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

[PATCH -next] ALSA: timer: Fix use-after-free problem
by Wang Wensheng 03 Nov '21

03 Nov '21

When the timer instance was add into ack_list but was not currently in process, the user could stop it via snd_timer_stop1() without delete it from the ack_list. Then the user could free the timer instance and when it was actually processed UAF occurred. This issue could be reproduced via testcase snd_timer01 in ltp - running several instances of that testcase at the same time. What I actually met was that the ack_list of the timer broken and the kernel went into deadloop with irqoff. That could be detected by hardlockup detector on board or when we run it on qemu, we could use gdb to dump the ack_list when the console has no response. To fix this issue, we introduce a new flag SNDRV_TIMER_IFLG_ACKING to indicate the state where the timer instance is in ack_list but not currently processed and check against the new flag in snd_timer_stop1() and delete it from ack_list if the flag is set. Signed-off-by: Wang Wensheng <wangwensheng4(a)huawei.com> --- include/sound/timer.h | 1 + sound/core/timer.c | 12 +++++++++--- 2 files changed, 10 insertions(+), 3 deletions(-) diff --git a/include/sound/timer.h b/include/sound/timer.h index 760e132cc0cd..549288e94a39 100644 --- a/include/sound/timer.h +++ b/include/sound/timer.h @@ -31,6 +31,7 @@ #define SNDRV_TIMER_IFLG_CALLBACK 0x00000020 /* timer callback is active */ #define SNDRV_TIMER_IFLG_EXCLUSIVE 0x00000040 /* exclusive owner - no more instances */ #define SNDRV_TIMER_IFLG_EARLY_EVENT 0x00000080 /* write early event to the poll queue */ +#define SNDRV_TIMER_IFLG_ACKING 0x00000100 /* the timeri was added to ack_list */ #define SNDRV_TIMER_FLG_CHANGE 0x00000001 #define SNDRV_TIMER_FLG_RESCHED 0x00000002 /* need reschedule */ diff --git a/sound/core/timer.c b/sound/core/timer.c index 92b7008fcdb8..1d1e4274919c 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -625,10 +625,12 @@ static int snd_timer_stop1(struct snd_timer_instance *timeri, bool stop) return -EINVAL; spin_lock_irqsave(&timer->lock, flags); if (!(timeri->flags & (SNDRV_TIMER_IFLG_RUNNING | - SNDRV_TIMER_IFLG_START))) { + SNDRV_TIMER_IFLG_START | + SNDRV_TIMER_IFLG_ACKING))) { result = -EBUSY; goto unlock; } + list_del_init(&timeri->ack_list); list_del_init(&timeri->active_list); if (timer->card && timer->card->shutdown) @@ -649,7 +651,8 @@ static int snd_timer_stop1(struct snd_timer_instance *timeri, bool stop) } } } - timeri->flags &= ~(SNDRV_TIMER_IFLG_RUNNING | SNDRV_TIMER_IFLG_START); + timeri->flags &= ~(SNDRV_TIMER_IFLG_RUNNING | SNDRV_TIMER_IFLG_START | + SNDRV_TIMER_IFLG_ACKING); if (stop) timeri->flags &= ~SNDRV_TIMER_IFLG_PAUSED; else @@ -786,6 +789,7 @@ static void snd_timer_process_callbacks(struct snd_timer *timer, /* remove from ack_list and make empty */ list_del_init(&ti->ack_list); + ti->flags &= ~SNDRV_TIMER_IFLG_ACKING; if (!(ti->flags & SNDRV_TIMER_IFLG_DEAD)) { ticks = ti->pticks; @@ -890,8 +894,10 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left) ack_list_head = &timer->ack_list_head; else ack_list_head = &timer->sack_list_head; - if (list_empty(&ti->ack_list)) + if (list_empty(&ti->ack_list)) { list_add_tail(&ti->ack_list, ack_list_head); + ti->flags |= SNDRV_TIMER_IFLG_ACKING; + } list_for_each_entry(ts, &ti->slave_active_head, active_list) { ts->pticks = ti->pticks; ts->resolution = resolution; -- 2.17.1

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[PATCH v5 0/2] Machine driver to support LPASS SC7280 sound card registration
by Srinivasa Rao Mandadapu 02 Nov '21

02 Nov '21

This patch set is to add support for SC7280 sound card registration and to add dt-bindings documentation file. Srinivasa Rao Mandadapu (2): ASoC: google: dt-bindings: Add sc7280-herobrine machine bindings ASoC: qcom: SC7280: Add machine driver Changes Since V4: -- Add COMPILE_TEST flag in sc7280 configuration. -- Remove redundant startup and shutdown callbacks of snd_soc_ops. -- Fix typo errors. Changes Since V3: -- Change audio jack playpause key value. Changes Since V2: -- updated required field in bindings -- updated Return type check with proper enum in sc7280.c -- Updated Header name and Typos in sc7280.c Changes Since V1: -- Indentation changes and typo. .../bindings/sound/google,sc7280-herobrine.yaml | 170 ++++++++++++ sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7280.c | 299 +++++++++++++++++++++ 4 files changed, 483 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml create mode 100644 sound/soc/qcom/sc7280.c -- 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: rsnd: Fix an error handling path in 'rsnd_node_count()'
by Christophe JAILLET 02 Nov '21

02 Nov '21

If we return before the end of the 'for_each_child_of_node()' iterator, the reference taken on 'np' must be released. Add the missing 'of_node_put()' call. Fixes: c413983eb66a ("ASoC: rsnd: adjust disabled module") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/sh/rcar/core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 978bd0406729..6a8fe0da7670 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1225,6 +1225,7 @@ int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name if (i < 0) { dev_err(dev, "strange node numbering (%s)", of_node_full_name(node)); + of_node_put(np); return 0; } i++; -- 2.30.2

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[PATCH] sound: soc: tlv320aic3x: Make aic3x_remove() return void
by Uwe Kleine-König 02 Nov '21

02 Nov '21

Up to now aic3x_remove() returns zero unconditionally. Make it return void instead which makes it easier to see in the callers that there is no error to handle. Also the return value of i2c and spi remove callbacks is ignored anyway. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/tlv320aic3x-i2c.c | 4 +++- sound/soc/codecs/tlv320aic3x-spi.c | 4 +++- sound/soc/codecs/tlv320aic3x.c | 3 +-- sound/soc/codecs/tlv320aic3x.h | 2 +- 4 files changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x-i2c.c b/sound/soc/codecs/tlv320aic3x-i2c.c index cd0558ed4dd4..2f272bc3f5da 100644 --- a/sound/soc/codecs/tlv320aic3x-i2c.c +++ b/sound/soc/codecs/tlv320aic3x-i2c.c @@ -32,7 +32,9 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *i static int aic3x_i2c_remove(struct i2c_client *i2c) { - return aic3x_remove(&i2c->dev); + aic3x_remove(&i2c->dev); + + return 0; } static const struct i2c_device_id aic3x_i2c_id[] = { diff --git a/sound/soc/codecs/tlv320aic3x-spi.c b/sound/soc/codecs/tlv320aic3x-spi.c index 8c7b6bb9223f..494e84402232 100644 --- a/sound/soc/codecs/tlv320aic3x-spi.c +++ b/sound/soc/codecs/tlv320aic3x-spi.c @@ -37,7 +37,9 @@ static int aic3x_spi_probe(struct spi_device *spi) static int aic3x_spi_remove(struct spi_device *spi) { - return aic3x_remove(&spi->dev); + aic3x_remove(&spi->dev); + + return 0; } static const struct spi_device_id aic3x_spi_id[] = { diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 7731593a5509..d53037b1509d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1870,7 +1870,7 @@ int aic3x_probe(struct device *dev, struct regmap *regmap, kernel_ulong_t driver } EXPORT_SYMBOL(aic3x_probe); -int aic3x_remove(struct device *dev) +void aic3x_remove(struct device *dev) { struct aic3x_priv *aic3x = dev_get_drvdata(dev); @@ -1881,7 +1881,6 @@ int aic3x_remove(struct device *dev) gpio_set_value(aic3x->gpio_reset, 0); gpio_free(aic3x->gpio_reset); } - return 0; } EXPORT_SYMBOL(aic3x_remove); diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 7e0063913017..14298f9e6d9b 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -14,7 +14,7 @@ struct regmap_config; extern const struct regmap_config aic3x_regmap; int aic3x_probe(struct device *dev, struct regmap *regmap, kernel_ulong_t driver_data); -int aic3x_remove(struct device *dev); +void aic3x_remove(struct device *dev); #define AIC3X_MODEL_3X 0 #define AIC3X_MODEL_33 1 -- 2.30.2

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[PATCH v4 0/2] Add code to manage DSP clocks and provide dts-binding document
by YC Hung 02 Nov '21

02 Nov '21

From: "yc.hung" <yc.hung(a)mediatek.com> This code is based on top of SOF topic/sof-dev branch and we want to have a review with ALSA and device Tree communities the it will be merged to SOF tree and then merged into ALSA tree. It provides two patches, one is for mt8195 dsp clocks related. Another is for mt8195 dsp dts binding decription. YC Hung (2): ASoC: SOF: mediatek: Add mt8195 dsp clock support dt-bindings: dsp: mediatek: Add mt8195 DSP binding support .../bindings/dsp/mtk,mt8195-dsp.yaml | 139 +++++++++++++++ sound/soc/sof/mediatek/adsp_helper.h | 2 +- sound/soc/sof/mediatek/mt8195/Makefile | 2 +- sound/soc/sof/mediatek/mt8195/mt8195-clk.c | 158 ++++++++++++++++++ sound/soc/sof/mediatek/mt8195/mt8195-clk.h | 28 ++++ sound/soc/sof/mediatek/mt8195/mt8195.c | 22 ++- 6 files changed, 347 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/dsp/mtk,mt8195-dsp.yaml create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195-clk.c create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195-clk.h -- 2.18.0

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[PATCH] ALSA: usb-audio: Add quirk for Audient iD14
by Takashi Iwai 02 Nov '21

02 Nov '21

Audient iD14 (2708:0002) may get a control message error that interferes the operation e.g. with alsactl. Add the quirk to ignore such errors like other devices. BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1191247 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 1d418c3fc598..64e1c20311ed 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1888,6 +1888,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x21b4, 0x0081, /* AudioQuest DragonFly */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x2708, 0x0002, /* Audient iD14 */ + QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x2912, 0x30c8, /* Audioengine D1 */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x30be, 0x0101, /* Schiit Hel */ -- 2.31.1

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[PATCH] Make top/rear speaker, mute and micmute leds work on HP x360 14-ea000 laptops that use Realtek 245 codec
by Jonathan Clarke 02 Nov '21

02 Nov '21

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=210633 Signed-off-by: Jonathan Clarke <jonathan.a.clarke(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 46 +++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 22d27b12c..e3c6d17ea 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4409,6 +4409,13 @@ static void alc245_fixup_hp_x360_amp(struct hda_codec *codec, case HDA_FIXUP_ACT_PRE_PROBE: spec->gpio_mask |= 0x01; spec->gpio_dir |= 0x01; + + /* use only amp at 0x02 for bottom(front) speaker, + * otherwise it is set to use 0x02,0x03,0x06 and when used in conjunction + * with top(rear) speaker 0x14, gets locked at full volume */ + static const hda_nid_t conn1[] = { 0x02 }; + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn1), conn1); + break; case HDA_FIXUP_ACT_INIT: /* need to toggle GPIO to enable the amp */ @@ -4503,6 +4510,26 @@ static void alc236_fixup_hp_mute_led_coefbit(struct hda_codec *codec, } } +static void alc245_fixup_hp_x360_mute_leds(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + /* mic mute is set via gpio 0x04 */ + spec->micmute_led_polarity = 1; + codec->power_filter = led_power_filter; + alc_fixup_hp_gpio_led(codec, action, 0x00, 0x04); + + /* output mute is set via SET_COEF_INDEX,SET_PROC_COEF */ + spec->mute_led_polarity = 0; + spec->mute_led_coef.idx = 0x0b; + spec->mute_led_coef.mask = 0xffff; + spec->mute_led_coef.on = 0xa02f; + spec->mute_led_coef.off = 0x7774; + snd_hda_gen_add_mute_led_cdev(codec, coef_mute_led_set); + } +} + /* turn on/off mic-mute LED per capture hook by coef bit */ static int coef_micmute_led_set(struct led_classdev *led_cdev, enum led_brightness brightness) @@ -6557,6 +6584,8 @@ enum { ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, ALC280_FIXUP_HP_9480M, ALC245_FIXUP_HP_X360_AMP, + ALC245_FIXUP_HP_X360_MUTE_LEDS, + ALC245_FIXUP_HP_X360_GPIO_TOP_SPEAKER, ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, ALC288_FIXUP_DELL_XPS_13, @@ -7293,6 +7322,21 @@ static const struct hda_fixup alc269_fixups[] = { [ALC245_FIXUP_HP_X360_AMP] = { .type = HDA_FIXUP_FUNC, .v.func = alc245_fixup_hp_x360_amp, + .chained = true, + .chain_id = ALC245_FIXUP_HP_X360_MUTE_LEDS + }, + [ALC245_FIXUP_HP_X360_MUTE_LEDS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc245_fixup_hp_x360_mute_leds, + .chained = true, + .chain_id = ALC245_FIXUP_HP_X360_GPIO_TOP_SPEAKER + }, + [ALC245_FIXUP_HP_X360_GPIO_TOP_SPEAKER] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x90170110 }, /* enable top(back) speaker in addition to bottom(front) speaker at 0x17 */ + { }, + } }, [ALC288_FIXUP_DELL_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, @@ -9003,6 +9047,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"}, {.id = ALC274_FIXUP_HP_MIC, .name = "alc274-hp-mic-detect"}, {.id = ALC245_FIXUP_HP_X360_AMP, .name = "alc245-hp-x360-amp"}, + {.id = ALC245_FIXUP_HP_X360_MUTE_LEDS, .name = "alc245-hp-x360-mute-leds"}, + {.id = ALC245_FIXUP_HP_X360_GPIO_TOP_SPEAKER, .name = "alc245-hp-x360-gpio-top-speaker"}, {.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"}, {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, {.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"}, -- 2.25.1

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[PATCH v2] ALSA: seq: Fix RCU stall in snd_seq_write()
by Zqiang 02 Nov '21

02 Nov '21

If we have a lot of cell object, this cycle may take a long time, and trigger RCU stall. insert a conditional reschedule point to fix it. rcu: INFO: rcu_preempt self-detected stall on CPU rcu: 1-....: (1 GPs behind) idle=9f5/1/0x4000000000000000 softirq=16474/16475 fqs=4916 (t=10500 jiffies g=19249 q=192515) NMI backtrace for cpu 1 ...... asm_sysvec_apic_timer_interrupt RIP: 0010:_raw_spin_unlock_irqrestore+0x38/0x70 spin_unlock_irqrestore snd_seq_prioq_cell_out+0x1dc/0x360 snd_seq_check_queue+0x1a6/0x3f0 snd_seq_enqueue_event+0x1ed/0x3e0 snd_seq_client_enqueue_event.constprop.0+0x19a/0x3c0 snd_seq_write+0x2db/0x510 vfs_write+0x1c4/0x900 ksys_write+0x171/0x1d0 do_syscall_64+0x35/0xb0 Reported-by: syzbot+bb950e68b400ab4f65f8(a)syzkaller.appspotmail.com Signed-off-by: Zqiang <qiang.zhang1211(a)gmail.com> --- v1->v2: Add atomic context detection. sound/core/seq/seq_queue.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index d6c02dea976c..3a05b5e0d555 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -263,6 +263,8 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) if (!cell) break; snd_seq_dispatch_event(cell, atomic, hop); + if (!atomic) + cond_resched(); } /* Process time queue... */ @@ -272,6 +274,8 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) if (!cell) break; snd_seq_dispatch_event(cell, atomic, hop); + if (!atomic) + cond_resched(); } /* free lock */ -- 2.17.1

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Config file for a52 plugin (60-a52-encoder.conf) produces no/garbled sound via optical digital out (S/PDIF)
by GitHub issues - edited 02 Nov '21

02 Nov '21

alsa-project/alsa-plugins issue #38 was edited from jensgw: Dear Maintainer, this bug was originally reported on the debian tracker. They advised me to post it here. https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=998125 * What led up to the situation? Fresh Debian 11 install. I want to output 5.1 sound via toslink. Used KDE settings to switch from analog output to digital out 5.1. This produced no or garbled sound and Youtube videos would play back way to fast without sound. Then I switched to digital out stereo, this worked as expected but only stereo. I could trace back the error to the shipped config file for alsa's a52 plugin: 60-a52-encoder.conf. When I replaced that file with a much simpler one (see below) 5.1 digital out worked as expected. I tried with onboard sound (PCH9) and dedicated card (Audigy Z) with the same result, the inlcuded a52 config file does not work. == Start working a52.conf ``` pcm.a52 { @args [CARD] @args.CARD { type string } type rate slave { pcm { type a52 bitrate 448 channels 6 card $CARD } rate 48000 } } ``` == End working a52.conf Therefore I assume the shipped config file with debian has an error somewhere. Please fix this. Best reagards Jens -- System Information: Debian Release: 11.1 Architecture: amd64 (x86_64) Kernel: Linux 5.14.0-0.bpo.2-amd64 (SMP w/4 CPU threads) Kernel taint flags: TAINT_UNSIGNED_MODULE Locale: LANG=de_DE.UTF-8, LC_CTYPE=de_DE.UTF-8 (charmap=UTF-8), LANGUAGE=en_US:en Shell: /bin/sh linked to /usr/bin/dash Init: systemd (via /run/systemd/system) Versions of packages libasound2-plugins depends on: ii libasound2 1.2.4-1.1 ii libavcodec-extra58 [libavcodec58] 7:4.3.2-0+deb11u2 ii libavresample4 7:4.3.2-0+deb11u2 ii libavutil56 7:4.3.2-0+deb11u2 ii libc6 2.31-13+deb11u2 ii libjack0 [libjack-0.125] 1:0.125.0-3+b1 ii libpulse0 14.2-2 ii libsamplerate0 0.2.1+ds0-1 Issue URL : https://github.com/alsa-project/alsa-plugins/issues/38 Repository URL: https://github.com/alsa-project/alsa-plugins

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Config file for a52 plugin (60-a52-encoder.conf) produces no/garbled sound via optical digital out (S/PDIF)
by GitHub issues - edited 02 Nov '21

02 Nov '21

alsa-project/alsa-plugins issue #38 was edited from jensgw: Dear Maintainer, this bug was originally reported on the debian tracker. They advised me to post it here. https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=998125 * What led up to the situation? Fresh Debian 11 install. I want to output 5.1 sound via toslink. Used KDE settings to switch from analog output to digital out 5.1. This produced no or garbled sound and Youtube videos would play back way to fast without sound. Then I switched to digital out stereo, this worked as expected but only stereo. I could trace back the error to the shipped config file for alsa's a52 plugin: 60-a52-encoder.conf. When I replaced that file with a much simpler one (see below) 5.1 digital out worked as expected. I tried with onboard sound (PCH9) and dedicated card (Audigy Z) with the same result, the inlcuded a52 config file does not work. == Start a52.conf ``` pcm.a52 { @args [CARD] @args.CARD { type string } type rate slave { pcm { type a52 bitrate 448 channels 6 card $CARD } rate 48000 } } ``` == End a52.conf Therefore I assume the shipped config file with debian has an error somewhere. Please fix this. Best reagards Jens -- System Information: Debian Release: 11.1 Architecture: amd64 (x86_64) Kernel: Linux 5.14.0-0.bpo.2-amd64 (SMP w/4 CPU threads) Kernel taint flags: TAINT_UNSIGNED_MODULE Locale: LANG=de_DE.UTF-8, LC_CTYPE=de_DE.UTF-8 (charmap=UTF-8), LANGUAGE=en_US:en Shell: /bin/sh linked to /usr/bin/dash Init: systemd (via /run/systemd/system) Versions of packages libasound2-plugins depends on: ii libasound2 1.2.4-1.1 ii libavcodec-extra58 [libavcodec58] 7:4.3.2-0+deb11u2 ii libavresample4 7:4.3.2-0+deb11u2 ii libavutil56 7:4.3.2-0+deb11u2 ii libc6 2.31-13+deb11u2 ii libjack0 [libjack-0.125] 1:0.125.0-3+b1 ii libpulse0 14.2-2 ii libsamplerate0 0.2.1+ds0-1 Issue URL : https://github.com/alsa-project/alsa-plugins/issues/38 Repository URL: https://github.com/alsa-project/alsa-plugins

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