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

[PATCH AUTOSEL 6.2 35/60] ASoC: kirkwood: Iterate over array indexes instead of using pointer math
by Sasha Levin 27 Feb '23

27 Feb '23

From: Kees Cook <keescook(a)chromium.org> [ Upstream commit b3bcedc0402fcdc5c8624c433562d9d1882749d8 ] Walking the dram->cs array was seen as accesses beyond the first array item by the compiler. Instead, use the array index directly. This allows for run-time bounds checking under CONFIG_UBSAN_BOUNDS as well. Seen with GCC 13 with -fstrict-flex-arrays: ../sound/soc/kirkwood/kirkwood-dma.c: In function 'kirkwood_dma_conf_mbus_windows.constprop': ../sound/soc/kirkwood/kirkwood-dma.c:90:24: warning: array subscript 0 is outside array bounds of 'const struct mbus_dram_window[0]' [-Warray-bounds=] 90 | if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { | ~~^~~~~~ Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> Link: https://lore.kernel.org/r/20230127224128.never.410-kees@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 700a18561a940..640cebd2983e2 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -86,7 +86,7 @@ kirkwood_dma_conf_mbus_windows(void __iomem *base, int win, /* try to find matching cs for current dma address */ for (i = 0; i < dram->num_cs; i++) { - const struct mbus_dram_window *cs = dram->cs + i; + const struct mbus_dram_window *cs = &dram->cs[i]; if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { writel(cs->base & 0xffff0000, base + KIRKWOOD_AUDIO_WIN_BASE_REG(win)); -- 2.39.0

1 0

[PATCH AUTOSEL 6.2 34/60] ASoC: soc-compress: Reposition and add pcm_mutex
by Sasha Levin 27 Feb '23

27 Feb '23

From: 강신형 <s47.kang(a)samsung.com> [ Upstream commit aa9ff6a4955fdba02b54fbc4386db876603703b7 ] If panic_on_warn is set and compress stream(DPCM) is started, then kernel panic occurred because card->pcm_mutex isn't held appropriately. In the following functions, warning were issued at this line "snd_soc_dpcm_mutex_assert_held". static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { ... snd_soc_dpcm_mutex_assert_held(fe); ... } void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { ... snd_soc_dpcm_mutex_assert_held(fe); ... } void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, int stream, int action) { ... snd_soc_dpcm_mutex_assert_held(rtd); ... } int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event) { ... snd_soc_dpcm_mutex_assert_held(fe); ... } These functions are called by soc_compr_set_params_fe, soc_compr_open_fe and soc_compr_free_fe without pcm_mutex locking. And this is call stack. [ 414.527841][ T2179] pc : dpcm_process_paths+0x5a4/0x750 [ 414.527848][ T2179] lr : dpcm_process_paths+0x37c/0x750 [ 414.527945][ T2179] Call trace: [ 414.527949][ T2179] dpcm_process_paths+0x5a4/0x750 [ 414.527955][ T2179] soc_compr_open_fe+0xb0/0x2cc [ 414.527972][ T2179] snd_compr_open+0x180/0x248 [ 414.527981][ T2179] snd_open+0x15c/0x194 [ 414.528003][ T2179] chrdev_open+0x1b0/0x220 [ 414.528023][ T2179] do_dentry_open+0x30c/0x594 [ 414.528045][ T2179] vfs_open+0x34/0x44 [ 414.528053][ T2179] path_openat+0x914/0xb08 [ 414.528062][ T2179] do_filp_open+0xc0/0x170 [ 414.528068][ T2179] do_sys_openat2+0x94/0x18c [ 414.528076][ T2179] __arm64_sys_openat+0x78/0xa4 [ 414.528084][ T2179] invoke_syscall+0x48/0x10c [ 414.528094][ T2179] el0_svc_common+0xbc/0x104 [ 414.528099][ T2179] do_el0_svc+0x34/0xd8 [ 414.528103][ T2179] el0_svc+0x34/0xc4 [ 414.528125][ T2179] el0t_64_sync_handler+0x8c/0xfc [ 414.528133][ T2179] el0t_64_sync+0x1a0/0x1a4 [ 414.528142][ T2179] Kernel panic - not syncing: panic_on_warn set ... So, I reposition and add pcm_mutex to resolve lockdep error. Signed-off-by: Shinhyung Kang <s47.kang(a)samsung.com> Link: https://lore.kernel.org/r/016401d90ac4$7b6848c0$7238da40$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-compress.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 870f13e1d389c..7bce5088b4554 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -149,6 +149,8 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) if (ret < 0) goto be_err; + mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); + /* calculate valid and active FE <-> BE dpcms */ dpcm_process_paths(fe, stream, &list, 1); fe->dpcm[stream].runtime = fe_substream->runtime; @@ -184,7 +186,6 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; - mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); snd_soc_runtime_activate(fe, stream); mutex_unlock(&fe->card->pcm_mutex); @@ -215,7 +216,6 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); snd_soc_runtime_deactivate(fe, stream); - mutex_unlock(&fe->card->pcm_mutex); fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; @@ -234,6 +234,8 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) dpcm_be_disconnect(fe, stream); + mutex_unlock(&fe->card->pcm_mutex); + fe->dpcm[stream].runtime = NULL; snd_soc_link_compr_shutdown(cstream, 0); @@ -409,8 +411,9 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, ret = snd_soc_link_compr_set_params(cstream); if (ret < 0) goto out; - + mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START); + mutex_unlock(&fe->card->pcm_mutex); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; out: -- 2.39.0

1 0

Alsamixer not detecting headphones
by GitHub issues - opened 27 Feb '23

27 Feb '23

alsa-project/alsa-ucm-conf issue #283 was opened from lgmventura: I'm using the motherboard ASUS ROG Strix B760-f Gaming Wifi, which has 7.1 audio channels on the back side and connectors for an audio panel at the front of the case - a microphone and a headphones jack. When I plug my headphones into the front channels connector (at the back of the PC case) there is normal sound. When I plug it into the headphones jack (at the front of the case), it gets automatically detected by `pavucontrol`, which switches to the headphones as output, as it should be: ![image](https://user-images.githubusercontent.com/16710238/221439991-903fd90b-43b2-41a0-af7f-1ed02085238e.png) However, there is no sound. When I open `alsamixer` and choose the respective soundcard, which is weirdly called "USB audio" there, I see no volume control for the headphones, as there should be. All it shows is this: ![image](https://user-images.githubusercontent.com/16710238/221440193-381eadfb-fed9-48ad-807b-05c736b65b24.png) The only way I got to get audio output from the headphones jack is by running `aplay` and choosing the hardware directly from there `aplay -D plughw:1,1 /usr/share/sounds/alsa/Front_Center.wav` or on Audacity by selecting "USB Audio 1: (hw:1,1)" as playback device. I also tried to use JACK Audio Connection Kit, but found no configuration there that routes the audio to the headphones jack, even trying all 4 possible analogue outputs. System: - Motherboard ASUS ROG Strix B760-f Gaming Wifi - Intel core i7-13700K - Ubuntu 22.04 (freshly installed) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/283 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Alsamixer not detecting headphones
by GitHub issues - edited 27 Feb '23

27 Feb '23

alsa-project/alsa-ucm-conf issue #283 was edited from lgmventura: I'm using the motherboard ASUS ROG Strix B760-f Gaming Wifi, which has 7.1 audio channels on the back side and connectors for an audio panel at the front of the case - a microphone and a headphones jack. When I plug my headphones into the front channels connector (at the back of the PC case) there is normal sound. When I plug it into the headphones jack (at the front of the case), it gets automatically detected by `pavucontrol`, which switches to the headphones as output, as it should be: ![image](https://user-images.githubusercontent.com/16710238/221439991-903fd90b-43b2-41a0-af7f-1ed02085238e.png) However, there is no sound. When I open `alsamixer` and choose the respective soundcard, which is weirdly called "USB audio" there, I see no volume control for the headphones, as there should be. All it shows is this: ![image](https://user-images.githubusercontent.com/16710238/221440193-381eadfb-fed9-48ad-807b-05c736b65b24.png) The only way I got to get audio output from the headphones jack is by running `aplay` and choosing the hardware directly from there `aplay -D plughw:1,1 /usr/share/sounds/alsa/Front_Center.wav` or on Audacity by selecting "USB Audio 1: (hw:1,1)" as playback device. I also tried to use JACK Audio Connection Kit, but found no configuration there that routes the audio to the headphones jack, even trying all 4 possible analogue outputs. System: - Motherboard ASUS ROG Strix B760-f Gaming Wifi - Intel core i7-13700K - Ubuntu 22.04 (freshly installed) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/283 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Alsamixer not detecting headphones
by GitHub issues - opened 27 Feb '23

27 Feb '23

alsa-project/alsa-lib issue #304 was opened from lgmventura: I'm using the motherboard ASUS ROG Strix B760-f Gaming Wifi, which has 7.1 audio channels on the back side and connectors for an audio panel at the front of the case - a microphone and a headphones jack. When I plug my headphones into the front channels connector (at the back of the PC case) there is normal sound. When I plug it into the headphones jack (at the front of the case), it gets automatically detected by `pavucontrol`, which switches to the headphones as output, as it should be: ![image](https://user-images.githubusercontent.com/16710238/221439991-903fd90b-43b2-41a0-af7f-1ed02085238e.png) However, there is no sound. When I open `alsamixer` and choose the respective soundcard, which is weirdly called "USB audio" there, I see no volume control for the headphones, as there should be. All it shows is this: ![image](https://user-images.githubusercontent.com/16710238/221440193-381eadfb-fed9-48ad-807b-05c736b65b24.png) The only way I got to get audio output from the headphones jack is by running `aplay` and choosing the hardware directly from there `aplay -D plughw:1,1 /usr/share/sounds/alsa/Front_Center.wav` or on Audacity by selecting "USB Audio 1: (hw:1,1)" as playback device. I also tried to use JACK Audio Connection Kit, but found no configuration there that routes the audio to the headphones jack, even trying all 4 possible analogue outputs. System: - Motherboard ASUS ROG Strix B760-f Gaming Wifi - Intel core i7-13700K - Ubuntu 22.04 (freshly installed) Issue URL : https://github.com/alsa-project/alsa-lib/issues/304 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

infinite loop in src/pcm/pcm_params.c function snd_pcm_sw_params_default()
by bumpelo＠gmail.com 25 Feb '23

25 Feb '23

I found this infinite loop while playing with the test/latency.c function. When I specify too small of a minimum latency (in my case -m 256 or smaller) it winds up: at line line 2337: params->boundary = pcm->buffer_size; while (params->boundary * 2 <= LONG_MAX - pcm->buffer_size) params->boundary *= 2; return 0; but when pcm->buffer_size is zero this is an infinite loop. It does this *after* setting the scheduling to round-robin which makes the infinite loop so much more painful. The return value from this function is ignored where it's called so I'm not sure what the best fix is. I'm guessing it's a rare case since it's not been previously reported but it deserves fixing anyway.

2 1

xone mixer
by GitHub issues - opened 25 Feb '23

25 Feb '23

alsa-project/alsa-lib issue #303 was opened from molli1312: Having a few problems with the Xone32c though not sure if this is an ALSA issue or general USB stack problem. It's a USB soundcard that is supposed to have 4 inputs and 4 outputs. It seems to have 2 USB configurations (bNumConfigurations : 0x2). When plugged in, bConfigurationValue is 0x1 and only 2 inputs and 2 outputs show up. Change this value to 0x2 and it shows up (as it should) as 4 inputs and 4 outputs. I can change the bConfigurationValue with this udev rule, or with a `echo 2 /sys/bus/usb/devices/1-1.4/bConfigurationValue`: This has all been working OK on Ubuntu 20.04 running kernel 5.4.0-33-generic. However, I just updated to 5.4.0-37-generic and now there is only a single USB configuration (bNumConfigurations : 0x1) and bConfigurationValue can't be changed to 0x2: Issue URL : https://github.com/alsa-project/alsa-lib/issues/303 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

OpenBSD (and maybe other *BSDs) support (2)
by GitHub pull_request - opened 25 Feb '23

25 Feb '23

alsa-project/alsa-utils pull request #193 was opened from jg1uaa: #186 addendum. forget to fix alsactl/lock.c, this is the fix. Request URL : https://github.com/alsa-project/alsa-utils/pull/193 Patch URL : https://github.com/alsa-project/alsa-utils/pull/193.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

hda-verb improvements for Asus Zenbook Pro Duo UX582LR
by Marcel Brouillet 25 Feb '23

25 Feb '23

Hello, I have a Asus Zenbook Pro Duo UX582LR that does not have sound out-of-the box. But with either this : hda-verb /dev/snd/hwC0D0 0x20 0x500 0xf hda-verb /dev/snd/hwC0D0 0x20 0x400 0x7774 hda-verb /dev/snd/hwC0D0 0x20 0x500 0x45 hda-verb /dev/snd/hwC0D0 0x20 0x400 0x5289 Or this : hda-verb /dev/snd/hwC0D0 0x20 0x500 0x1b hda-verb /dev/snd/hwC0D0 0x20 0x477 0x4a4b hda-verb /dev/snd/hwC0D0 0x20 0x500 0xf hda-verb /dev/snd/hwC0D0 0x20 0x477 0x74 Then 1. I do get output from the laptop speakers ✅ 2. I do get output through a headset when pluged in (and no sound through speakers) ✅ 3. I do get input from laptop mics, *but far too high gain. *Setting the mic level to 35% in Pulse Audio is roughly good 4. Input *remains from laptop mics even when I plug in headsets* (TRRS jack) I've been OK with this for 18 months, but decided to chip in the time to solve 3 and 4 and make it available to the community. I looked at https://github.com/alsa-project/alsa-tools/tree/master/hda-verb and understand the hda-verb syntax. But *where do I find the verb arguments (strings or the above cryptic 0x477 or 0x400) ?* Do I need to read this 220 pages document ? https://www.intel.com/content/www/us/en/standards/high-definition-audio-spe… *If anyone is willing to direct me to the few verbs to try, and please tell me if there is any risk of killing my sound interface with wrong parameters, a big thank you to you.* Regards, Marcel. Hardware : Asus Zenbook Pro Duo UX582LR Software : Ubuntu Studio 22.04.2 LTS jammy. Fresh installation 10 days ago, fully updated upgraded. alsa-base/jammy,jammy,now 1.0.25+dfsg-0ubuntu7 all [installed,automatic] alsa-tools/jammy,now 1.2.5-2 amd64 [installed,automatic] alsa-tools-gui/jammy,now 1.2.5-2 amd64 [installed,automatic] alsa-topology-conf/jammy,jammy,now 1.2.5.1-2 all [installed,automatic] alsa-ucm-conf/jammy-updates,jammy-updates,now 1.2.6.3-1ubuntu1.4 all [installed,automatic] alsa-utils/jammy,now 1.2.6-1ubuntu1 amd64 [installed,automatic]

1 0

[PATCH 1/3] ASoC: apple: mca: Fix final status read on SERDES reset
by Martin Povišer 24 Feb '23

24 Feb '23

From within the early trigger we are doing a reset of the SERDES unit, but the final status read is on a bad address. Add the missing SERDES unit offset in calculation of the address. Fixes: 3df5d0d97289 ("ASoC: apple: mca: Start new platform driver") Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- sound/soc/apple/mca.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/apple/mca.c b/sound/soc/apple/mca.c index 24381c42eb54..9cceeb259952 100644 --- a/sound/soc/apple/mca.c +++ b/sound/soc/apple/mca.c @@ -210,7 +210,7 @@ static void mca_fe_early_trigger(struct snd_pcm_substream *substream, int cmd, SERDES_CONF_SOME_RST); readl_relaxed(cl->base + serdes_conf); mca_modify(cl, serdes_conf, SERDES_STATUS_RST, 0); - WARN_ON(readl_relaxed(cl->base + REG_SERDES_STATUS) & + WARN_ON(readl_relaxed(cl->base + serdes_unit + REG_SERDES_STATUS) & SERDES_STATUS_RST); break; default: -- 2.33.0

2 3