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[PATCH] ALSA: bebob: correct duplicated entries with TerraTec OUI
by Takashi Sakamoto 05 Jul '21

05 Jul '21

ALSA bebob driver has duplicated entries for modalias of 'ieee1394:ven00000AACmo00000002sp0000A02Dver00010001' since entries for two devices below have the same parameters: * Acoustic Reality eAR Master One, Eroica, Figaro, and Ciaccona * TerraTec Aureon 7.1 FireWire I relied on FFADO revision 737 to add the former entry, on the other hand, the latter is based on message posted by actual user with information of sysfs node: * https://sourceforge.net/p/ffado/mailman/ffado-user/thread/5743F969.2080204%… It appears that they have OUI of Terratec Electronic GmbH (0x000aac) and the same model ID, thus suffice to say that they have something common in their internals. Although it's not going to make a big difference, this commit arranges the entries. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 42980da45fbf..06a7ced218e2 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -403,8 +403,6 @@ static const struct ieee1394_device_id bebob_id_table[] = { SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x01eeee, &spec_normal), /* ESI, Quatafire610 */ SND_BEBOB_DEV_ENTRY(VEN_ESI, 0x00010064, &spec_normal), - // AcousticReality, eARMasterOne. Terratec OEM. - SND_BEBOB_DEV_ENTRY(VEN_TERRATEC, 0x00000002, &spec_normal), /* CME, MatrixKFW */ SND_BEBOB_DEV_ENTRY(VEN_CME, 0x00030000, &spec_normal), // Phonic Helix Board 12 FireWire MkII. @@ -434,7 +432,8 @@ static const struct ieee1394_device_id bebob_id_table[] = { SND_BEBOB_DEV_ENTRY(VEN_TERRATEC, 0x00000007, &yamaha_terratec_spec), /* TerraTec Electronic GmbH, EWS MIC2/MIC8 */ SND_BEBOB_DEV_ENTRY(VEN_TERRATEC, 0x00000005, &spec_normal), - /* Terratec Electronic GmbH, Aureon 7.1 Firewire */ + // Terratec Electronic GmbH, Aureon 7.1 Firewire. + // AcousticReality, eAR Master One, Eroica, Figaro, and Ciaccona. Perhaps Terratec OEM. SND_BEBOB_DEV_ENTRY(VEN_TERRATEC, 0x00000002, &spec_normal), /* Yamaha, GO44 */ SND_BEBOB_DEV_ENTRY(VEN_YAMAHA, 0x0010000b, &yamaha_terratec_spec), -- 2.30.2

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[PATCH] sound: x86: fix spelling mistakes
by gushengxian 05 Jul '21

05 Jul '21

From: gushengxian <gushengxian(a)yulong.com> Fix some spelling mistakes as follows: regiter ==> register confgiuration ==> configuration playabck ==> playback platoform ==> platform Signed-off-by: gushengxian <gushengxian(a)yulong.com> --- sound/x86/intel_hdmi_audio.c | 6 +++--- sound/x86/intel_hdmi_audio.h | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 33b12aa67cf5..a34d7d9c2a57 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -236,7 +236,7 @@ static void had_write_register(struct snd_intelhad *ctx, u32 reg, u32 val) * updating AUD_CONFIG register. * This is because: * Bit6 of AUD_CONFIG register is writeonly due to a silicon bug on VLV2 - * HDMI IP. As a result a read-modify of AUD_CONFIG regiter will always + * HDMI IP. As a result a read-modify of AUD_CONFIG register will always * clear bit6. AUD_CONFIG[6:4] represents the "channels" field of the * register. This field should be 1xy binary for configuration with 6 or * more channels. Read-modify of AUD_CONFIG (Eg. for enabling audio) @@ -342,7 +342,7 @@ static int had_prog_status_reg(struct snd_pcm_substream *substream, /* * function to initialize audio - * registers and buffer confgiuration registers + * registers and buffer configuration registers * This function is called in the prepare callback */ static int had_init_audio_ctrl(struct snd_pcm_substream *substream, @@ -1790,7 +1790,7 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) pcm->private_data = ctx; pcm->info_flags = 0; strscpy(pcm->name, card->shortname, strlen(card->shortname)); - /* setup the ops for playabck */ + /* setup the ops for playback */ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &had_pcm_ops); /* allocate dma pages; diff --git a/sound/x86/intel_hdmi_audio.h b/sound/x86/intel_hdmi_audio.h index 0d91bb5dbab7..bb3853195922 100644 --- a/sound/x86/intel_hdmi_audio.h +++ b/sound/x86/intel_hdmi_audio.h @@ -96,7 +96,7 @@ struct pcm_stream_info { * @had_spinlock: driver lock * @aes_bits: IEC958 status bits * @buff_done: id of current buffer done intr - * @dev: platoform device handle + * @dev: platform device handle * @chmap: holds channel map info */ struct snd_intelhad { -- 2.25.1

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Re: [PATCH] ALSA: hda/realtek: fix mute led of the HP Pavilion 15-eh1xxx series
by Takashi Iwai 05 Jul '21

05 Jul '21

On Sat, 03 Jul 2021 15:54:16 +0200, Frank Schäfer wrote: > > The HP Pavilion 15-eh1xxx series uses the HP mainboard 88D0 with ALC287 and needs > the ALC287_FIXUP_HP_GPIO_LED quirk to make the mute led working. > Tested with a HP Pavilion 15-eh1557ng. > > Signed-off-by: Frank Schäfer <fschaefer.oss(a)googlemail.com> > Cc: <stable(a)vger.kernel.org> Thanks, applied. Takashi

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[PATCH 0/5] ASoC: ti: davinci-mcasp: Fix the DIT mode and OMAP4 support
by Peter Ujfalusi 05 Jul '21

05 Jul '21

Hi, it has been on my todo list for several years to support McASP on OMAP4 devices. For Galaxy Nexus we had an omap-mcasp driver (which was mostly a stripped down davinci-mcasp driver) to support what was needed on that specific phone + it's dock for S/PDIF (48KHz, 16bit, stereo). Not many (if any) device available to test the DIT mode of McASP. I have used BeagleBone White (McASP1 AXR3 can be routed to a pin) to get the S/PDIF mode working then PandaES for OMAP4 support (on PandaES the gpio_121 is not used and the signal is routed to expansion J6 pin14) In theory the McASP in OMAP5 should be working after this series, but the OMAP5 TRM is not public and I do not have one to check the addresses and see if there is a way to test it on omap5-uevm. Mark, Tony: The ASoC and dts patches can go via separate tree I felt that it is better if they are together, at least initially. Nikolaus: fyi, this might be useful for Pyra? Regards, Péter --- Peter Ujfalusi (5): ASoC: ti: davinci-mcasp: Fix DIT mode support ASoC: dt-bindings: davinci-mcasp: Add compatible string for OMAP4 ASoC: ti: davinci-mcasp: Add support for the OMAP4 version of McASP ARM: dts: omap4-l4-abe: Correct sidle modes for McASP ARM: dts: omap4-l4-abe: Add McASP configuration .../bindings/sound/davinci-mcasp-audio.txt | 1 + arch/arm/boot/dts/omap4-l4-abe.dtsi | 39 ++-- include/linux/platform_data/davinci_asp.h | 1 + sound/soc/ti/Kconfig | 1 + sound/soc/ti/davinci-mcasp.c | 168 +++++++++++++++--- 5 files changed, 168 insertions(+), 42 deletions(-) -- 2.32.0

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[PATCH v2 0/8] ASoC: mediatek: Add support for MT8195 SoC
by Trevor Wu 05 Jul '21

05 Jul '21

This series of patches adds support for Mediatek AFE of MT8195 SoC. Patches are based on broonie tree "for-next" branch. Changes since v1: - fixed some problems related to dt-bindings - add some missing properties to dt-bindings - add depency declaration on dt-bindings - fixed some warnings found by kernel test robot Trevor Wu (8): ASoC: mediatek: mt8195: update mediatek common driver ASoC: mediatek: mt8195: support etdm in platform driver ASoC: mediatek: mt8195: support adda in platform driver ASoC: mediatek: mt8195: support pcm in platform driver ASoC: mediatek: mt8195: add platform driver dt-bindings: mediatek: mt8195: add audio afe document ASoC: mediatek: mt8195: add machine driver with mt6359, rt1019 and rt5682 dt-bindings: mediatek: mt8195: add mt8195-mt6359-rt1019-rt5682 document .../bindings/sound/mt8195-afe-pcm.yaml | 136 + .../sound/mt8195-mt6359-rt1019-rt5682.yaml | 39 + sound/soc/mediatek/Kconfig | 23 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 22 +- sound/soc/mediatek/common/mtk-base-afe.h | 10 +- sound/soc/mediatek/mt8195/Makefile | 14 + sound/soc/mediatek/mt8195/mt8195-afe-clk.c | 899 +++++ sound/soc/mediatek/mt8195/mt8195-afe-clk.h | 201 + sound/soc/mediatek/mt8195/mt8195-afe-common.h | 200 + sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 3264 +++++++++++++++++ sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 929 +++++ sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 2666 ++++++++++++++ sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 393 ++ .../mt8195/mt8195-mt6359-rt1019-rt5682.c | 977 +++++ sound/soc/mediatek/mt8195/mt8195-reg.h | 2793 ++++++++++++++ 16 files changed, 12562 insertions(+), 5 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8195-mt6359-rt1019-rt5682.yaml create mode 100644 sound/soc/mediatek/mt8195/Makefile create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-clk.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-clk.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-common.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-dai-adda.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-dai-etdm.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-reg.h -- 2.18.0

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[PATCH v5 2/2] ASoC: sun50i-dmic: dt-bindings: add DT bindings for DMIC controller
by fengzheng923＠gmail.com 04 Jul '21

04 Jul '21

From: Ban Tao <fengzheng923(a)gmail.com> DT binding documentation for this new ASoC driver. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- v1->v2: 1.Fix some build errors. --- v2->v3: 1.Fix some build errors. --- v3->v4: 1.None. --- v4->v5: 1.Add interrupt. 2.Keep clock and reset index. --- .../sound/allwinner,sun50i-h6-dmic.yaml | 79 +++++++++++++++++++ 1 file changed, 79 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml new file mode 100644 index 000000000000..0cfc07f369bd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun50i-h6-dmic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner H6 DMIC Device Tree Bindings + +maintainers: + - Ban Tao <fengzheng923(a)gmail.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: allwinner,sun50i-h6-dmic + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: bus + - const: mod + + dmas: + items: + - description: RX DMA Channel + + dma-names: + items: + - const: rx + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - resets + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + #include <dt-bindings/clock/sun50i-h6-ccu.h> + #include <dt-bindings/reset/sun50i-h6-ccu.h> + + dmic: dmic@5095000 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun50i-h6-dmic"; + reg = <0x05095000 0x400>; + interrupts = <GIC_SPI 22 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&ccu CLK_BUS_DMIC>, <&ccu CLK_DMIC>; + clock-names = "bus", "mod"; + dmas = <&dma 7>; + dma-names = "rx"; + resets = <&ccu RST_BUS_DMIC>; + }; + +... -- 2.17.1

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Re: [GIT PULL] sound updates for 5.14-rc1
by Linus Torvalds 03 Jul '21

03 Jul '21

On Fri, Jul 2, 2021 at 9:37 PM Linus Torvalds <torvalds(a)linux-foundation.org> wrote: > > But I thought I'd report this as a likely candidate. Confirmed. The watchdog hang bisects right down to commit 9ce650a75a3b ("ALSA: usb-audio: Reduce latency at playback start"). And reverting it on top of my tree also fixes the hang, so it's not some bisection fluke. I have no idea what is actually wrong with that commit, but it most definitely is the problem, and I have reverted it in my tree so that I can continue merging stuff tomorrow. Linus

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[PATCH] ALSA: hda: Enable runtime PM when codec probe fails
by Kai-Heng Feng 02 Jul '21

02 Jul '21

When codec probe fails, it doesn't enable runtime suspend, and can prevent graphics card from getting powered down: [ 4.280991] snd_hda_intel 0000:01:00.1: no codecs initialized $ cat /sys/bus/pci/devices/0000:01:00.1/power/runtime_status active So enable runtime PM when codec probe fails, to let graphics card be able to runtime suspend again. Merge azx_probe_continue() into azx_probe() and just let probe fail for this case could be a better approach. However that's a much bigger task so let's settle with a quirk workaround. BugLink: https://bugs.launchpad.net/bugs/1907212 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/hda_intel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6852668f1bcb..3fd920069268 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2328,7 +2328,7 @@ static int azx_probe_continue(struct azx *chip) if (bus->codec_mask) { err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); if (err < 0) - goto out_free; + goto out_enable_rpm; } #ifdef CONFIG_SND_HDA_PATCH_LOADER @@ -2360,6 +2360,7 @@ static int azx_probe_continue(struct azx *chip) set_default_power_save(chip); +out_enable_rpm: if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); pm_runtime_allow(&pci->dev); -- 2.29.2

8 32

timestamp
by folkert 02 Jul '21

02 Jul '21

Hi, Please consider including the following patch. This patch adds a '-T' flag which prepends MIDI messages outputted by '--dump' with a timestamp. E.g.: 1625245188.993861810) C964 1625245188.994318675) F8 1625245188.995468101) 99647F 1625245189.494490183) F8 1625245189.495448424) 99647F 1625245189.995470260) F8 1625245189.996461381) 99647F 1625245190.494519924) F8 [...] diff -uNr t3/alsa-utils-1.2.3/amidi/amidi.1 alsa-utils-1.2.3/amidi/amidi.1 --- t3/alsa-utils-1.2.3/amidi/amidi.1 2020-06-07 19:19:42.000000000 +0200 +++ alsa-utils-1.2.3/amidi/amidi.1 2021-07-02 14:35:25.416330576 +0200 @@ -6,6 +6,7 @@ .SH SYNOPSIS \fBamidi\fP [\fI\-p port\fP] [\fI\-s file\fP | \fI\-S data\fP] [\fI\-r file\fP] [\fI\-d\fP] [\fI\-t seconds\fP] [\fI\-a\fP] +[\fI\-T\fP] .SH DESCRIPTION .B amidi @@ -41,6 +42,10 @@ options to specify what data to send or receive. .TP +.I \-T +Adds a timestamp in front of each MIDI message. + +.TP .I \-h, \-\-help Help: prints a list of options. diff -uNr t3/alsa-utils-1.2.3/amidi/amidi.c alsa-utils-1.2.3/amidi/amidi.c --- t3/alsa-utils-1.2.3/amidi/amidi.c 2020-06-07 19:19:42.000000000 +0200 +++ alsa-utils-1.2.3/amidi/amidi.c 2021-07-02 19:41:34.476470438 +0200 @@ -36,6 +36,7 @@ #include <unistd.h> #include <fcntl.h> #include <alsa/asoundlib.h> +#include <time.h> #include "aconfig.h" #include "version.h" @@ -80,6 +81,7 @@ "-r, --receive=file write received data into a file\n" "-S, --send-hex=\"...\" send hexadecimal bytes\n" "-d, --dump print received data as hexadecimal bytes\n" + "-T, --timestamp adds a timestamp in front of each dumped message\n" "-t, --timeout=seconds exits when no data has been received\n" " for the specified duration\n" "-a, --active-sensing include active sensing bytes\n" @@ -356,7 +358,7 @@ /* * prints MIDI commands, formatting them nicely */ -static void print_byte(unsigned char byte) +static void print_byte(unsigned char byte, struct timespec *ts) { static enum { STATE_UNKNOWN, @@ -426,7 +428,18 @@ if (running_status) fputs("\n ", stdout); } - printf("%c%02X", newline ? '\n' : ' ', byte); + + if (newline) { + printf("\n"); + + /* Nanoseconds does not make a lot of sense for serial MIDI (the + * 31250 bps one) but I'm not sure about MIDI over USB. + */ + if (ts) + printf("%lld.%.9ld) ", (long long)ts->tv_sec, ts->tv_nsec); + } + + printf("%02X", byte); } static void sig_handler(int dummy) @@ -454,7 +467,7 @@ int main(int argc, char *argv[]) { - static const char short_options[] = "hVlLp:s:r:S::dt:aci:"; + static const char short_options[] = "hVlLp:s:r:S::dt:aci:T"; static const struct option long_options[] = { {"help", 0, NULL, 'h'}, {"version", 0, NULL, 'V'}, @@ -465,6 +478,7 @@ {"receive", 1, NULL, 'r'}, {"send-hex", 2, NULL, 'S'}, {"dump", 0, NULL, 'd'}, + {"timestamp", 0, NULL, 'T'}, {"timeout", 1, NULL, 't'}, {"active-sensing", 0, NULL, 'a'}, {"clock", 0, NULL, 'c'}, @@ -475,6 +489,7 @@ int ignore_active_sensing = 1; int ignore_clock = 1; int do_send_hex = 0; + int do_print_timestamp = 0; struct itimerspec itimerspec = { .it_interval = { 0, 0 } }; while ((c = getopt_long(argc, argv, short_options, @@ -509,6 +524,9 @@ case 'd': dump = 1; break; + case 'T': + do_print_timestamp = 1; + break; case 't': if (optarg) timeout = atof(optarg); @@ -611,6 +629,7 @@ } if (inputp) { + int need_timestamp = 0; int read = 0; int npfds; struct pollfd *pfds; @@ -648,6 +667,7 @@ unsigned char buf[256]; int i, length; unsigned short revents; + struct timespec ts; err = poll(pfds, npfds, -1); if (stop || (err < 0 && errno == EINTR)) @@ -657,6 +677,11 @@ break; } + if (clock_gettime(CLOCK_REALTIME, &ts) < 0) { + error("clock_getres failed: %s", strerror(errno)); + break; + } + err = snd_rawmidi_poll_descriptors_revents(input, &pfds[1], npfds - 1, &revents); if (err < 0) { error("cannot get poll events: %s", snd_strerror(errno)); @@ -692,7 +717,8 @@ write(receive_file, buf, length); if (dump) { for (i = 0; i < length; ++i) - print_byte(buf[i]); + print_byte(buf[i], do_print_timestamp ? &ts : NULL); + fflush(stdout); } diff -uNr t3/alsa-utils-1.2.3/configure.ac alsa-utils-1.2.3/configure.ac --- t3/alsa-utils-1.2.3/configure.ac 2021-07-02 19:41:46.000000000 +0200 +++ alsa-utils-1.2.3/configure.ac 2021-07-02 15:55:13.245585248 +0200 @@ -97,7 +97,7 @@ if test x$bat = xtrue; then saved_CFLAGS="$CFLAGS" - saved_LDFLAGS="$LDFLAGS" + saved_LDFLAGS="$LDFLAGS -lrt" saved_LIBS="$LIBS" FFTW_INC="" FFTW_LIB="" @@ -116,7 +116,7 @@ FFTW_CFLAGS="$CFLAGS" FFTW_LIB="$LIBS" CFLAGS="$saved_CFLAGS" - LDFLAGS="$saved_LDFLAGS" + LDFLAGS="$saved_LDFLAGS -lrt" LIBS="$saved_LIBS" AC_SUBST(FFTW_INC) AC_SUBST(FFTW_LIB) @@ -292,11 +292,11 @@ fi saved_CFLAGS="$CFLAGS" - saved_LDFLAGS="$LDFLAGS" + saved_LDFLAGS="$LDFLAGS -lrt" saved_LIBS="$LIBS" CFLAGS="$CFLAGS $CURSES_CFLAGS" if test -n "$CURSESLIBDIR"; then - LDFLAGS="$LDFLAGS -L$CURSESLIBDIR" + LDFLAGS="$LDFLAGS -L$CURSESLIBDIR -lrt" fi LIBS="$CURSESLIB $LIBS" @@ -342,7 +342,7 @@ [AC_MSG_ERROR([form$NCURSESLIBSUFFIX library not found])]) CFLAGS="$saved_CFLAGS" - LDFLAGS="$saved_LDFLAGS" + LDFLAGS="$saved_LDFLAGS -lrt" LIBS="$saved_LIBS" if test -n "$CURSESLIBDIR"; then

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[tiwai/for-next] hard lockups with pipewire and snd-usb-audio
by Hector Martin 02 Jul '21

02 Jul '21

Hi Takashi, I've been testing builds from your for-next branch to get the recent firewire fixes. All that works fine, but instead I've been getting hard lockups with pipewire and snd-usb-audio. I'm currently on 9a7b7ec3c6. Backtrace: do_raw_spin_lock+0x8e/0x90 _raw_spin_lock_irqsave+0x3e/0x50 ? snd_pcm_period_elapsed+0x10/0x30 [snd_pcm] snd_pcm_period_elapsed+0x10/0x30 [snd_pcm] snd_usb_endpoint_start+0xd8/0x270 [snd_usb_audio] start_endpoints+0x50/0x70 [snd_usb_audio] snd_usb_substream_playback_trigger+0x58/0x100 [snd_usb_audio] snd_pcm_action+0x74/0xb0 [snd_pcm] snd_pcm_action_lock_irq+0x36/0x80 [snd_pcm] snd_pcm_ioctl+0x1e/0x30 [snd_pcm] __x64_sys_ioctl+0x7d/0xb0 do_syscall_64+0x45/0x80 entry_SYSCALL_64_after_hwframe+0x44/0xae To me it looks like it's recursively trying to lock the stream spinlock. I'm not sure what version this is a regression from; I've only triggered it with pipewire so far, which I wasn't previously using. It's easy to reproduce. Usually, pulling up pavucontrol on pipewire with the USB audio device on an active profile will do it. Just thought you might want to know. Let me know if you need any more info. -- Hector Martin (marcan(a)marcan.st) Public Key: https://mrcn.st/pub

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