample resolutions, multi-lane data output for HDMI audio and some bug fixes that have been discovered along the way. I can see more usage of the tdm property since I last attempted to push these patches and the examples currently in mainline sort of the opposite to what I'm trying to achieve. When we first started looking at the i2s driver, the codecs that we were using allowed for the frame width to be determined based on the sampling resolution but in most use cases it seems that a fixed width is required(my highest priority should be to get HDMI audio support in). We're using the tdm property to override the old way to calculate the frame width. What I've seen in what has already been mainlined is that the i2s driver has a frame width that is fixed to 32 bits and this can be overridden using the tdm property. I still need to investigate the FIFO syncing issues which i've not had a chance to change or address the concerns that broonie and wens brought up. This change has been moved to the top of the patch stack. BR, CK --- v4 changes compared to v3 are: - Moved patches around so that the more controversial of patches are at the top of the stack. - Added more details to commit messages. - Fixed 20bit audio PCM format to use 4 bytes. - Reduced number of flags used to indicate a new SoC. v3 changes compared to v2 are: - added back slave mode changes - added back the use of tdm properties - changes to regmap and caching - removed loopback functionality - fixes to the channel offset mask v2 changes compared to v1 are: - removed slave mode changes which didn't set mclk and bclk div. - removed use of tdm and now use a dedicated property. - fix commit message to better explain reason for sign extending - add divider calculations for newer SoCs. - add support for multi-lane i2s data output. - add support for 20, 24 and 32 bit samples. - add loopback property so blocks can be tested without a codec. Marcus Cooper (9): ASoC: sun4i-i2s: Fix sun8i tx channel offset mask ASoC: sun4i-i2s: Add offset to RX channel select ASoC: sun4i-i2s: Add regmap field to sign extend sample ASoC: sun4i-i2s: Reduce quirks for sun8i-h3 ASoC: sun4i-i2s: Add set_tdm_slot functionality ASoC: sun4i-i2s: Add multi-lane functionality ASoC: sun4i-i2s: Add multichannel functionality ASoc: sun4i-i2s: Add 20, 24 and 32 bit support ASoC: sun4i-i2s: Adjust regmap settings sound/soc/sunxi/sun4i-i2s.c | 242 ++++++++++++++++++++++++------------ 1 file changed, 164 insertions(+), 78 deletions(-) -- 2.21.0

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[alsa-devel] [asoc:for-5.4 175/205] include/linux/dynamic_debug.h:82:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t'
by kbuild test robot 11 Aug '19

11 Aug '19

tree: https://kernel.googlesource.com/pub/scm/linux/kernel/git/broonie/sound.git for-5.4 head: 83ee7ab1627b7597bbb9acc26b37e9f96fff40c9 commit: abf31feea26c0f412a191c83f408311a0de7435c [175/205] ASoC: Intel: Update request-reply IPC model config: i386-randconfig-a004-201932 (attached as .config) compiler: gcc-4.9 (Debian 4.9.2-10+deb8u1) 4.9.2 reproduce: git checkout abf31feea26c0f412a191c83f408311a0de7435c # save the attached .config to linux build tree make ARCH=i386 If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All warnings (new ones prefixed by >>): In file included from include/linux/printk.h:332:0, from include/linux/kernel.h:15, from sound/soc/intel/haswell/sst-haswell-ipc.c:9: sound/soc/intel/haswell/sst-haswell-ipc.c: In function 'sst_hsw_module_set_param': >> include/linux/dynamic_debug.h:82:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t' [-Wformat=] static struct _ddebug __aligned(8) \ ^ include/linux/dynamic_debug.h:123:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^ include/linux/dynamic_debug.h:143:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^ include/linux/dynamic_debug.h:157:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^ include/linux/device.h:1503:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^ sound/soc/intel/haswell/sst-haswell-ipc.c:2018:3: note: in expansion of macro 'dev_dbg' dev_dbg(dev, "transfer parameter size : %lu\n", ^ >> include/linux/dynamic_debug.h:82:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t' [-Wformat=] static struct _ddebug __aligned(8) \ ^ include/linux/dynamic_debug.h:123:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^ include/linux/dynamic_debug.h:143:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^ include/linux/dynamic_debug.h:157:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^ include/linux/device.h:1503:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^ sound/soc/intel/haswell/sst-haswell-ipc.c:2022:3: note: in expansion of macro 'dev_dbg' dev_dbg(dev, "transfer parameter aligned size : %lu\n", ^ -- In file included from include/linux/printk.h:332:0, from include/linux/kernel.h:15, from sound/soc//intel/haswell/sst-haswell-ipc.c:9: sound/soc//intel/haswell/sst-haswell-ipc.c: In function 'sst_hsw_module_set_param': >> include/linux/dynamic_debug.h:82:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t' [-Wformat=] static struct _ddebug __aligned(8) \ ^ include/linux/dynamic_debug.h:123:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^ include/linux/dynamic_debug.h:143:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^ include/linux/dynamic_debug.h:157:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^ include/linux/device.h:1503:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^ sound/soc//intel/haswell/sst-haswell-ipc.c:2018:3: note: in expansion of macro 'dev_dbg' dev_dbg(dev, "transfer parameter size : %lu\n", ^ >> include/linux/dynamic_debug.h:82:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t' [-Wformat=] static struct _ddebug __aligned(8) \ ^ include/linux/dynamic_debug.h:123:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^ include/linux/dynamic_debug.h:143:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^ include/linux/dynamic_debug.h:157:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^ include/linux/device.h:1503:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^ sound/soc//intel/haswell/sst-haswell-ipc.c:2022:3: note: in expansion of macro 'dev_dbg' dev_dbg(dev, "transfer parameter aligned size : %lu\n", ^ vim +82 include/linux/dynamic_debug.h 923abb9d797ba0 Gal Pressman 2019-05-01 75 923abb9d797ba0 Gal Pressman 2019-05-01 76 extern __printf(3, 4) 923abb9d797ba0 Gal Pressman 2019-05-01 77 void __dynamic_ibdev_dbg(struct _ddebug *descriptor, 923abb9d797ba0 Gal Pressman 2019-05-01 78 const struct ib_device *ibdev, 923abb9d797ba0 Gal Pressman 2019-05-01 79 const char *fmt, ...); 923abb9d797ba0 Gal Pressman 2019-05-01 80 2bdde670beedf7 Rasmus Villemoes 2019-03-07 81 #define DEFINE_DYNAMIC_DEBUG_METADATA(name, fmt) \ c0d2af63786394 Joe Perches 2012-10-18 @82 static struct _ddebug __aligned(8) \ 07613b0b5ef857 Jason Baron 2011-10-04 83 __attribute__((section("__verbose"))) name = { \ 07613b0b5ef857 Jason Baron 2011-10-04 84 .modname = KBUILD_MODNAME, \ 07613b0b5ef857 Jason Baron 2011-10-04 85 .function = __func__, \ 07613b0b5ef857 Jason Baron 2011-10-04 86 .filename = __FILE__, \ 07613b0b5ef857 Jason Baron 2011-10-04 87 .format = (fmt), \ 07613b0b5ef857 Jason Baron 2011-10-04 88 .lineno = __LINE__, \ 07613b0b5ef857 Jason Baron 2011-10-04 89 .flags = _DPRINTK_FLAGS_DEFAULT, \ 2bdde670beedf7 Rasmus Villemoes 2019-03-07 90 _DPRINTK_KEY_INIT \ 07613b0b5ef857 Jason Baron 2011-10-04 91 } 07613b0b5ef857 Jason Baron 2011-10-04 92 :::::: The code at line 82 was first introduced by commit :::::: c0d2af637863940b1a4fb208224ca7acb905c39f dynamic_debug: Remove unnecessary __used :::::: TO: Joe Perches <joe(a)perches.com> :::::: CC: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] snd_pcm_start question
by Eric Desjardins 11 Aug '19

11 Aug '19

Hi, This is a simple question. I call: snd_pcm_prepare snd_pcm_writei and it works fine but I would like to delay playback start. There is snd_pcm_start for that. There is a method that is deprecated to specify a start mode but I was wondering what was the correct new way to do that? Thanks, Eric

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[alsa-devel] bcm2835.conf for RPi v. 4 incl. HDMI
by Pavel Hofman 10 Aug '19

10 Aug '19

Hi, may I ask for help with cards/bcm2835.conf for the new RPi? No such config exists yet: pi@raspberrypi:~ $ strace -f aplay -L 2>&1 | grep -i bcm2835 access("/usr/share/alsa/cards/bcm2835_alsa.conf", R_OK) = -1 ENOENT (No such file or directory) I understand the _alsa suffix is handled in aliases (e.g. https://www.raspberrypi.org/forums/viewtopic.php?t=221734#p1361104 ) The reason I am looking for this file are Pulseaudio profiles. PA requires front:X, surroundX, hdmi:X, etc. PCM devices, yet non of these are created due to the missing card config: pi@raspberrypi:~ $ aplay -L default Playback/recording through the PulseAudio sound server null Discard all samples (playback) or generate zero samples (capture) jack JACK Audio Connection Kit pulse PulseAudio Sound Server sysdefault:CARD=ALSA bcm2835 ALSA, bcm2835 ALSA Default Audio Device dmix:CARD=ALSA,DEV=0 bcm2835 ALSA, bcm2835 ALSA Direct sample mixing device dmix:CARD=ALSA,DEV=1 bcm2835 ALSA, bcm2835 IEC958/HDMI Direct sample mixing device dsnoop:CARD=ALSA,DEV=0 bcm2835 ALSA, bcm2835 ALSA Direct sample snooping device dsnoop:CARD=ALSA,DEV=1 bcm2835 ALSA, bcm2835 IEC958/HDMI Direct sample snooping device hw:CARD=ALSA,DEV=0 bcm2835 ALSA, bcm2835 ALSA Direct hardware device without any conversions hw:CARD=ALSA,DEV=1 bcm2835 ALSA, bcm2835 IEC958/HDMI Direct hardware device without any conversions plughw:CARD=ALSA,DEV=0 bcm2835 ALSA, bcm2835 ALSA Hardware device with all software conversions plughw:CARD=ALSA,DEV=1 bcm2835 ALSA, bcm2835 IEC958/HDMI Hardware device with all software conversions usbstream:CARD=ALSA bcm2835 ALSA USB Stream Output As a result, only the Analog Mono PA profile (based on the only existing hw:0) is available in PA: D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Mono (analog-mono) D: [pulseaudio] alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... D: [pulseaudio] alsa-util.c: Managed to open hw:0 ... D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Stereo (analog-stereo) D: [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... I: [pulseaudio] (alsa-lib)conf.c: Unknown parameters 0 I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM front:0 --- D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo) D: [pulseaudio] alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ... I: [pulseaudio] (alsa-lib)conf.c: Unknown parameters 0 I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM hdmi:0 ... D: [pulseaudio] alsa-mixer.c: Checking for playback on Digital Surround 7.1 (HDMI) (hdmi-surround71) D: [pulseaudio] alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ... I: [pulseaudio] (alsa-lib)conf.c: Unknown parameters 0 I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM hdmi:0 RPi v.4 has 2 HDMI outputs: pi@raspberrypi:~ $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: ALSA [bcm2835 ALSA], device 0: bcm2835 ALSA [bcm2835 ALSA] Subdevices: 6/7 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 card 0: ALSA [bcm2835 ALSA], device 1: bcm2835 IEC958/HDMI [bcm2835 IEC958/HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: ALSA [bcm2835 ALSA], device 2: bcm2835 IEC958/HDMI1 [bcm2835 IEC958/HDMI1] Subdevices: 1/1 Subdevice #0: subdevice #0 Each is 2 - 8 channel: pi@raspberrypi:~ $ aplay -v --dump-hw-params -D hw:0,1 /dev/zero Playing raw data '/dev/zero' : Unsigned 8 bit, Rate 8000 Hz, Mono HW Params of device "hw:0,1": -------------------- ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: [32 128] CHANNELS: [2 8] RATE: [44100 192000] PERIOD_TIME: [10000 743039) PERIOD_SIZE: [441 32768] PERIOD_BYTES: [1776 524288] PERIODS: [1 75) BUFFER_TIME: (2296 743039) BUFFER_SIZE: [441 32768] BUFFER_BYTES: [1764 131072] TICK_TIME: ALL I looked at hdmi section of cards/Intel-HDA.conf but that one is very complicated. May I ask for help how the config should look like to work both for multichannel PCM and digital formats (i.e. support for the AES vars)? I very much appreciate any help. Best regards, Pavel.

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