pulseaudio issueshttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues2020-01-01T05:07:53Zhttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/233Pulseaudio not recognize active session correctly under some circumstance2020-01-01T05:07:53ZBugzilla Migration UserPulseaudio not recognize active session correctly under some circumstance## Submitted by Kevin
Assigned to **pul..@..op.org**
**[Link to original bug (#76672)](https://bugs.freedesktop.org/show_bug.cgi?id=76672)**
## Description
I get no sound in my desktop when using the following (a bit usual) way to...## Submitted by Kevin
Assigned to **pul..@..op.org**
**[Link to original bug (#76672)](https://bugs.freedesktop.org/show_bug.cgi?id=76672)**
## Description
I get no sound in my desktop when using the following (a bit usual) way to start the desktop:
from a virtual terminal, say vt2, run:
(setsid startx -- vt7 &).
That is, startx in a virtual terminal other than the current one. The "setsid" is not relevant here, without it, (startx -- vt7 &) don't work, either.
But, if I play something in the desktop and then SWITCH BACK to vt2, the sound comes. If I log into another virtual terminal, for example, vt1, the sound comes, too. Just as long as I switch back to desktop on vt7, the sound stops.
I guess this is because pulseaudio fails to recognise the active session: if I run `pacmd list-sinks` in the desktop(vt7), it shows that the status of the active sink is "SUSPENDED" and the suspend cause is "SESSION" while running the same command in vt2 shows the status is "RUNNING".
Also, `loginctl user-status` shows that, all processes of my desktop(vt7) and vt2 belong to the same session (in systemd's notion).
And if I start X by:
(setsid startx -- vt2 &)
that is, in the same virtual terminal, then all problems are gone, pulseaudio works perfectly.
I'm using arch linux, mate desktop(1.8.0), pulseaudio(5.0), all packages are from arch's official repository.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/219[cleanup] When creating a new sink input, the core could request a rewind2018-07-30T10:07:38ZBugzilla Migration User[cleanup] When creating a new sink input, the core could request a rewind## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#54243)](https://bugs.freedesktop.org/show_bug.cgi?id=54243)**
## Description
Comment in the PA_SINK_MESSAGE_ADD_INPUT handler in sink....## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#54243)](https://bugs.freedesktop.org/show_bug.cgi?id=54243)**
## Description
Comment in the PA_SINK_MESSAGE_ADD_INPUT handler in sink.c:
/* We don't rewind here automatically. This is left to the
* sink input implementor because some sink inputs need a
* slow start, i.e. need some time to buffer client
* samples before beginning streaming. */
Not doing the rewind automatically has led to a situation where every filter sink (and some other sink input implementations, like module-sine) has this sink input state change callback:
static void sink_input_state_change_cb(pa_sink_input *i, pa_sink_input_state_t state) {
struct userdata *u;
pa_sink_input_assert_ref(i);
pa_assert_se(u = i->userdata);
pa_log_debug("Sink input %d state %d", i->index, state);
/* If we are added for the first time, ask for a rewinding so that
* we are heard right-away. */
if (PA_SINK_INPUT_IS_LINKED(state) &&
i->thread_info.state == PA_SINK_INPUT_INIT) {
pa_log_debug("Requesting rewind due to state change.");
pa_sink_input_request_rewind(i, 0, FALSE, TRUE, TRUE);
}
}
Repeating that in every filter sink shouldn't be needed. The core could request the rewind itself, without pushing the responsibility to the sink input implementors. Avoiding the rewind when it's not needed is only an optimization. In my opinion the optimization is good to have, but it could be implemented by having a sink input flag that says that this input "needs a slow start".https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/218pulseaudio crashed with SIGABRT in pa_hook_fire()2019-12-27T05:10:36ZBugzilla Migration Userpulseaudio crashed with SIGABRT in pa_hook_fire()## Submitted by Cristian Aravena
Assigned to **pul..@..op.org**
**[Link to original bug (#88979)](https://bugs.freedesktop.org/show_bug.cgi?id=88979)**
## Description
Open bug in launchpad.net
https://bugs.launchpad.net/ubuntu/+so...## Submitted by Cristian Aravena
Assigned to **pul..@..op.org**
**[Link to original bug (#88979)](https://bugs.freedesktop.org/show_bug.cgi?id=88979)**
## Description
Open bug in launchpad.net
https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/1416228
```
#0 0x00007fa658bb8e37 in __GI_raise (sig=sig@entry=6) at ../nptl/sysdeps/unix/sysv/linux/raise.c:56
resultvar = 0
pid = 2769
selftid = 2769
#1 0x00007fa658bba528 in __GI_abort () at abort.c:89
save_stage = 2
act = {__sigaction_handler = {sa_handler = 0x262f67e, sa_sigaction = 0x262f67e}, sa_mask = {__val = {140352445554172, 107491743040, 140734008247600, 140734008247592, 1, 206158430316, 140350941298689, 140733193388038, 40040062, 40040057, 10213294085693711360, 3, 40535904, 39899968, 140734008247952, 39931844}}, sa_flags = 1132253023, sa_restorer = 0x7fa6437d2d80}
sigs = {__val = {32, 0 <repeats 15 times>}}
#2 0x00007fa6437ccf68 in ?? () from /usr/lib/pulse-4.0/modules/module-bluetooth-device.so
No symbol table info available.
#3 0x00007fa65a3afffc in pa_hook_fire () from /usr/lib/libpulsecore-4.0.so
No symbol table info available.
#4 0x00007fa64aaaeda6 in ?? () from /usr/lib/pulse-4.0/modules/libbluetooth-util.so
No symbol table info available.
#5 0x00007fa64aab26c3 in ?? () from /usr/lib/pulse-4.0/modules/libbluetooth-util.so
No symbol table info available.
#6 0x00007fa659a8b161 in dbus_connection_dispatch () from /lib/x86_64-linux-gnu/libdbus-1.so.3
No symbol table info available.
#7 0x00007fa65a163114 in ?? () from /usr/lib/x86_64-linux-gnu/pulseaudio/libpulsecommon-4.0.so
No symbol table info available.
#8 0x00007fa659ef0dca in pa_mainloop_dispatch () from /usr/lib/x86_64-linux-gnu/libpulse.so.0
No symbol table info available.
#9 0x00007fa659ef0fdc in pa_mainloop_iterate () from /usr/lib/x86_64-linux-gnu/libpulse.so.0
No symbol table info available.
#10 0x00007fa659ef1080 in pa_mainloop_run () from /usr/lib/x86_64-linux-gnu/libpulse.so.0
No symbol table info available.
#11 0x0000000000407058 in main ()
No symbol table info available.
```https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/213Virtual sink and source flags are not updated when moving them to different m...2019-12-02T14:33:57ZBugzilla Migration UserVirtual sink and source flags are not updated when moving them to different master device## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#39405)](https://bugs.freedesktop.org/show_bug.cgi?id=39405)**
## Description## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#39405)](https://bugs.freedesktop.org/show_bug.cgi?id=39405)**
## Descriptionhttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/209No way to setup profile auto-switching on headphone connect2018-07-30T10:05:53ZBugzilla Migration UserNo way to setup profile auto-switching on headphone connect## Submitted by Rion
Assigned to **pul..@..op.org**
**[Link to original bug (#94413)](https://bugs.freedesktop.org/show_bug.cgi?id=94413)**
## Description
Hi
I have a laptop with speakers and analog headphone jack.
Usually I also...## Submitted by Rion
Assigned to **pul..@..op.org**
**[Link to original bug (#94413)](https://bugs.freedesktop.org/show_bug.cgi?id=94413)**
## Description
Hi
I have a laptop with speakers and analog headphone jack.
Usually I also have second hdmi-connected monitor.
Sound via hdmi is much better so I usually use it. But sometimes I plug headphone and expect my sound card to switch to fully analog profile.
Unfortunately this does not happen and I continue listening audio via hdmi. So I switch profile manually via pavucontrol every time I need this.
Well it's quite annoying to go through all these settings every time.
I googled about the problem and found this https://www.freedesktop.org/wiki/Software/PulseAudio/RFC/PriorityRouting/
Looks like exactly what I need. But seems it's still not implemented.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/207Assertion '!s->thread_info.rewind_requested' failed at pulsecore/sink.c:1271,...2019-11-27T18:34:12ZBugzilla Migration UserAssertion '!s->thread_info.rewind_requested' failed at pulsecore/sink.c:1271, function pa_sink_render_into_full(). Aborting.## Submitted by Matthijs Kooijman
Assigned to **pul..@..op.org**
**[Link to original bug (#54881)](https://bugs.freedesktop.org/show_bug.cgi?id=54881)**
## Description
Created attachment 67112
pulseaudio -vv output
This assertion...## Submitted by Matthijs Kooijman
Assigned to **pul..@..op.org**
**[Link to original bug (#54881)](https://bugs.freedesktop.org/show_bug.cgi?id=54881)**
## Description
Created attachment 67112
pulseaudio -vv output
This assertion occured when playing an audio stream and starting pavucontrol halfway through the stream. I happened to have pulseaudio running with -vv, so I'm attaching that log output. I unfortunately don't have a stacktrace, nor have I succeeded in reproducing this particular assert.
This assertion was observed running pulseaudio 2.0 from Debian (2.0-3).
Here's the tail of the log:
D: [pulseaudio] protocol-native.c: Client pavucontrol changes volume of sink input 'A Night Like This' by 'Caro Emerald'.
D: [alsa-sink] alsa-sink.c: Requested to rewind 384000 bytes.
D: [alsa-sink] alsa-sink.c: Limited to 3584 bytes.
D: [alsa-sink] alsa-sink.c: before: 896
D: [alsa-sink] alsa-sink.c: after: 896
D: [alsa-sink] alsa-sink.c: Rewound 3584 bytes.
D: [alsa-sink] sink.c: Processing rewind...
D: [alsa-sink] sink-input.c: Have to rewind 3584 bytes on render memblockq.
D: [alsa-sink] module-equalizer-sink.c: Rewind callback!
D: [alsa-sink] sink-input.c: Have to rewind 3584 bytes on render memblockq.
D: [alsa-sink] source.c: Processing rewind...
I: [pulseaudio] module-stream-restore.c: Storing volume/mute/device for stream sink-input-by-media-role:music.
I: [alsa-sink] alsa-sink.c: Underrun!
I: [alsa-sink] alsa-sink.c: Increasing minimal latency to 1.00 ms
D: [alsa-sink] alsa-sink.c: Latency set to 1.00ms
D: [alsa-sink] alsa-sink.c: hwbuf_unused=383808
D: [alsa-sink] alsa-sink.c: setting avail_min=95977
D: [alsa-sink] alsa-sink.c: Requesting rewind due to latency change.
D: [alsa-sink] alsa-sink.c: Latency set to 1.00ms
D: [alsa-sink] alsa-sink.c: hwbuf_unused=383808
D: [alsa-sink] alsa-sink.c: setting avail_min=95977
D: [alsa-sink] alsa-sink.c: Latency set to 1.00ms
D: [alsa-sink] alsa-sink.c: hwbuf_unused=383808
D: [alsa-sink] alsa-sink.c: setting avail_min=95977
D: [alsa-sink] alsa-sink.c: Latency set to 1.00ms
D: [alsa-sink] alsa-sink.c: hwbuf_unused=383808
D: [alsa-sink] alsa-sink.c: setting avail_min=95977
E: [alsa-sink] sink.c: Assertion '!s->thread_info.rewind_requested' failed at pulsecore/sink.c:1271, function pa_sink_render_into_full(). Aborting.
**Attachment 67112**, "pulseaudio -vv output":
[pulse.log](/uploads/f75b41ebad7d10644de579d255f5028d/pulse.log)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/204Synchronous name service queries are done from the main thread -> potential l...2018-07-30T10:05:24ZBugzilla Migration UserSynchronous name service queries are done from the main thread -> potential long delays## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#81802)](https://bugs.freedesktop.org/show_bug.cgi?id=81802)**
## Description
pa_get_fqdn() does a name service query with getaddrinfo(...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#81802)](https://bugs.freedesktop.org/show_bug.cgi?id=81802)**
## Description
pa_get_fqdn() does a name service query with getaddrinfo(). The query is synchronous, and can take a long time, and since pa_get_fqdn() is called from the server's main thread (e.g. module-native-protocol-tcp via pa_socket_server_get_address()), the server will get stalled for the duration of the query. This was originally reported in Tizen's bug tracker: https://bugs.tizen.org/jira/browse/TC-1130
If the query to get the FQDN takes a long time, the network configuration can perhaps be considered broken, but nevertheless, I think we should be resistant to misconfigured networks. pa_get_fqdn() can't be changed to an asynchronous version, because it's part of the stable API, but we can add a new function pa_get_fqdn_async() that uses libasyncns. Then all code that currently uses pa_get_fqdn() should be reviewed and changed to use pa_get_fqdn_async() if necessary.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/203Closing PulseAudio freezes SDL audio2018-07-30T10:05:20ZBugzilla Migration UserClosing PulseAudio freezes SDL audio## Submitted by Jussi Pakkanen
Assigned to **pul..@..op.org**
**[Link to original bug (#103498)](https://bugs.freedesktop.org/show_bug.cgi?id=103498)**
## Description
This is a slightly weird bug. When using SDL mixer (1.2) to pla...## Submitted by Jussi Pakkanen
Assigned to **pul..@..op.org**
**[Link to original bug (#103498)](https://bugs.freedesktop.org/show_bug.cgi?id=103498)**
## Description
This is a slightly weird bug. When using SDL mixer (1.2) to play music on 32 bit x86 Linux boxes, things works perfectly most of the time. However every now and then things break down when closing down the SDL audio device.
The actual freeze happens inside PulseAudio. SDL's deinit calls into PA, which does a pthread_join that never returns. When this happens if there is an another SDL sound player it keeps on working perfectly. Trying to open a new SDL audio player fails. Restarting PA fixes the issue so new music players can be opened, but it does not work reliably.
Detected with PA 5 but replicated as well with PA 10.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/199USB audio gives continous heavily distorted sound (random bug) (Alesis core 1)2022-10-12T06:03:35ZBugzilla Migration UserUSB audio gives continous heavily distorted sound (random bug) (Alesis core 1)## Submitted by tro..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#98863)](https://bugs.freedesktop.org/show_bug.cgi?id=98863)**
## Description
I just bought an Alesis Core 1. (USB audio interface.)
A few tim...## Submitted by tro..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#98863)](https://bugs.freedesktop.org/show_bug.cgi?id=98863)**
## Description
I just bought an Alesis Core 1. (USB audio interface.)
A few times I have returned to the computer and found that the sound output has become heavily! distorted / metallic / crackling. There is silence (as it should be) when no sound is playing.
After unplugging the usb cable and plugging it back in, the sound is OK.
This is PulseAudio 4.0 on Ubuntu Studio 14.04.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/192The ffmpeg resampler implementation could be improved2018-07-30T10:03:45ZBugzilla Migration UserThe ffmpeg resampler implementation could be improved## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#49714)](https://bugs.freedesktop.org/show_bug.cgi?id=49714)**
## Description
It looks like the ffmpeg resampler implementation doesn't...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#49714)](https://bugs.freedesktop.org/show_bug.cgi?id=49714)**
## Description
It looks like the ffmpeg resampler implementation doesn't suffer from [bug 47156](https://bugs.freedesktop.org/show_bug.cgi?id=47156), but the solution for leftover data doesn't seem entirely correct: if there is leftover data, it's not taken into account in pa_resampler_result(), pa_resampler_max_block_size() and pa_resampler_reset().
I originally started to write this bug report, because there seemed to be unnecessary copying of data going on in ffmpeg_resample(), but that's probably not the case, because the ffmpeg resampler handles only mono audio, so multichannel data has to be deinterleaved for processing and again interleaved after processing. If the input is mono to begin with, then there would be room for optimizing for that case still, though.
### Depends on
* [Bug 47156](https://bugs.freedesktop.org/show_bug.cgi?id=47156)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/189Unique names for ports2018-07-30T10:03:31ZBugzilla Migration UserUnique names for ports## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#44793)](https://bugs.freedesktop.org/show_bug.cgi?id=44793)**
## Description
I have written a function:
static const char *get_port_n...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#44793)](https://bugs.freedesktop.org/show_bug.cgi?id=44793)**
## Description
I have written a function:
static const char *get_port_name(pa_device_port *port) {
if (port)
return port->name;
else
return "(no port)"
}
The port name isn't guaranteed to be unique across different cards. I'd like the port name to be an unique identifier, because the result of get_port_name() is used in a log message. If the name is not unique, the log message may be ambiguous. I could change get_port_name() so that it would allocate a new string: pa_sprintf_malloc("%s on %s", port->name, port->card->name), but freeing the string would be inconvenient for the caller.
Now, logging being slightly inconvenient may not be the most convincing argument when the requested change would be relatively big. So, here's another argument: we will anyway have to make the name unique, because it would be awesome if pa_connect_playback() could take a port name in it's dev argument - if the requested port isn't active, it will be automatically made active. And we're going to merge the sink and port concepts anyway, aren't we, so all this makes just perfect sense, right?https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/185unplugging hdmi from intel gpu doesn't redirect audio back to local speakers2018-12-14T08:51:45ZBugzilla Migration Userunplugging hdmi from intel gpu doesn't redirect audio back to local speakers## Submitted by Christian Hergert
Assigned to **pul..@..op.org**
**[Link to original bug (#96211)](https://bugs.freedesktop.org/show_bug.cgi?id=96211)**
## Description
Setup:
Lenovo Gen3 X1 Carbon w/ Intel GPU
Audio redirecte...## Submitted by Christian Hergert
Assigned to **pul..@..op.org**
**[Link to original bug (#96211)](https://bugs.freedesktop.org/show_bug.cgi?id=96211)**
## Description
Setup:
Lenovo Gen3 X1 Carbon w/ Intel GPU
Audio redirected via HDMI to a samsung TV
Both Wayland and Xorg sessions
I manually set the audio output to the HDMI output in GNOME control center. I thought it would remember my setting like it does for monitor layout, but that does not seem to happen. It sounds like is supposed to be handled by PulseAudio?
Likewise, when I remove the HDMI device, I would expect audio to be redirected back to my laptop speakers.
Is it correct that this should be handled by PulseAudio and not by something like gnome-settings-daemon/control-center?https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/184Ability to choose/override the sample rate of a sink2020-10-25T16:38:54ZBugzilla Migration UserAbility to choose/override the sample rate of a sink## Submitted by Jérôme Carretero
Assigned to **pul..@..op.org**
**[Link to original bug (#104781)](https://bugs.freedesktop.org/show_bug.cgi?id=104781)**
## Description
While performing tests on a USB audio DAC, I wanted to select...## Submitted by Jérôme Carretero
Assigned to **pul..@..op.org**
**[Link to original bug (#104781)](https://bugs.freedesktop.org/show_bug.cgi?id=104781)**
## Description
While performing tests on a USB audio DAC, I wanted to select a specific sample rate, but noticed that the only way to do so was by disabling anything that might get plugged on the sink (eg. running pavucontrol creates peak-detect elements, and I've seen it affect the sample rate selection) and playing sounds having that particular sample rate.
Having the ability to force a sink sample rate (and sample type) would be quite practical in this particular (niche I admit) use case, I don't know if there are other relevant ones (testing of resamplers? defective sample rate on a device?).https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/182Assertion 'b' failed at pulsecore/memblock.c:4542018-10-03T07:25:20ZBugzilla Migration UserAssertion 'b' failed at pulsecore/memblock.c:454## Submitted by Jasmin
Assigned to **pul..@..op.org**
**[Link to original bug (#42449)](https://bugs.freedesktop.org/show_bug.cgi?id=42449)**
## Description
Hello,
unfortunately pulseaudio 1.0 exits with the following messages:
...## Submitted by Jasmin
Assigned to **pul..@..op.org**
**[Link to original bug (#42449)](https://bugs.freedesktop.org/show_bug.cgi?id=42449)**
## Description
Hello,
unfortunately pulseaudio 1.0 exits with the following messages:
I: [pulseaudio] client.c: Created 2 "Native client (UNIX socket client)"
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Core1.Client added for object /org/pulseaudio/core1/client2
D: [pulseaudio] protocol-native.c: Protocol version: remote 23, local 23
I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
D: [pulseaudio] protocol-native.c: SHM possible: yes
D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for totem
D: [pulseaudio] module-augment-properties.c: Found /usr/share/applications/totem.desktop.
I: [pulseaudio] client.c: Freed 2 "Totem Video-Player"
I: [pulseaudio] protocol-native.c: Connection died.
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Core1.Client removed from object /org/pulseaudio/core1/client2
I: [pulseaudio] client.c: Created 3 "Native client (UNIX socket client)"
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Core1.Client added for object /org/pulseaudio/core1/client3
D: [pulseaudio] protocol-native.c: Protocol version: remote 23, local 23
I: [pulseaudio] protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
D: [pulseaudio] protocol-native.c: SHM possible: yes
D: [pulseaudio] protocol-native.c: Negotiated SHM: yes
D: [pulseaudio] module-augment-properties.c: Looking for .desktop file for totem
D: [pulseaudio] module-suspend-on-idle.c: Sink ladspa_output.mbeq_1197.mbeq becomes busy.
I: [pulseaudio] resampler.c: Using resampler 'speex-float-3'
I: [pulseaudio] resampler.c: Using float32le as working format.
I: [pulseaudio] resampler.c: Choosing speex quality setting 3.
D: [pulseaudio] memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=8, prebuf=0, minreq=1 maxrewind=0
D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=8, prebuf=0, minreq=8 maxrewind=0
I: [pulseaudio] sink-input.c: Created input 1 "Playback Stream" on ladspa_output.mbeq_1197.mbeq with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: [pulseaudio] sink-input.c: media.name = "Playback Stream"
I: [pulseaudio] sink-input.c: application.name = "Totem Video-Player"
I: [pulseaudio] sink-input.c: native-protocol.peer = "UNIX socket client"
I: [pulseaudio] sink-input.c: native-protocol.version = "23"
I: [pulseaudio] sink-input.c: media.role = "video"
I: [pulseaudio] sink-input.c: application.process.id = "26015"
I: [pulseaudio] sink-input.c: application.process.user = "jasmin"
I: [pulseaudio] sink-input.c: application.process.host = "jasmin-mobil-01"
I: [pulseaudio] sink-input.c: application.process.binary = "totem"
I: [pulseaudio] sink-input.c: application.icon_name = "totem"
I: [pulseaudio] sink-input.c: window.x11.display = ":0.0"
I: [pulseaudio] sink-input.c: application.language = "de_DE.utf8"
I: [pulseaudio] sink-input.c: application.process.machine_id = "7bae1dd2796f04b29e3a6bda0000091d"
I: [pulseaudio] sink-input.c: application.process.session_id = "7bae1dd2796f04b29e3a6bda0000091d-1320085098.612345-2004452429"
I: [pulseaudio] sink-input.c: module-stream-restore.id = "sink-input-by-media-role:video"
I: [pulseaudio] protocol-native.c: Requested tlength=200,00 ms, minreq=10,00 ms
D: [pulseaudio] protocol-native.c: Adjust latency mode enabled, configuring sink latency to half of overall latency.
D: [pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=27288, base=4, prebuf=0, minreq=1764 maxrewind=0
D: [pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=27288, base=4, prebuf=0, minreq=1764 maxrewind=0
I: [pulseaudio] protocol-native.c: Final latency 200,01 ms = 134,69 ms + 2*10,00 ms + 45,32 ms
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Core1.Stream added for object /org/pulseaudio/core1/playback_stream1
D: [bluetooth] protocol-native.c: Requesting rewind due to end of underrun.
D: [bluetooth] protocol-native.c: Requesting rewind due to end of underrun.
D: [bluetooth] protocol-native.c: Requesting rewind due to end of underrun.
D: [bluetooth] protocol-native.c: Requesting rewind due to end of underrun.
D: [bluetooth] protocol-native.c: Requesting rewind due to end of underrun.
D: [bluetooth] protocol-native.c: Requesting rewind due to end of underrun.
D: [bluetooth] protocol-native.c: Requesting rewind due to end of underrun.
D: [bluetooth] protocol-native.c: Requesting rewind due to end of underrun.
D: [bluetooth] protocol-native.c: Requesting rewind due to end of underrun.
D: [bluetooth] protocol-native.c: Requesting rewind due to end of underrun.
D: [bluetooth] protocol-native.c: Requesting rewind due to end of underrun.
D: [bluetooth] protocol-native.c: Requesting rewind due to end of underrun.
D: [bluetooth] sink-input.c: Requesting rewind due to uncorking
D: [bluetooth] sink-input.c: Have to rewind 16 bytes on implementor.
D: [pulseaudio] module-suspend-on-idle.c: Sink ladspa_output.mbeq_1197.mbeq becomes busy.
E: [bluetooth] memblock.c: Assertion 'b' failed at pulsecore/memblock.c:454, function pa_memblock_acquire(). Aborting.
It happens with different applications connecting to PA. pa-mbeq is active and everytime a stream begins/ends/pauses(corks?), the problem (see above) occurs.
If pa-mbeq connects to alsa it works. If it connects to bluetooth or another pulseaudio-server via Lan it works not.
I tried another laptop, same debian packages (wheezy) same version, same configuration. mbeq => bluetooth (same headset) works.
So the problem occurs on one of my laptops and if mbeq connects to bluetooth or another pulseaudio server..... ?
Thanks for any help!
Jasminhttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/177pulseaudio blocked in kernel2019-09-26T10:59:12ZBugzilla Migration Userpulseaudio blocked in kernel## Submitted by Brian J. Murrell
Assigned to **pul..@..op.org**
**[Link to original bug (#95055)](https://bugs.freedesktop.org/show_bug.cgi?id=95055)**
## Description
I have a pulseaudio process that seems to be blocked in the ker...## Submitted by Brian J. Murrell
Assigned to **pul..@..op.org**
**[Link to original bug (#95055)](https://bugs.freedesktop.org/show_bug.cgi?id=95055)**
## Description
I have a pulseaudio process that seems to be blocked in the kernel:
kernel: sysrq: SysRq : Show Blocked State
kernel: task PC stack pid father
kernel: pulseaudio D ffff8801191a3b98 0 26427 1 0x00000004
kernel: ffff8801191a3b98 ffffffff81c10500 ffff880002788000 ffff8801191a4000
kernel: ffff8804afa0e080 ffff8801191a3bd0 ffff8804afa0e080 ffff88049b053000
kernel: ffff8801191a3bb0 ffffffff81795895 0000000205922d36 ffff8801191a3c60
kernel: Call Trace:
kernel: [`<ffffffff81795895>`] schedule+0x35/0x80
kernel: [`<ffffffff817986b3>`] schedule_timeout+0x123/0x240
kernel: [`<ffffffff8110b7c0>`] ? trace_event_raw_event_tick_stop+0x120/0x120
kernel: [`<ffffffff817993de>`] ? _raw_spin_unlock_irqrestore+0xe/0x10
kernel: [`<ffffffffa0477f05>`] snd_power_wait+0xb5/0x110 [snd]
kernel: [`<ffffffff810cc790>`] ? wake_up_q+0x70/0x70
kernel: [`<ffffffffa0479901>`] snd_ctl_elem_info_user+0x61/0xf0 [snd]
kernel: [`<ffffffffa047c33c>`] snd_ctl_ioctl+0x5ec/0x6c0 [snd]
kernel: [`<ffffffff8133baac>`] ? selinux_file_ioctl+0x10c/0x1c0
kernel: [`<ffffffff8123e7f8>`] do_vfs_ioctl+0x298/0x480
kernel: [`<ffffffff81333323>`] ? security_file_ioctl+0x43/0x60
kernel: [`<ffffffff8123ea59>`] SyS_ioctl+0x79/0x90
kernel: [`<ffffffff8179996e>`] entry_SYSCALL_64_fastpath+0x12/0x71
Not sure how it got that way though.
Any advise or hints?https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/175Automatic profile switching2019-10-13T12:37:56ZBugzilla Migration UserAutomatic profile switching## Submitted by Arun Raghavan `@arun`
Assigned to **pul..@..op.org**
**[Link to original bug (#90792)](https://bugs.freedesktop.org/show_bug.cgi?id=90792)**
## Description
It would be nice to be able to optionally switch profiles ...## Submitted by Arun Raghavan `@arun`
Assigned to **pul..@..op.org**
**[Link to original bug (#90792)](https://bugs.freedesktop.org/show_bug.cgi?id=90792)**
## Description
It would be nice to be able to optionally switch profiles automatically based on the number of channels being played out. This requires a few pieces:
1. Detecting how many channels are actually supported: picking this information from ELD for HDMI would be nice
2. Overriding via settings: since ELD can and (afaik) does lie, we should make this an opt-in feature, and also allow user-provided overrides
3. We can then potentially have policy to switch profiles on a card based on supported channels
(1) and (2) would be generally useful for passthrough as well (detecting what compressed formats are supported by the receiver).https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/170Support multiple devices in plugins2018-07-30T10:01:08ZBugzilla Migration UserSupport multiple devices in plugins## Submitted by Amar Akshat
Assigned to **pul..@..op.org**
**[Link to original bug (#51562)](https://bugs.freedesktop.org/show_bug.cgi?id=51562)**
## Description
Currently we can configure our devices in asoundrc like,
pcm.pulse...## Submitted by Amar Akshat
Assigned to **pul..@..op.org**
**[Link to original bug (#51562)](https://bugs.freedesktop.org/show_bug.cgi?id=51562)**
## Description
Currently we can configure our devices in asoundrc like,
pcm.pulse_i {
type pulse
device alsa_input.pci-0000_00_1b.0.analog-stereo
}
pcm.pulse_o {
type pulse
device alsa_output.pci-0000_00_1b.0.analog-mono
}
I believe it should be possible to either allow multiple device definitions for one virtual interface (like input and output)
or
It should have separate "sink" and "source" parametershttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/167High cpu useage when running Skype2018-07-30T10:00:53ZBugzilla Migration UserHigh cpu useage when running Skype## Submitted by Clemens Eisserer
Assigned to **pul..@..op.org**
**[Link to original bug (#81288)](https://bugs.freedesktop.org/show_bug.cgi?id=81288)**
## Description
Created attachment 102690
sysprof screenshot of system-wide pro...## Submitted by Clemens Eisserer
Assigned to **pul..@..op.org**
**[Link to original bug (#81288)](https://bugs.freedesktop.org/show_bug.cgi?id=81288)**
## Description
Created attachment 102690
sysprof screenshot of system-wide profile run
Running skype on my mother's old Pentium-4 based laptop, I noticed pulseaudio consumes a fairly high share of CPU ressources.
As can be seen in the attached sysprof-screenshot, pulseaudio consumes almost half the cycles skype does. However this was a video-call - so skype was not only de- and encoding audio, but also video. Furthermore, pulseaudio consumed almost 3 times the CPU cycles of Xorg required to push the video pixels to the display.
System:
- Pentium 4, 2.6ghz
- Fedora 20 + latest updates
- pulseaudio-4.0-13.gitf81e3.fc20.i686
I've seen similar behaviour on my modern (sandy bridge core-i7) x86_64 laptop, where pulseaudio consumes as much CPU cycles just playing sound as mplayer needs to decode and display/output video and audio.
~~**Attachment 102690**~~, "sysprof screenshot of system-wide profile run":
![pulseaudio_sysprof](/uploads/20dc451ef92a59de57635801898ad3f2/pulseaudio_sysprof.png)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/164tcp transport always fails: PULSE_SERVER=127.0.0.1 src/pacat, Connection fail...2019-09-21T14:40:29ZBugzilla Migration Usertcp transport always fails: PULSE_SERVER=127.0.0.1 src/pacat, Connection failure: Connection terminated## Submitted by Sergei Trofimovich
Assigned to **pul..@..op.org**
**[Link to original bug (#96873)](https://bugs.freedesktop.org/show_bug.cgi?id=96873)**
## Description
Noticed breakage on real applications like mpv and other medi...## Submitted by Sergei Trofimovich
Assigned to **pul..@..op.org**
**[Link to original bug (#96873)](https://bugs.freedesktop.org/show_bug.cgi?id=96873)**
## Description
Noticed breakage on real applications like mpv and other media apps.
PA used to work but I'm not sure what exactly changed in my environment.
Currently pulseaudio over TCP is broken on the following versions:
7.1, 8.0, 9.0.
How to reproduce:
=================
1. run pulseaudio server on one terminal:
$ pulseaudio -vvv
2. run client with PULSE_SERVER=127.0.0.1 set
dev/git/pulseaudio $ PULSE_SERVER=127.0.0.1 src/pacat
Analysis:
=========
1. strace shows there is a failed sendmsg(cmsg_type=SCM_CREDENTIALS) on TCP socket.
$ PULSE_SERVER=127.0.0.1 strace -f src/pacat
ppoll([{fd=3, events=POLLIN}, {fd=13, events=POLLIN|POLLOUT}, {fd=0, events=POLLIN}, {fd=5, events=POLLIN}], 4, {29, 999755000}, NULL, 8) = 1 ([{fd=13, revents=POLLOUT}], left {29, 999753448})
write(4, "W", 1) = 1
sendmsg(13, {msg_name(0)=NULL, msg_iov(1)=[{"\0\0\1\24\377\377\377\377\0\0\0\0\0\0\0\0\0\0\0\0", 20}], msg_controllen=32, {cmsg_len=28, cmsg_level=SOL_SOCKET, cmsg_type=SCM_CREDENTIALS{pid=5412, uid=1000, gid=100}}, msg_flags=0}, MSG_NOSIGNAL) = -1 EINVAL (Invalid argument)
write(2, "Connection failure: Connection t"..., 42Connection failure: Connection terminated
) = 42
write(4, "W", 1) = 1
write(4, "W", 1) = 1
close(13)
server side sees it as abrupt connection:
I: [pulseaudio] socket-server.c: TCP connection accepted by tcpwrap.
I: [pulseaudio] client.c: Created 8 "Native client (TCP/IP client from 127.0.0.1:46160)"
I: [pulseaudio] client.c: Freed 8 "Native client (TCP/IP client from 127.0.0.1:46160)"
I: [pulseaudio] protocol-native.c: Connection died.
Looking at the code there is a few functions that assume UNIX sockets:
pa_iochannel_write_with_creds()
pa_iochannel_write_with_fds()
I've added a few asserts there to explicitly state the invariant of UNIX socket being handled:
diff --git a/src/pulsecore/iochannel.c b/src/pulsecore/iochannel.c
index e62750b..a85de41 100644
--- a/src/pulsecore/iochannel.c
+++ b/src/pulsecore/iochannel.c
@@ -313,2 +313,3 @@ ssize_t pa_iochannel_write_with_creds(pa_iochannel*io, const void*data, size_t l
pa_assert(io->ofd >= 0);
+ pa_assert(pa_iochannel_creds_supported(io));
@@ -365,2 +366,3 @@ ssize_t pa_iochannel_write_with_fds(pa_iochannel*io, const void*data, size_t l,
pa_assert(nfd <= MAX_ANCIL_DATA_FDS);
+ pa_assert(pa_iochannel_creds_supported(io));
That allows to get exact backtrace how crash happens:
dev/git/pulseaudio $ PULSE_SERVER=127.0.0.1 src/pacat
Assertion 'pa_iochannel_creds_supported(io)' failed at pulsecore/iochannel.c:314, function pa_iochannel_write_with_creds(). Aborting.
Aborted (core dumped)
dev/git/pulseaudio $ gdb src/.libs/pacat core.29381
#0 0x00007f814e4591c8 in __GI_raise (sig=sig@entry=6) at ../sysdeps/unix/sysv/linux/raise.c:54
54 ../sysdeps/unix/sysv/linux/raise.c: No such file or directory.
(gdb) bt
#0 0x00007f814e4591c8 in __GI_raise (sig=sig@entry=6) at ../sysdeps/unix/sysv/linux/raise.c:54
#1 0x00007f814e45a61a in __GI_abort () at abort.c:89
#2 0x00007f815164dc5e in pa_iochannel_write_with_creds (io=0xb08830, data=0xb08aa8, l=20, ucred=0xb0a944)
at pulsecore/iochannel.c:314
#3 0x00007f815166958a in do_write (p=0xb08a60) at pulsecore/pstream.c:777
#4 0x00007f8151666fc1 in do_pstream_read_write (p=0xb08a60) at pulsecore/pstream.c:258
#5 0x00007f8151667358 in io_callback (io=0xb08830, userdata=0xb08a60) at pulsecore/pstream.c:304
#6 0x00007f815164ceff in callback (m=0xb02748, e=0xb08a10, fd=13, f=PA_IO_EVENT_OUTPUT, userdata=0xb08830)
at pulsecore/iochannel.c:158
#7 0x00007f81518d922a in dispatch_pollfds (m=0xb026f0) at pulse/mainloop.c:655
#8 0x00007f81518da011 in pa_mainloop_dispatch (m=0xb026f0) at pulse/mainloop.c:898
#9 0x00007f81518da18e in pa_mainloop_iterate (m=0xb026f0, block=1, retval=0x7fff98276708) at pulse/mainloop.c:929
#10 0x00007f81518da1ee in pa_mainloop_run (m=0xb026f0, retval=0x7fff98276708) at pulse/mainloop.c:944
#11 0x0000000000406e43 in main (argc=1, argv=0x7fff98276a28) at utils/pacat.c:1202https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/163Adjusting volume of paused stream causes audible crackling in another2020-09-11T14:37:44ZBugzilla Migration UserAdjusting volume of paused stream causes audible crackling in another## Submitted by James
Assigned to **pul..@..op.org**
**[Link to original bug (#50113)](https://bugs.freedesktop.org/show_bug.cgi?id=50113)**
## Description
Created attachment 61836
alsa-info (Conexant codec)
[This has been around...## Submitted by James
Assigned to **pul..@..op.org**
**[Link to original bug (#50113)](https://bugs.freedesktop.org/show_bug.cgi?id=50113)**
## Description
Created attachment 61836
alsa-info (Conexant codec)
[This has been around since before March 2010, and was originally reported in RHBZ https://bugzilla.redhat.com/show_bug.cgi?id=576358 .]
Description of problem:
Adjusting volume of paused stream causes audible crackling in another stream
that is currently playing. This seems particularly pronounced if the two
streams have different sample rates.
Seen on Intel HDA with Realtek ALC 883 and Conexant CX20585 codecs, and ATI IXP audio.
Version-Release number of selected component (if applicable):
pulseaudio-1.1-9.fc17.x86_64
How reproducible:
Always.
Steps to Reproduce:
1. Start two streams, A and B.
2. Pause stream A in such a way that it remains open.
3. Adjust the volume of the stream A in Sound Preferences.
Actual results:
Crackling in stream B.
Expected results:
No crackling.
**Attachment 61836**, "alsa-info (Conexant codec)":
[alsa-info.txt.6P2lO0bxT6](/uploads/e5d46264ef201aa1e07118f1cd7d310b/alsa-info.txt.6P2lO0bxT6)