July 2022 - Alsa-devel - mailman.alsa-project.org

Audio interface unable to be loaded in 1.2.7.2
by GitHub issues - edited 17 Jul '22

17 Jul '22

alsa-project/alsa-ucm-conf issue #190 was edited from KevHg: I upgraded from 1.2.6.1 to 1.2.7.2 and found that pulseaudio no longer detects my Motu M2 interface. Main error log seems to be: `module-alsa-card.c: Failed to find a working profile.` `failed to load module module-alsa-card` I'm new to debugging Linux issues, let me know what other info I can provide to help out on this topic. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/190 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Audio interface unable to be loaded in 1.2.7.2
by GitHub issues - opened 17 Jul '22

17 Jul '22

alsa-project/alsa-ucm-conf issue #190 was opened from KevHg: I upgraded from 1.2.6.1 to 1.2.7.2 and found that pulseaudio no longer detects my Motu M2 interface. Main error log seems to be: `module-alsa-card.c: Failed to find a working profile.` `failed to load module module-alsa-card` I'm new to debugging Linux issues, let me know what other info I can provide to help out on this topic. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/190 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH alsa-utils] aplay/aplay.c: make UCM support optional
by Thomas Petazzoni 17 Jul '22

17 Jul '22

Commit 90f59671784a7e47b40485095cd66892d4840ed7 ("topology: pre-processor: Move the call to expand variables") modified aplay by unconditionally invoking features of the use case manager (UCM) from alsa-lib. However, alsa-lib can be compiled with UCM support. In order to properly support this situation, this commit changes aplay to only conditionally compile the UCM related code. Signed-off-by: Thomas Petazzoni <thomas.petazzoni(a)bootlin.com> --- aplay/aplay.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/aplay/aplay.c b/aplay/aplay.c index b3b3635..59f937d 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -41,7 +41,9 @@ #include <time.h> #include <locale.h> #include <alsa/asoundlib.h> +#ifdef HAVE_ALSA_USE_CASE_H #include <alsa/use-case.h> +#endif #include <assert.h> #include <termios.h> #include <signal.h> @@ -453,6 +455,7 @@ static ssize_t xwrite(int fd, const void *buf, size_t count) return offset; } +#ifdef HAVE_ALSA_USE_CASE_H static int open_ucm(snd_use_case_mgr_t **uc_mgr, char **pcm_name, const char *name) { char *s, *p; @@ -476,6 +479,7 @@ static int open_ucm(snd_use_case_mgr_t **uc_mgr, char **pcm_name, const char *na } return err; } +#endif static long parse_long(const char *str, int *err) { @@ -553,7 +557,9 @@ int main(int argc, char *argv[]) int do_device_list = 0, do_pcm_list = 0, force_sample_format = 0; snd_pcm_info_t *info; FILE *direction; +#ifdef HAVE_ALSA_USE_CASE_H snd_use_case_mgr_t *uc_mgr = NULL; +#endif #ifdef ENABLE_NLS setlocale(LC_ALL, ""); @@ -852,6 +858,7 @@ int main(int argc, char *argv[]) goto __end; } +#ifdef HAVE_ALSA_USE_CASE_H if (strncmp(pcm_name, "ucm.", 4) == 0) { err = open_ucm(&uc_mgr, &pcm_name, pcm_name + 4); if (err < 0) { @@ -861,6 +868,7 @@ int main(int argc, char *argv[]) if (verbose) fprintf(stderr, _("Found UCM PCM device: %s\n"), pcm_name); } +#endif err = snd_pcm_open(&handle, pcm_name, stream, open_mode); if (err < 0) { @@ -951,8 +959,10 @@ int main(int argc, char *argv[]) if (verbose==2) putchar('\n'); snd_pcm_close(handle); +#ifdef HAVE_ALSA_USE_CASE_H if (uc_mgr) snd_use_case_mgr_close(uc_mgr); +#endif handle = NULL; free(audiobuf); __end: -- 2.36.1

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[PATCH alsa-utils] alsactl/info.c: fix conditionals on __ALSA_PCM_H and __ALSA_RAWMIDI_H
by Thomas Petazzoni 17 Jul '22

17 Jul '22

Commit bbc74a61ac7c35e506c3d7f76ecf943cb55736a6 ("alsactl: implement 'info' command") implemented an alsactl info command. In this implementation, there was an attempt to properly address optional features from alsa-lib by using conditions on __ALSA_PCM_H, __ALSA_RAWMIDI_H. Unfortunately, this attempt does not work entirely: only the code inside pcm_device_list(), rawmidi_device_list() was conditionally compiled, but their very prototype also use type definitions provided in pcm.h and rawmidi.h. So really, it's the entire function that needs to be conditionally implemented. Also, snd_rawmidi_stream_name() was not handled properly, for the same reason. This commit implements pcm_device_list() only if __ALSA_PCM_H is defined, and implements snd_rawmidi_stream_name() and rawmidi_device_list() only if __ALSA_RAWMIDI_H is defined. general_card_info() is modified to not call the PCM or raw MIDI functions when support is not available. Signed-off-by: Thomas Petazzoni <thomas.petazzoni(a)bootlin.com> --- alsactl/info.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/alsactl/info.c b/alsactl/info.c index 253539d..9bd72af 100644 --- a/alsactl/info.c +++ b/alsactl/info.c @@ -22,9 +22,9 @@ #include "aconfig.h" #include "alsactl.h" +#ifdef __ALSA_PCM_H static int pcm_device_list(snd_ctl_t *ctl, snd_pcm_stream_t stream, bool *first) { -#ifdef __ALSA_PCM_H int err, dev, idx; unsigned int count; snd_pcm_info_t *pcminfo; @@ -76,10 +76,12 @@ static int pcm_device_list(snd_ctl_t *ctl, snd_pcm_stream_t stream, bool *first) idx, snd_pcm_info_get_subdevice_name(pcminfo)); } } -#endif + return 0; } +#endif +#ifdef __ALSA_RAWMIDI_H static const char *snd_rawmidi_stream_name(snd_rawmidi_stream_t stream) { if (stream == SND_RAWMIDI_STREAM_INPUT) @@ -91,7 +93,6 @@ static const char *snd_rawmidi_stream_name(snd_rawmidi_stream_t stream) static int rawmidi_device_list(snd_ctl_t *ctl, snd_rawmidi_stream_t stream, bool *first) { -#ifdef __ALSA_RAWMIDI_H int err, dev, idx; unsigned int count; snd_rawmidi_info_t *info; @@ -143,9 +144,10 @@ static int rawmidi_device_list(snd_ctl_t *ctl, snd_rawmidi_stream_t stream, bool idx, snd_rawmidi_info_get_subdevice_name(info)); } } -#endif + return 0; } +#endif static int hwdep_device_list(snd_ctl_t *ctl) { @@ -227,17 +229,21 @@ int general_card_info(int cardno) } err = card_info(ctl); +#ifdef __ALSA_PCM_H first = true; if (err >= 0) err = pcm_device_list(ctl, SND_PCM_STREAM_PLAYBACK, &first); if (err >= 0) err = pcm_device_list(ctl, SND_PCM_STREAM_CAPTURE, &first); +#endif +#ifdef __ALSA_RAWMIDI_H first = true; if (err >= 0) err = rawmidi_device_list(ctl, SND_PCM_STREAM_PLAYBACK, &first); if (err >= 0) err = rawmidi_device_list(ctl, SND_PCM_STREAM_CAPTURE, &first); +#endif if (err >= 0) err = hwdep_device_list(ctl); -- 2.36.1

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Re: [PATCH 6/6] i2c: Make remove callback return void
by Pavel Machek 17 Jul '22

17 Jul '22

Hi! > From: Uwe Kleine-König <uwe(a)kleine-koenig.org> > > The value returned by an i2c driver's remove function is mostly ignored. > (Only an error message is printed if the value is non-zero that the > error is ignored.) > > So change the prototype of the remove function to return no value. This > way driver authors are not tempted to assume that passing an error to > the upper layer is a good idea. All drivers are adapted accordingly. > There is no intended change of behaviour, all callbacks were prepared to > return 0 before. > > Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> 2-4: Acked-by: Pavel Machek <pavel(a)ucw.cz> Best regards, Pavel -- People of Russia, stop Putin before his war on Ukraine escalates.

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Inappropriate Envy24 rate locking/XRUNS for playback and/or recording
by GitHub issues - opened 17 Jul '22

17 Jul '22

alsa-project/alsa-tools issue #11 was opened from foobers: On Debian 'oldstable', there's a rate-locking and 'low pitch' problem with following setup (same on Mint/Unstable): - dedicated audio card: Multimedia audio controller: VIA Technologies Inc. ICE1712 [Envy24] PCI Multi-Channel I/O Controller (rev 02) based - integrated audio in graphics card: Audio device: Advanced Micro Devices, Inc. [AMD/ATI] Device xxxx - onboard audio chip: Audio device: Advanced Micro Devices, Inc. [AMD/ATI] xxx (Intel HDA) (xxxx) Initially, in 'alsamixer', there are two values shown for 'Multi Track Internal Clock': '[44100]' and 'Default [41000]' for the dedicated AC. In alsamixer, it's only possible, to change the first value, but without any noticable effect (only using 'Envy24 Control' or 'Mudita24' seem to have). Now, using the audio card, sometimes, right from the start, playback of internet audio streams (in Firefox) seems to have a wrong sampling rate set (i. e. probably 48 kHz), resulting in audibly lower 'pitch' and sound distortion. Even worse, playback isn't always affected from start, but recordings from 'gnome-sound-recorder' will be in either case. This can result in several hours of recorded audio being borked, since there's no controlling of the outcome, until recording is stopped, which is very undesirable, especially in ad-hoc recording situations. I was able to track down the problem being linked to 'Master clock', 'Rate state' settings 'Locked' and 'Reset' in 'Envy24 Control Utility': checking and unchecking / switching forth to 48kHz and back to 44100Hz, helped resetting both playback and recording sampling rate to 44100Hz, to perform regular audio recordings with 'gnome-sound-recorder'. The rate locking problem can also occur, after playing MP4 videos with Mplayer (media info: sampling rate 44100Hz); then, audio files (media info: bit rate 112Kbit/s, sampling rate 44100Hz), previously kown-good audio records will be played at a noticably lower 'pitch', also recording (with gnome-recorder) now will record at a lower pitch, i. e wrong rate. In alsamixer, there are now two different values shown for 'Multi Track Internal Clock': '[44100]' and 'Default [48000]'. Also, when trying to play a regular film DVD with VLC now, it very likely shows short dropouts of black screen and audio, at steady intervals of ~2s, requiring performing a logoff and login, to reset rates again. Interestingly (and unexpectedly), setting 'Master clock' to 'S/PDIF in', seemed to internally freeze the clock / source to 'S/PDIF', in effect preventing any more sound output or recording from the device; trying to switch back to 44110Hz or 48000Hz would fail and the could only be reset, after until doing a complete reboot. I've seen references, pointing to the rate locking issue, which suggest, that unchecking 'locked' and 'reset' for 'Rate state' in 'Envy24' would be the key here - only, on my system, those values didn't seem to be initially checked (at least not visually): https://discourse.ardour.org/t/all-sound-is-low-pitched/82149 So the solution here would seem to be, to ensure unset 'Rate state' options at boot time; in contrast to this suggestion, the documentation says, 'locked' might cause 'errors' and 'XRUNS', and 'reset' only would be the recommended setting by default, to allow applications alter the clock sample rate to their needs, resetting them to the selected 'Master clock' default afterwards: https://wiki.archlinux.org/title/Envy24control#Rate_State While this seems to make sense, 'reset' not being set appropriatly, might also explain some of the weird behavior described above, namely, when some application sets and locks 48kHz for playback and/or input channels, which then don't match the input and/or output of another application, started after that. The effect of VLC/MPlayer showing 'dropouts' may also have its cause in the mentioned 'XRUNS', in this case, supposedly buffer underruns, when applying an 48kHz sampling rate to video files with 44,1kHz audio streams, or buffer overflows, when applying 44,1kHz sampling to files with 48kHz audio streams: https://unix.stackexchange.com/questions/199498/what-are-xruns What (combination) of 'Rate state' settings (if any) would be a 'bullet-proof' remedy here, I cannot tell, since my settings obviously both empty. So, either this was a false visual feedback, or 'reset' does not work as intended. So far, I can't confirm, if these settings differ in Mint, where the problem shows, too, as said (or if the defaults have been altered). Currently I can't try perma-setting 'reset' only (the recommended default), but I think it didn't work out before for recording. Issue URL : https://github.com/alsa-project/alsa-tools/issues/11 Repository URL: https://github.com/alsa-project/alsa-tools

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[PATCH] ASoC: DPCM: Don't pick up BE without substream
by Takashi Iwai 17 Jul '22

17 Jul '22

When DPCM tries to add valid BE connections at dpcm_add_paths(), it doesn't check whether the picked BE actually supports for the given stream direction. Due to that, when an asymmetric BE stream is present, it picks up wrongly and this may result in a NULL dereference at a later point where the code assumes the existence of a corresponding BE substream. This patch adds the check for the presence of the substream for the target BE for avoiding the problem above. Note that we have already some fix for non-existing BE substream at commit 6246f283d5e0 ("ASoC: dpcm: skip missing substream while applying symmetry"). But the code path we've hit recently is rather happening before the previous fix. So this patch tries to fix at picking up a BE instead of parsing BE lists. Fixes: bbf7d3b1c4f4 ("ASoC: soc-pcm: align BE 'atomicity' with that of the FE") Reported-by: Alex Natalsson <harmoniesworlds(a)gmail.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/CADs9LoPZH_D+eJ9qjTxSLE5jGyhKsjMN7g2NighZ16biVxsy… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a827cc3c158a..0c1de5624842 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1318,6 +1318,9 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (!be->dai_link->no_pcm) continue; + if (!snd_soc_dpcm_get_substream(be, stream)) + continue; + for_each_rtd_dais(be, i, dai) { w = snd_soc_dai_get_widget(dai, stream); -- 2.35.3

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Audio interface unable to be loaded in 1.2.7.2
by GitHub issues - opened 17 Jul '22

17 Jul '22

alsa-project/alsa-lib issue #254 was opened from KevHg: I upgraded from 1.2.6.1 to 1.2.7.2 and found that pulseaudio no longer detects my Motu M2 interface. Main error log seems to be: `module-alsa-card.c: Failed to find a working profile.` `failed to load module module-alsa-card` I'm new to debugging Linux issues, let me know what other info I can provide to help out on this topic. Issue URL : https://github.com/alsa-project/alsa-lib/issues/254 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH 0/5] ALSA: Drop async signal support
by Takashi Iwai 16 Jul '22

16 Jul '22

Hi, this is a patch series to drop one of unused features, the async signal support, from ALSA core. The async signal itself is very difficult to use properly due to various restrictions (e.g. you cannot perform any I/O in the context), hence it's a feature that has been never used by real applications. OTOH, the real problem is that there have been quite a few syzcaller reports indicating that fasync code path may lead to some potential deadlocks for long time. Dropping the feature is the easiest solution, obviously. The corresponding update for alsa-lib will follow once when we agree with this approach. thanks, Takashi === Takashi Iwai (5): ALSA: timer: Drop async signal support ALSA: pcm: Drop async signal support ALSA: control: Drop async signal support ALSA: core: Drop async signal support ALSA: doc: Drop stale fasync entry .../kernel-api/writing-an-alsa-driver.rst | 1 - include/sound/control.h | 1 - include/sound/pcm.h | 1 - sound/core/control.c | 11 ----------- sound/core/init.c | 11 +---------- sound/core/pcm_lib.c | 4 +--- sound/core/pcm_native.c | 18 ------------------ sound/core/timer.c | 13 ------------- 8 files changed, 2 insertions(+), 58 deletions(-) -- 2.35.3

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[PATCH 0/4] ASoC: SOF: Intel: add support for SoundWire-based HP Omen16
by Pierre-Louis Bossart 15 Jul '22

15 Jul '22

This device exposes a headset codec on link0 and an amplifier on link3. This is a very unusual pin-muxing, usually the microphones are pin-muxed with link2/link3. This resulted in a problematic error handling leading to a kernel oops, and invalidated a hard-coded assumption. Full support for this device requires a DMI quirk shared separately ("soundwire: dmi-quirks: add remapping for HP Omen 16-k0005TX"). Pierre-Louis Bossart (4): ASoC: Intel: sof_sdw: avoid oops in error handling ASoC: Intel: soc-acpi: add table for HP Omen 16-k0005TX ASoC: Intel: sof_sdw: add quirk for HP Omen 16-k0005TX ASoC: SOF: Intel: enable dmic handling with 2 or fewer SoundWire links sound/soc/intel/boards/sof_sdw.c | 9 ++++++ sound/soc/intel/boards/sof_sdw_rt711.c | 3 ++ sound/soc/intel/boards/sof_sdw_rt711_sdca.c | 3 ++ .../intel/common/soc-acpi-intel-adl-match.c | 29 +++++++++++++++++++ sound/soc/sof/intel/hda.c | 8 ++--- 5 files changed, 48 insertions(+), 4 deletions(-) -- 2.34.1

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