June 2021 - Alsa-devel - mailman.alsa-project.org

Re: [PATCH 0/2] ALSA: usb-audio: scarlett2: Read all configuration at init time
by Takashi Iwai 07 Jun '21

07 Jun '21

On Mon, 07 Jun 2021 17:00:10 +0200, Vladimir Sadovnikov wrote: > > Hello! > > I would like to say some words from my side. > > The Scarlett device (especially 18i20) is pretty complicated device > and holds a lot of settings in it's internal > configuration area (hardware and software). > > So this is not the only patch which will configure the driver in proper way. > Since the device stores it's internal state (and that's good for power > safety and mobility), ideally, we should get > the almost fully compatible mixer settings with the original Focusrite > Control Software. > > The huge amount of job I've already done i my fork of Geoffrey's driver: > https://github.com/sadko4u/focusrite-scarlett-backports/blob/master/prod-dr… > > So we're planning to work on integrating our changes into the common > patch sets and will submit changes here. Sure, I don't mean against the patches, this looks like an acceptable approach. So don't worry, I'd take the patches once when the fixed version is submitted. However, from the system design POV, all those configurations should be a software issue, and ideally we shouldn't rely on the hardware preset state which has been done *somehow* -- it may allow malfunction easily. One thing I've learned over years is that you can never trust hardware :) Takashi > > Best, > Vladimir > > 07.06.2021 10:23, Takashi Iwai пишет: > > On Sun, 06 Jun 2021 16:16:44 +0200, > > Geoffrey D. Bennett wrote: > >> These two patches add support for reading the mixer volumes and mux > >> configuration from the hardware when the driver is initialising. > >> > >> Previously the ALSA volume controls were initialised to zero and the > >> mux configuration set to a fixed default instead of being initialised > >> to match the hardware state. > >> > >> The ALSA controls for the Scarlett Gen 2 interfaces should now always > >> be in sync with the hardware. Thanks to Vladimir Sadovnikov for > >> figuring out how to do this. > >> > >> Takashi, if these pass your review, I believe that they are > >> appropriate for: > >> #Cc: stable(a)vger.kernel.org > > Well, in general, having a proper fixed value for the initial mixer > > value is the right thing, which is a part of the driver's role. > > Though, in snd-usb-audio, we don't set up the initial values just > > because of laziness; since the topology in USB audio is variable per > > device and often hard to parse correctly, it's difficult to determine > > the suitable initial values, hence we leave untouched. So, in that > > sense, setting the zero isn't wrong, rather safer, per se. > > > > However, Scarlett 2 seems to want to be different; it has already some > > initialization code to read the existing configs. So this change > > sounds more or less acceptable. But it's questionable whether it's > > really for stable as a "fix". > > > > In anyway, please fix the bug ktest bot spotted, the missing endian > > conversions and resubmit. > > > > > > thanks, > > > > Takashi > >

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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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