December 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: hda/realtek: Add quirk for MSI-GP73
by Takashi Iwai 20 Dec '20

20 Dec '20

MSI-GP73 (with SSID 1462:1229) requires yet again ALC1220_FIXUP_CLEVO_P950 quirk like other MSI models. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=210793 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c203eba8d9ff..5478a89d94e3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2516,6 +2516,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x11f7, "MSI-GE63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1462, 0x1229, "MSI-GP73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), -- 2.26.2

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[RFC][PATCH v2 0/1] design a way to change audio Jack state by software
by Hui Wang 19 Dec '20

19 Dec '20

the changes in the V2: - using debugfs instead of sysfs - using jack_ctrl to create a folder instead of snd_jack, since ASoC drivers could create multi jack_ctrls within a snd_jack - create a folder for each jack_ctrl instead for all jack_ctrls This is the layout of folder on a machine with 2 sound cards: root@hwang4-ThinkPad-P1-Gen-3:/sys/kernel/debug# tree /sys/kernel/debug/sound-core/ /sys/kernel/debug/sound-core/ ├── card0 │ ├── HDMI!DP,pcm=10 Jack │ │ ├── jackin_inject │ │ └── sw_inject_enable │ ├── HDMI!DP,pcm=11 Jack │ │ ├── jackin_inject │ │ └── sw_inject_enable │ ├── HDMI!DP,pcm=3 Jack │ │ ├── jackin_inject │ │ └── sw_inject_enable │ ├── HDMI!DP,pcm=7 Jack │ │ ├── jackin_inject │ │ └── sw_inject_enable │ ├── HDMI!DP,pcm=8 Jack │ │ ├── jackin_inject │ │ └── sw_inject_enable │ └── HDMI!DP,pcm=9 Jack │ ├── jackin_inject │ └── sw_inject_enable └── card1 ├── HDMI!DP,pcm=3 Jack │ ├── jackin_inject │ └── sw_inject_enable ├── HDMI!DP,pcm=4 Jack │ ├── jackin_inject │ └── sw_inject_enable ├── HDMI!DP,pcm=5 Jack │ ├── jackin_inject │ └── sw_inject_enable ├── Headphone Jack │ ├── jackin_inject │ └── sw_inject_enable ├── Headset Jack │ ├── jackin_inject │ └── sw_inject_enable └── Mic Jack ├── jackin_inject └── sw_inject_enable The sw_inject_enable has rw mode, the jackin_inject has write_only mode. If users want to inject a plugin to Headphone Jack, they could run: echo 1 > 'Headphone Jack'/sw_inject_enable /* now, this jack state can't be changed by hw events */ cat 'Headphone Jack'/sw_inject_enable ack: Headphone Jack Inject Enabled: 1 echo 1 > 'Headphone Jack'/jackin_inject /* this will inject a plugin to Headphone Jack */ After testing, run echo 0 > 'Headphone Jack'/sw_inject_enable /* the hw events will change the jack state */ Hui Wang (1): alsa: jack: implement software jack injection via debugfs include/sound/core.h | 2 + sound/core/init.c | 7 ++ sound/core/jack.c | 202 ++++++++++++++++++++++++++++++++++++------- sound/core/sound.c | 8 ++ 4 files changed, 188 insertions(+), 31 deletions(-) -- 2.25.1

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[PATCH] Fix mic sound on Jieli webcam
by Marco Giunta 19 Dec '20

19 Dec '20

Hi, recently I've bought a usb webcam with integrated mic: Jieli Technology USB PHY 2.0 (1224:2a25) The video part works well, but the mic sound is speedups, "like minions" (cit.). When I connect the camera, these are the dmesg messages: kernel: usb 1-8: current rate 0 is different from the runtime rate 8000 kernel: usb 1-8: current rate 0 is different from the runtime rate 16000 kernel: usb 1-8: current rate 0 is different from the runtime rate 44100 kernel: usb 1-8: current rate 0 is different from the runtime rate 48000 kernel: usb 1-8: Warning! Unlikely big volume range (=4096), cval->res is probably wrong. kernel: usb 1-8: [3] FU [Mic Capture Volume] ch = 1, val = 0/4096/1 kernel: usbcore: registered new interface driver snd-usb-audio and after a while, dmesg log is filled, every 5 seconds, with: kernel: retire_capture_urb: 84 callbacks suppressed kernel: retire_capture_urb: 1714 callbacks suppressed A guy reports on ArchLinux bug website the same problem (https://bugs.archlinux.org/task/68141?opened=12995&status%5B0%5D=) and provides a patch to fix the sound issue. I've applied the patch on kernel 5.9.13 (Fedora 33 x86_64) and now the mic works, no more minions voice effect. Now dmesg messages are only: kernel: usb 1-8: Warning! Unlikely big volume range (=4096), cval->res is probably wrong. kernel: usb 1-8: [3] FU [Mic Capture Volume] ch = 1, val = 0/4096/1 kernel: usbcore: registered new interface driver snd-usb-audio the retire_capture_urb messages are gone. All credits for the patch go to him but I don't know how to contact that guy nor I don't know if he has already contacted you, so my question is if you could review his patch and finally apply upstream. If you need other information or you need a tester, I'm here. Thanks, Marco --- a/sound/usb/format.c 2020-10-01 18:36:35.000000000 +0300 +++ b/sound/usb/format.c 2020-10-04 02:10:21.678685952 +0300 @@ -217,6 +217,21 @@ (chip->usb_id == USB_ID(0x041e, 0x4064) || chip->usb_id == USB_ID(0x041e, 0x4068))) rate = 8000; + + // hack for "Jieli Technology USB PHY 2.0" webcam + if (chip->usb_id == USB_ID(0x1224, 0x2a25)) { + switch (rate) { + case 8000: + fp->datainterval += 4; + break; + case 16000: + fp->datainterval += 3; + break; + default: + fp->datainterval += 1; + break; + } + } fp->rate_table[fp->nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) --- a/sound/usb/endpoint.c 2020-10-01 18:36:35.000000000 +0300 +++ b/sound/usb/endpoint.c 2020-10-04 02:09:09.471978982 +0300 @@ -882,6 +882,8 @@ if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) { packs_per_ms = 8 >> ep->datainterval; max_packs_per_urb = MAX_PACKS_HS; + if (!packs_per_ms) + packs_per_ms = 1; } else { packs_per_ms = 1; max_packs_per_urb = MAX_PACKS; --- a/sound/usb/quirks.c 2020-10-01 18:36:35.000000000 +0300 +++ b/sound/usb/quirks.c 2020-10-04 02:14:04.532196519 +0300 @@ -1516,6 +1516,7 @@ case USB_ID(0x1901, 0x0191): /* GE B850V3 CP2114 audio interface */ case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */ case USB_ID(0x2912, 0x30c8): /* Audioengine D1 */ + case USB_ID(0x1224, 0x2a25): /* Jieli Technology USB PHY 2.0 */ return true; }

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[PATCH] ALSA: pcm: oss: Fix a few more UBSAN fixes
by Takashi Iwai 18 Dec '20

18 Dec '20

There are a few places that call round{up|down}_pow_of_two() with the value zero, and this causes undefined behavior warnings. Avoid calling those macros if such a nonsense value is passed; it's a minor optimization as well, as we handle it as either an error or a value to be skipped, instead. Reported-by: syzbot+33ef0b6639a8d2d42b4c(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_oss.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index de1917484647..142fc751a847 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -693,6 +693,8 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, oss_buffer_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, NULL)) * oss_frame_size; + if (!oss_buffer_size) + return -EINVAL; oss_buffer_size = rounddown_pow_of_two(oss_buffer_size); if (atomic_read(&substream->mmap_count)) { if (oss_buffer_size > runtime->oss.mmap_bytes) @@ -728,17 +730,21 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, min_period_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_min(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL)); - min_period_size *= oss_frame_size; - min_period_size = roundup_pow_of_two(min_period_size); - if (oss_period_size < min_period_size) - oss_period_size = min_period_size; + if (min_period_size) { + min_period_size *= oss_frame_size; + min_period_size = roundup_pow_of_two(min_period_size); + if (oss_period_size < min_period_size) + oss_period_size = min_period_size; + } max_period_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL)); - max_period_size *= oss_frame_size; - max_period_size = rounddown_pow_of_two(max_period_size); - if (oss_period_size > max_period_size) - oss_period_size = max_period_size; + if (max_period_size) { + max_period_size *= oss_frame_size; + max_period_size = rounddown_pow_of_two(max_period_size); + if (oss_period_size > max_period_size) + oss_period_size = max_period_size; + } oss_periods = oss_buffer_size / oss_period_size; -- 2.26.2

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[PATCH] ALSA: pcm: Remove snd_pcm_lib_preallocate_dma_free()
by Lars-Peter Clausen 18 Dec '20

18 Dec '20

Since commit d4cfb30fce03 ("ALSA: pcm: Set per-card upper limit of PCM buffer allocations") snd_pcm_lib_preallocate_dma_free() is a single line function that has one caller, which is another single line function. Clean this up a bit and remove snd_pcm_lib_preallocate_dma_free() and directly call do_free_pages() from snd_pcm_lib_preallocate_free(). This is a bit less boilerplate. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/core/pcm_memory.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 4f03ba8ed0ae..ee6e9c5eec45 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -89,14 +89,6 @@ static int preallocate_pcm_pages(struct snd_pcm_substream *substream, size_t siz return 0; } -/* - * release the preallocated buffer if not yet done. - */ -static void snd_pcm_lib_preallocate_dma_free(struct snd_pcm_substream *substream) -{ - do_free_pages(substream->pcm->card, &substream->dma_buffer); -} - /** * snd_pcm_lib_preallocate_free - release the preallocated buffer of the specified substream. * @substream: the pcm substream instance @@ -105,7 +97,7 @@ static void snd_pcm_lib_preallocate_dma_free(struct snd_pcm_substream *substream */ void snd_pcm_lib_preallocate_free(struct snd_pcm_substream *substream) { - snd_pcm_lib_preallocate_dma_free(substream); + do_free_pages(substream->pcm->card, &substream->dma_buffer); } /** -- 2.20.1

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[PATCH 0/2] ALSA: Clear PCM buffers more properly
by Takashi Iwai 18 Dec '20

18 Dec '20

Hi, here are two small fix patches for PCM buffer allocations. Basically those address two things that have been discussed in the recent thread [1]: * Clear the remaining bytes in PCM buffer pages that are exposed via mmap * Assure the page aligned allocations for genalloc This doesn't cover the cases where non-standard PCM buffer allocation is used. Those should handle the buffer clearance by themselves. Takashi [1] https://lore.kernel.org/r/05c824e5-0c33-4182-26fa-b116a42b10d6@metafoo.de === Takashi Iwai (2): ALSA: memalloc: Align buffer allocations in page size ALSA: pcm: Clear the full allocated memory at hw_params sound/core/memalloc.c | 1 + sound/core/pcm_native.c | 9 +++++++-- 2 files changed, 8 insertions(+), 2 deletions(-) -- 2.26.2

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[PATCH] ALSA: usb-audio: Disable sample read check if firmware doesn't give back
by Takashi Iwai 18 Dec '20

18 Dec '20

Some buggy firmware don't give the current sample rate but leaves zero. Handle this case more gracefully without warning but just skip the current rate verification from the next time. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/clock.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index e940dcee792b..31051f2be46d 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -534,6 +534,12 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, } crate = data[0] | (data[1] << 8) | (data[2] << 16); + if (!crate) { + dev_info(&dev->dev, "failed to read current rate; disabling the check\n"); + chip->sample_rate_read_error = 3; /* three strikes, see above */ + return 0; + } + if (crate != rate) { dev_warn(&dev->dev, "current rate %d is different from the runtime rate %d\n", crate, rate); // runtime->rate = crate; -- 2.26.2

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Re: [PATCH 06/18] arm64: dts: renesas: beacon: Configure Audio CODEC clocks
by Geert Uytterhoeven 18 Dec '20

18 Dec '20

Hi Adam, CC alsa-devel On Sun, Dec 13, 2020 at 7:38 PM Adam Ford <aford173(a)gmail.com> wrote: > With the newly added configurable clock options, the audio CODEC can > configure the mclk automatically. Add the reference to the versaclock. > Since the devices on I2C5 can communicate at 400KHz, let's also increase > that too > > Signed-off-by: Adam Ford <aford173(a)gmail.com> Thanks for your patch! > --- a/arch/arm64/boot/dts/renesas/beacon-renesom-baseboard.dtsi > +++ b/arch/arm64/boot/dts/renesas/beacon-renesom-baseboard.dtsi > @@ -424,13 +424,15 @@ &i2c0 { > > &i2c5 { > status = "okay"; > - clock-frequency = <100000>; > + clock-frequency = <400000>; > pinctrl-0 = <&i2c5_pins>; > pinctrl-names = "default"; > > codec: wm8962@1a { > compatible = "wlf,wm8962"; > reg = <0x1a>; > + clocks = <&versaclock6_bb 3>; > + clock-names = "mclk"; While the driver does get the (nameless) clock, the DT bindings lack any mention of a clocks property. It would be good to update the bindings. Note that arch/arm/boot/dts/imx6-logicpd-baseboard.dtsi and arch/arm64/boot/dts/freescale/imx8mm-beacon-baseboard.dtsi (both by your hand) use "xclk" instead of "mclk"? > DCVDD-supply = <&reg_audio>; > DBVDD-supply = <&reg_audio>; > AVDD-supply = <&reg_audio>; 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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Move SDM845 out of Qualcomm folder and add C360
by GitHub pull_request - opened 18 Dec '20

18 Dec '20

alsa-project/alsa-ucm-conf pull request #70 was opened from Srinivas-Kandagatla: With driver_name fix in sdm845 machine driver https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… /proc/asound/cards get populated correctly with proper driver name and card long name. With this in place we should be able to rearrange sdm845 directory to be inline with others and reuse some of the common code. Without this all the sdm845 based machines will get a generic ucm description which will not work, one of such instance is Lenovo C360 Laptop. This patchset also add support to Lenovo C360 support Thanks, srini Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/70 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/70.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ASoC: meson: axg-tdmin: fix axg skew offset
by Jerome Brunet 18 Dec '20

18 Dec '20

The signal captured on from tdm decoder of the AXG SoC is incorrect. It appears amplified. The skew offset of the decoder is wrong. Setting the skew offset to 3, like the g12 and sm1 SoCs, solves and gives correct data. Fixes: 13a22e6a98f8 ("ASoC: meson: add tdm input driver") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-tdmin.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c index 88ed95ae886b..b4faf9d5c1aa 100644 --- a/sound/soc/meson/axg-tdmin.c +++ b/sound/soc/meson/axg-tdmin.c @@ -224,15 +224,6 @@ static const struct axg_tdm_formatter_ops axg_tdmin_ops = { }; static const struct axg_tdm_formatter_driver axg_tdmin_drv = { - .component_drv = &axg_tdmin_component_drv, - .regmap_cfg = &axg_tdmin_regmap_cfg, - .ops = &axg_tdmin_ops, - .quirks = &(const struct axg_tdm_formatter_hw) { - .skew_offset = 2, - }, -}; - -static const struct axg_tdm_formatter_driver g12a_tdmin_drv = { .component_drv = &axg_tdmin_component_drv, .regmap_cfg = &axg_tdmin_regmap_cfg, .ops = &axg_tdmin_ops, @@ -247,10 +238,10 @@ static const struct of_device_id axg_tdmin_of_match[] = { .data = &axg_tdmin_drv, }, { .compatible = "amlogic,g12a-tdmin", - .data = &g12a_tdmin_drv, + .data = &axg_tdmin_drv, }, { .compatible = "amlogic,sm1-tdmin", - .data = &g12a_tdmin_drv, + .data = &axg_tdmin_drv, }, {} }; MODULE_DEVICE_TABLE(of, axg_tdmin_of_match); -- 2.29.2

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