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alsa-lib 1.2.5(-2 on arch) only one of two connected hdmi/displayport outputs visible at once. My mic is completely MIA
by GitHub issues - opened 07 Jun '21

07 Jun '21

alsa-project/alsa-lib issue #148 was opened from fkroener: Since the recent update to 1.2.5-2 on archlinux, only one of my two usb-c/displayport/hdmi devices show up in pavucontrol and kde mixer. I am able to switch between hdmi0 and hdmi1, by changing the Profile in pavucontrol of HD-Audio Generic. The microphone doesn't show up at all now. ![KDE Mixer](https://user-images.githubusercontent.com/28690007/121057421-8c36938… This doesn't appear related to #142 or #143, since I have revision 2 from arch installed, which should include the fixes. This is a Lenovo T14 AMD. ``` pactl list cards Card #0 Name: alsa_card.pci-0000_07_00.1 Driver: module-alsa-card.c Owner Module: 6 Properties: alsa.card = "0" alsa.card_name = "HD-Audio Generic" alsa.long_card_name = "HD-Audio Generic at 0xfd3c8000 irq 91" alsa.driver_name = "snd_hda_intel" device.bus_path = "pci-0000:07:00.1" sysfs.path = "/devices/pci0000:00/0000:00:08.1/0000:07:00.1/sound/card0" device.bus = "pci" device.vendor.id = "1002" device.vendor.name = "Advanced Micro Devices, Inc. [AMD/ATI]" device.product.id = "1637" device.string = "0" device.description = "HD-Audio Generic" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" Profiles: output:hdmi-stereo: Digital Stereo (HDMI) Output (sinks: 1, sources: 0, priority: 5900, available: yes) output:hdmi-stereo-extra1: Digital Stereo (HDMI 2) Output (sinks: 1, sources: 0, priority: 5700, available: yes) output:hdmi-stereo-extra2: Digital Stereo (HDMI 3) Output (sinks: 1, sources: 0, priority: 5700, available: no) output:hdmi-surround-extra2: Digital Surround 5.1 (HDMI 3) Output (sinks: 1, sources: 0, priority: 600, available: no) output:hdmi-surround71-extra2: Digital Surround 7.1 (HDMI 3) Output (sinks: 1, sources: 0, priority: 600, available: no) off: Off (sinks: 0, sources: 0, priority: 0, available: yes) Active Profile: output:hdmi-stereo Ports: hdmi-output-0: HDMI / DisplayPort (type: HDMI, priority: 5900, latency offset: 0 usec, available) Properties: device.icon_name = "video-display" device.product.name = "P27h-20" Part of profile(s): output:hdmi-stereo hdmi-output-1: HDMI / DisplayPort 2 (type: HDMI, priority: 5800, latency offset: 0 usec, available) Properties: device.icon_name = "video-display" device.product.name = "P27h-20" Part of profile(s): output:hdmi-stereo-extra1 hdmi-output-2: HDMI / DisplayPort 3 (type: HDMI, priority: 5700, latency offset: 0 usec, not available) Properties: device.icon_name = "video-display" Part of profile(s): output:hdmi-stereo-extra2, output:hdmi-surround-extra2, output:hdmi-surround71-extra2 Card #1 Name: alsa_card.pci-0000_07_00.6 Driver: module-alsa-card.c Owner Module: 8 Properties: alsa.card = "1" alsa.card_name = "HD-Audio Generic" alsa.long_card_name = "HD-Audio Generic at 0xfd3c0000 irq 92" alsa.driver_name = "snd_hda_intel" device.bus_path = "pci-0000:07:00.6" sysfs.path = "/devices/pci0000:00/0000:00:08.1/0000:07:00.6/sound/card1" device.bus = "pci" device.vendor.id = "1022" device.vendor.name = "Advanced Micro Devices, Inc. [AMD]" device.product.id = "15e3" device.product.name = "Family 17h (Models 10h-1fh) HD Audio Controller" device.string = "1" device.description = "Family 17h (Models 10h-1fh) HD Audio Controller" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" Profiles: input:analog-stereo: Analog Stereo Input (sinks: 0, sources: 1, priority: 65, available: no) output:analog-stereo: Analog Stereo Output (sinks: 1, sources: 0, priority: 6500, available: yes) output:analog-stereo+input:analog-stereo: Analog Stereo Duplex (sinks: 1, sources: 1, priority: 6565, available: no) off: Off (sinks: 0, sources: 0, priority: 0, available: yes) Active Profile: output:analog-stereo Ports: analog-input-mic: Microphone (type: Mic, priority: 8700, latency offset: 0 usec, not available) Properties: device.icon_name = "audio-input-microphone" Part of profile(s): input:analog-stereo, output:analog-stereo+input:analog-stereo analog-output-speaker: Speakers (type: Speaker, priority: 10000, latency offset: 0 usec, availability unknown) Properties: device.icon_name = "audio-speakers" Part of profile(s): output:analog-stereo, output:analog-stereo+input:analog-stereo analog-output-headphones: Headphones (type: Headphones, priority: 9900, latency offset: 0 usec, not available) Properties: device.icon_name = "audio-headphones" Part of profile(s): output:analog-stereo, output:analog-stereo+input:analog-stereo ``` If you need any more information, I'm gladly at your disposal. Thanks! Issue URL : https://github.com/alsa-project/alsa-lib/issues/148 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH AUTOSEL 5.12 27/49] ALSA: hda: Add AlderLake-M PCI ID
by Sasha Levin 07 Jun '21

07 Jun '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 4ad7935df6a566225c3d51900bde8f2f0f8b6de3 ] Add HD Audio PCI ID for Intel AlderLake-M. Add rules to snd_intel_dsp_find_config() to choose SOF driver for ADL-M systems with PCH-DMIC or Soundwire codecs, and legacy driver for the rest. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210528185123.48332-1-kai.vehmanen@linux.intel.c… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/intel-dsp-config.c | 4 ++++ sound/pci/hda/hda_intel.c | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index ab5ff7867eb9..d8be146793ee 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -331,6 +331,10 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x51c8, }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x51cc, + }, #endif }; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 79ade335c8a0..470753b36c8a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2485,6 +2485,9 @@ static const struct pci_device_id azx_ids[] = { /* Alderlake-P */ { PCI_DEVICE(0x8086, 0x51c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Alderlake-M */ + { PCI_DEVICE(0x8086, 0x51cc), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, -- 2.30.2

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[PATCH 0/9] ALSA: firewire: cease from delayed card registration
by Takashi Sakamoto 07 Jun '21

07 Jun '21

Hi, Since Linux kernel v4.7, delayed card registration was introduced to all drivers in ALSA firewire stack. Nowadays it brings less benefit than code complication. This patchset ceases from it. Takashi Sakamoto (9): ALSA: bebob: cease from delayed card registration ALSA: fireworks: cease from delayed card registration ALSA: oxfw: cease from delayed card registration ALSA: dice: cease from delayed card registration ALSA: firewire-digi00x: cease from delayed card registration ALSA: firewire-tascam: cease from delayed card registration ALSA: firewire-motu: cease from delayed card registration ALSA: fireface: cease from delayed card registration ALSA: firewire-lib: delete unused kernel API sound/firewire/bebob/bebob.c | 159 ++++++++++----------------- sound/firewire/bebob/bebob.h | 4 - sound/firewire/dice/dice.c | 138 ++++++++--------------- sound/firewire/dice/dice.h | 4 - sound/firewire/digi00x/digi00x.c | 101 +++++------------ sound/firewire/digi00x/digi00x.h | 3 - sound/firewire/fireface/ff.c | 90 +++++---------- sound/firewire/fireface/ff.h | 3 - sound/firewire/fireworks/fireworks.c | 107 ++++++------------ sound/firewire/fireworks/fireworks.h | 3 - sound/firewire/lib.c | 32 ------ sound/firewire/lib.h | 3 - sound/firewire/motu/motu.c | 84 +++++--------- sound/firewire/motu/motu.h | 3 - sound/firewire/oxfw/oxfw.c | 128 ++++++++------------- sound/firewire/oxfw/oxfw.h | 6 +- sound/firewire/tascam/tascam.c | 92 +++++----------- sound/firewire/tascam/tascam.h | 2 - 18 files changed, 296 insertions(+), 666 deletions(-) -- 2.27.0

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[PATCH v2] ASoC: qcom: Fix for DMA interrupt clear reg overwriting
by Srinivasa Rao Mandadapu 07 Jun '21

07 Jun '21

The DMA interrupt clear register overwritten during simultaneous playback and capture in lpass platform interrupt handler. It's causing playback or capture stuck in similtaneous plaback on speaker and capture on dmic test. Update appropriate reg fields of corresponding channel instead of entire register write. Fixes: commit c5c8635a04711 ("ASoC: qcom: Add LPASS platform driver") Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-platform.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 0df9481ea4c6..f220a2739ac3 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -526,7 +526,7 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, return -EINVAL; } - ret = regmap_write(map, reg_irqclr, val_irqclr); + ret = regmap_update_bits(map, reg_irqclr, val_irqclr, val_irqclr); if (ret) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", ret); return ret; @@ -650,7 +650,7 @@ static irqreturn_t lpass_dma_interrupt_handler( struct lpass_variant *v = drvdata->variant; irqreturn_t ret = IRQ_NONE; int rv; - unsigned int reg = 0, val = 0; + unsigned int reg, val, val_clr, val_mask; struct regmap *map; unsigned int dai_id = cpu_dai->driver->id; @@ -676,8 +676,9 @@ static irqreturn_t lpass_dma_interrupt_handler( return -EINVAL; } if (interrupts & LPAIF_IRQ_PER(chan)) { - - rv = regmap_write(map, reg, LPAIF_IRQ_PER(chan) | val); + val_clr = LPAIF_IRQ_PER(chan) | val; + val_mask = LPAIF_IRQ_ALL(chan); + rv = regmap_update_bits(map, reg, val_mask, val_clr); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); @@ -688,7 +689,9 @@ static irqreturn_t lpass_dma_interrupt_handler( } if (interrupts & LPAIF_IRQ_XRUN(chan)) { - rv = regmap_write(map, reg, LPAIF_IRQ_XRUN(chan) | val); + val_clr = (LPAIF_IRQ_XRUN(chan) | val); + val_mask = LPAIF_IRQ_ALL(chan); + rv = regmap_update_bits(map, reg, val_mask, val_clr); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); @@ -700,7 +703,9 @@ static irqreturn_t lpass_dma_interrupt_handler( } if (interrupts & LPAIF_IRQ_ERR(chan)) { - rv = regmap_write(map, reg, LPAIF_IRQ_ERR(chan) | val); + val_clr = (LPAIF_IRQ_ERR(chan) | val); + val_mask = LPAIF_IRQ_ALL(chan); + rv = regmap_update_bits(map, reg, val_mask, val_clr); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH] ASoC: Remove unneeded semicolon in rk817_codec.c
by Wan Jiabing 07 Jun '21

07 Jun '21

Fix following coccicheck warning: ./sound/soc/codecs/rk817_codec.c:49:2-3: Unneeded semicolon Signed-off-by: Wan Jiabing <wanjiabing(a)vivo.com> --- sound/soc/codecs/rk817_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index fd3a5ba034a9..414d21f22a05 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -46,7 +46,7 @@ static int rk817_init(struct snd_soc_component *component) if (rk817->mic_in_differential) { snd_soc_component_update_bits(component, RK817_CODEC_AMIC_CFG0, MIC_DIFF_MASK, MIC_DIFF_EN); - }; + } return 0; } -- 2.20.1

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Possible ALSA bug: Cannot record via mic while PA is running
by Meghdad Moradi 07 Jun '21

07 Jun '21

Hello, Please take a look at the following thread that I created some time ago: [1]https://ubuntuforums.org/showthread.php?t=2462882&p=14041574#post140 41574 I have included most relevant and useful info in the first and last post. pa-info output: [2]https://gist.github.com/no149/5c239626048a860a882f8d412013a4e6 I'd provide further info, if it's needed. References 1. https://ubuntuforums.org/showthread.php?t=2462882&p=14041574#post14041574 2. https://gist.github.com/no149/5c239626048a860a882f8d412013a4e6

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[RFC][PATCH 0/3] ALSA: pcm/firewire: allow to queue period elapse event in process context
by Takashi Sakamoto 07 Jun '21

07 Jun '21

Hi, All of drivers in ALSA firewire stack processes two chances to process isochronous packets in any isochronous context; in software IRQ context for 1394 OHCI, and in process context of ALSA PCM application. In the process context, callbacks of .pointer and .ack are utilized. The callbacks are done by ALSA PCM core under acquiring lock of PCM substream, In design of ALSA PCM core, call of snd_pcm_period_elapsed() is used for drivers to awaken user processes from waiting for available frames. The function voluntarily acquires lock of PCM substream, therefore it is not called in the process context since it causes dead lock. As a workaround to avoid the dead lock, all of drivers in ALSA firewire stack uses workqueue to delegate the call. This patchset is my attempt for the issue. A variant of 'snd_pcm_period_elapsed()' without lock acquisition is going to be added, named 'snd_pcm_period_elapsed_without_lock()'. This is used in callbacks of .pointer and .ack of snd_pcm_ops structure. The patchset is still under my test, but it looks to work well in my easy and rough test. Before posting for merge, I'd like to get your comment to the idea. When evaluating, please merge below two histories: * 64584f329352 (for-next) * 9981b20a5e36 (for-linus) Takashi Sakamoto (3): ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream ALSA: firewire-lib: queue event of period elapse in process context ALSA: firewire-lib: obsolete workqueue for period update include/sound/pcm.h | 53 ++++++++++++++++++++++++++++++++++- sound/core/pcm_lib.c | 25 +++-------------- sound/firewire/amdtp-stream.c | 46 +++++++++--------------------- sound/firewire/amdtp-stream.h | 1 - 4 files changed, 70 insertions(+), 55 deletions(-) -- 2.27.0

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New lavc api
by GitHub pull_request - opened 06 Jun '21

06 Jun '21

alsa-project/alsa-plugins pull request #23 was opened from jamrial: This is untested (I don't have access to a system with alsa, but the change was straightforward enough that i could write it). It is unlikely that it will fix the silence issue described in #22, but it nonetheless is a change required in order to support future versions of ffmpeg. The silence is, as far as i could tell, related to the fact that starting with ffmpeg 4.4, the ac3_fixed encoder takes S32 planar as input, versus the S16 planar it used to. This plugin should look at the AVCodec sample_fmts[0] value and feed the encoder the appropriate data. Request URL : https://github.com/alsa-project/alsa-plugins/pull/23 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/23.patch Repository URL: https://github.com/alsa-project/alsa-plugins

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[PATCH 0/8] ALSA: usb-audio: scarlett2: cleanup
by Geoffrey D. Bennett 06 Jun '21

06 Jun '21

Hi Takashi, Over the next few weeks I will have ready about 40 patches to clean up the Scarlett Gen 2 mixer driver a bit, add a few missing features, and add support for all Scarlett Gen 3 models. I was planning on submitting them in batches of 5-10 at a time; is that okay? Here are the first 8 patches which are ready to go; only small cleanups/renames/fixes. They are intended to be applied after these two patches that I submitted earlier: - ALSA: usb-audio: scarlett2: Read mixer volumes at init time - ALSA: usb-audio: scarlett2: Read mux at init time Thanks, Geoffrey. Geoffrey D. Bennett (8): ALSA: usb-audio: scarlett2: Remove incorrect S/PDIF comment ALSA: usb-audio: scarlett2: Coding style improvements ALSA: usb-audio: scarlett2: Remove unused/useless code ALSA: usb-audio: scarlett2: Remove interrupt debug message ALSA: usb-audio: scarlett2: Rename buttons/interrupts ALSA: usb-audio: scarlett2: Rename struct scarlett2_mixer_data ALSA: usb-audio: scarlett2: Add temp variable for consistency ALSA: usb-audio: scarlett2: Fix data_mutex lock sound/usb/mixer_scarlett_gen2.c | 229 ++++++++++++++------------------ 1 file changed, 102 insertions(+), 127 deletions(-) -- 2.31.1

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a52: not compatible with recent avcodec
by GitHub issues - opened 06 Jun '21

06 Jun '21

alsa-project/alsa-plugins issue #22 was opened from manio: Hi! Current a52 plugin is not compatible with recent changes in the FFmpeg lib. The symptom is that using newest FFmpeg the a52 plugin produces silent in the speakers. During the weekend I was able to bisect the problem and shed some light on this, and here is what I found out: 1. The a52 plugin is working fine before commit: https://github.com/FFmpeg/FFmpeg/commit/93016f5d1d280f9cb7856883af287fa66af… 2. Above commit is wrapping avcodec_encode_audio2(), which a52 plugin is using, into new API internally inside the avcodec 3. All applications should move to new so called ["avcodec_send_frame() API"](https://ffmpeg.org/doxygen/3.4/group__lavc__encdec.html) 4. After the commit in point 1 the a52 output plugin is not working anymore I was asking the commit author @jamrial about the commit and he said that the main culprit could rather be the commit: https://github.com/FFmpeg/FFmpeg/commit/2d85e6e7230406e33e61d903b9c29ab562e… which is changing the S16 to S32 for ac3_fixed encoder while the a52 plugin _"is ignoring the kind of format the encoder takes as input (Starting from the aforementioned commit, ac3_fixed takes S32 planar audio), and unconditionally feeding it S16 planar audio."_ I was trying to change the a52 from S16 to S32 but I think some more work is needed on this... I was also try to change the do_encode() function from avcodec_encode_audio2 to avcodec_send_frame() with no luck... Issue URL : https://github.com/alsa-project/alsa-plugins/issues/22 Repository URL: https://github.com/alsa-project/alsa-plugins

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