pulseaudio issueshttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues2023-10-08T19:47:49Zhttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/4Feature request: "stream volume modifiers"2023-10-08T19:47:49ZBugzilla Migration UserFeature request: "stream volume modifiers"## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#39556)](https://bugs.freedesktop.org/show_bug.cgi?id=39556)**
## Description
For supporting eg. replay gain and fading, I'd like to se...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#39556)](https://bugs.freedesktop.org/show_bug.cgi?id=39556)**
## Description
For supporting eg. replay gain and fading, I'd like to see a new concept implemented: "stream volume modifiers". They would be like a second stream volume in addition to the normal stream volume, but the volume modifiers wouldn't be treated as something that is set by the user and thus needs to be remembered. A music player could offload replay gain to pulseaudio by adding a volume modifier for that. Pulseaudio could then even use the hw volume for implementing the replay gain, if that's considered useful.
Stream fading could also be implemented using these volume modifiers, until a better (ie. server-side) solution becomes available.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/7It's possible to create audio loops with module-loopback2023-10-08T19:47:50ZBugzilla Migration UserIt's possible to create audio loops with module-loopback## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#61880)](https://bugs.freedesktop.org/show_bug.cgi?id=61880)**
## Description
It's possible to create loops with module-loopback. By lo...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#61880)](https://bugs.freedesktop.org/show_bug.cgi?id=61880)**
## Description
It's possible to create loops with module-loopback. By loops I mean a situation where the same audio circulates within the system forever (and probably gets infinitely amplified as a result).
For example: there are two hardware sinks: hw_sink_H1 and hw_sink_H2. Then there is a filter sink filter_sink_F connected to hw_sink_H1, and a loopback from hw_sink_H2's monitor to filter_sink_F:
+-------------+ +----------+
+-->|filter_sink_F|-->|hw_sink_H1|
| +-------------+ +----------+
|
| +------------------+
| | hw_sink_H2 |
| | - - - - - - - - -|
+---------------------|hw_sink_H2.monitor|
+------------------+
This works fine. Now, filter_sink_F is moved to hw_sink_H2:
+----------+
|hw_sink_H1|
+----------+
+-------------+ +------------------+
+-->|filter_sink_F|-->| hw_sink_H2 |
| +-------------+ | - - - - - - - - -|
+---------------------|hw_sink_H2.monitor|
+------------------+
This configuration is obviously broken. But what could we have done? Neither the sink input nor the source output of module-loopback was moved, so the may_move_to() callbacks were never called. Even if module-loopback could have detected the loop, what should it have done? It could have prevented the move, or it could have disabled itself. Neither options sounds particularly good.
The best solution that I can think of is to get rid of all the filter sinks and sources. There would still be possibilities for loops, for example by using two loopbacks to cross-connect two sinks by using their monitor sources, but I believe those cases are preventable. Regarding the feasibility of getting rid of the filter sinks and sources: DSP filters could be handled by attaching them directly to arbitrary streams and devices, and remapping and combining could be integrated in the core so that separate sinks/sources wouldn't be needed.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/9Make the pa_parse_address() interface less error prone2023-10-08T19:47:50ZBugzilla Migration UserMake the pa_parse_address() interface less error prone## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#89537)](https://bugs.freedesktop.org/show_bug.cgi?id=89537)**
## Description
pa_parse_address() takes a pa_parsed_address struct point...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#89537)](https://bugs.freedesktop.org/show_bug.cgi?id=89537)**
## Description
pa_parse_address() takes a pa_parsed_address struct pointer as a parameter, and fills it with data. The pa_parsed_address.path_or_host field is a dynamically allocated string, and it's very easy to forget to free that string after calling pa_parse_address(). The pa_parse_address() interface should be changed so that it's less easy to leak memory.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/10default.pa has 'load-module module-switch-on-port-available' commented out. ...2023-10-08T19:47:50ZBugzilla Migration Userdefault.pa has 'load-module module-switch-on-port-available' commented out. Output flickering between proper line-out and not-plugged in headphone jack.## Submitted by Mike Lieman
Assigned to **pul..@..op.org**
**[Link to original bug (#97767)](https://bugs.freedesktop.org/show_bug.cgi?id=97767)**
## Description
Sep 11 09:11:27 orion.lieman.net pulseaudio[1758]: [pulseaudio] devi...## Submitted by Mike Lieman
Assigned to **pul..@..op.org**
**[Link to original bug (#97767)](https://bugs.freedesktop.org/show_bug.cgi?id=97767)**
## Description
Sep 11 09:11:27 orion.lieman.net pulseaudio[1758]: [pulseaudio] device-port.c: Setting port analog-output-headphones to status no
Sep 11 09:11:27 orion.lieman.net pulseaudio[1758]: [pulseaudio] device-port.c: Setting port analog-output-lineout to status yes
Sep 11 09:11:27 orion.lieman.net pulseaudio[1758]: [pulseaudio] module-alsa-card.c: Jack 'Front Headphone Jack' is now unplugged
Sep 11 09:11:27 orion.lieman.net pulseaudio[1403]: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
Sep 11 09:11:27 orion.lieman.net pulseaudio[1403]: [pulseaudio] device-port.c: Setting port analog-output-headphones to status no
Sep 11 09:11:27 orion.lieman.net pulseaudio[1403]: [pulseaudio] device-port.c: Setting port analog-output-lineout to status yes
Sep 11 09:11:27 orion.lieman.net pulseaudio[1403]: [pulseaudio] module-alsa-card.c: Jack 'Front Headphone Jack' is now unplugged
Sep 11 09:11:27 orion.lieman.net pulseaudio[1758]: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
Sep 11 09:11:27 orion.lieman.net pulseaudio[1403]: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
Sep 11 09:11:27 orion.lieman.net pulseaudio[1758]: [pulseaudio] device-port.c: Setting port analog-output-lineout to status no
Sep 11 09:11:27 orion.lieman.net pulseaudio[1403]: [pulseaudio] device-port.c: Setting port analog-output-lineout to status no
Sep 11 09:11:27 orion.lieman.net pulseaudio[1758]: [pulseaudio] device-port.c: Setting port analog-output-headphones to status yes
Sep 11 09:11:27 orion.lieman.net pulseaudio[1403]: [pulseaudio] device-port.c: Setting port analog-output-headphones to status yes
Sep 11 09:11:27 orion.lieman.net pulseaudio[1758]: [pulseaudio] module-alsa-card.c: Jack 'Front Headphone Jack' is now plugged in
Sep 11 09:11:27 orion.lieman.net pulseaudio[1403]: [pulseaudio] module-alsa-card.c: Jack 'Front Headphone Jack' is now plugged in
Sep 11 09:11:26 orion.lieman.net pulseaudio[1758]: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
Sep 11 09:11:26 orion.lieman.net pulseaudio[1758]: [pulseaudio] device-port.c: Setting port analog-output-headphones to status no
Sep 11 09:11:26 orion.lieman.net pulseaudio[1403]: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
Sep 11 09:11:26 orion.lieman.net pulseaudio[1758]: [pulseaudio] device-port.c: Setting port analog-output-lineout to status yes
Sep 11 09:11:26 orion.lieman.net pulseaudio[1403]: [pulseaudio] device-port.c: Setting port analog-output-headphones to status no
Sep 11 09:11:26 orion.lieman.net pulseaudio[1403]: [pulseaudio] device-port.c: Setting port analog-output-lineout to status yes
Sep 11 09:11:26 orion.lieman.net pulseaudio[1758]: [pulseaudio] module-alsa-card.c: Jack 'Front Headphone Jack' is now unplugged
Sep 11 09:11:26 orion.lieman.net pulseaudio[1403]: [pulseaudio] module-alsa-card.c: Jack 'Front Headphone Jack' is now unplugged
Sep 11 09:11:26 orion.lieman.net pulseaudio[1403]: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
Sep 11 09:11:26 orion.lieman.net pulseaudio[1403]: [pulseaudio] device-port.c: Setting port analog-output-lineout to status no
Sep 11 09:11:26 orion.lieman.net pulseaudio[1403]: [pulseaudio] device-port.c: Setting port analog-output-headphones to status yes
Sep 11 09:11:26 orion.lieman.net pulseaudio[1403]: [pulseaudio] module-alsa-card.c: Jack 'Front Headphone Jack' is now plugged in
Sep 11 09:11:26 orion.lieman.net pulseaudio[1758]: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
Sep 11 09:11:26 orion.lieman.net pulseaudio[1758]: [pulseaudio] device-port.c: Setting port analog-output-lineout to status no
Sep 11 09:11:26 orion.lieman.net pulseaudio[1758]: [pulseaudio] device-port.c: Setting port analog-output-headphones to status yes
Sep 11 09:11:26 orion.lieman.net pulseaudio[1758]: [pulseaudio] module-alsa-card.c: Jack 'Front Headphone Jack' is now plugged in
Headphones are **NEVER** plugged in.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/14Noise blasting at full volume (usb audio, alesis core 1)2023-10-08T19:47:51ZBugzilla Migration UserNoise blasting at full volume (usb audio, alesis core 1)## Submitted by tro..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#99112)](https://bugs.freedesktop.org/show_bug.cgi?id=99112)**
## Description
Created attachment 128499
pactl list
On youtube (html5 player in...## Submitted by tro..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#99112)](https://bugs.freedesktop.org/show_bug.cgi?id=99112)**
## Description
Created attachment 128499
pactl list
On youtube (html5 player in SeaMonkey), when clicking a new video, I was suddenly blasted with "noise" at a VERY high volume. Muting pulseaudio, pausing the video, lowering PA volume, and resuming made the sound normal again. Then I increased the volume with no problems.
This behaviour could seriously damage hearing.
PA package: 1:4.0-0ubuntu11.1
OS: Ubuntu studio 14.04
Device: Alesis core 1
Kernel: Linux ubuntu-studio 3.13.0-105-lowlatency #152-Ubuntu SMP PREEMPT Fri Dec 2 16:52:00 UTC 2016 x86_64 x86_64 x86_64 GNU/Linux
See also [Bug 98863](https://bugs.freedesktop.org/show_bug.cgi?id=98863).
**Attachment 128499**, "pactl list":
[pactl_list_blast.txt](/uploads/20b0c7a0c7a7df75c60ea68d36e7e690/pactl_list_blast.txt)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/16Alienware14 lower range sound output until pin 0x1a is set for a full deeper...2023-10-08T19:47:51ZBugzilla Migration UserAlienware14 lower range sound output until pin 0x1a is set for a full deeper stereo sound output## Submitted by mohammed imran
Assigned to **pul..@..op.org**
**[Link to original bug (#101931)](https://bugs.freedesktop.org/show_bug.cgi?id=101931)**
## Description
Created attachment 132993
before_pin_set
I have Alienware14 la...## Submitted by mohammed imran
Assigned to **pul..@..op.org**
**[Link to original bug (#101931)](https://bugs.freedesktop.org/show_bug.cgi?id=101931)**
## Description
Created attachment 132993
before_pin_set
I have Alienware14 laptop, it has three jacks, a dedicated Mic jack, Headset and Headphone jacks.
But in sound panel i only see, Headphone and Speakers option and input via inbuilt Mic.
I have to use hdajackretask to set 0x1a pin to get full sound with bass from my laptops in built speakers, but loose headphone option in sound panel.
So can you be of help to resolve this issue for all Alienware laptop users.
Card: HDA Intel PCH
Chip: Realtek ALC3661
On Windows everything works, and when i plug in a headphone/w mic i am asked what i have plugged in but on Linux_ubuntu i don't see as such.
Also, i plug in my headphone i get full proper stereo sound.
Please advise?
Regards.
**Attachment 132993**, "before_pin_set":
[before_pinset_output](/uploads/5467520640463b45066f746fb5902a5d/before_pinset_output)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/18[cleanup] sync_input_volumes_within_thread() has unnecessary code duplication2023-08-10T09:42:15ZBugzilla Migration User[cleanup] sync_input_volumes_within_thread() has unnecessary code duplication## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#54252)](https://bugs.freedesktop.org/show_bug.cgi?id=54252)**
## Description
This is sync_input_volumes_within_thread():
static void ...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#54252)](https://bugs.freedesktop.org/show_bug.cgi?id=54252)**
## Description
This is sync_input_volumes_within_thread():
static void sync_input_volumes_within_thread(pa_sink *s) {
pa_sink_input *i;
void *state = NULL;
pa_sink_assert_ref(s);
pa_sink_assert_io_context(s);
PA_HASHMAP_FOREACH(i, s->thread_info.inputs, state) {
if (pa_cvolume_equal(&i->thread_info.soft_volume, &i->soft_volume))
continue;
i->thread_info.soft_volume = i->soft_volume;
pa_sink_input_request_rewind(i, 0, TRUE, FALSE, FALSE);
}
}
And this is the PA_SINK_INPUT_MESSAGE_SET_SOFT_VOLUME handler:
case PA_SINK_INPUT_MESSAGE_SET_SOFT_VOLUME:
if (!pa_cvolume_equal(&i->thread_info.soft_volume, &i->soft_volume)) {
i->thread_info.soft_volume = i->soft_volume;
pa_sink_input_request_rewind(i, 0, TRUE, FALSE, FALSE);
}
return 0;
Instead of duplicating the code in the SET_SOFT_VOLUME handler, sync_input_volumes_within_thread() should call i->process_msg(PA_SINK_INPUT_MESSAGE_SET_SOFT_VOLUME).https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/21Cookie not created & PA may not start when XDG_CONFIG_HOME is set on Windows2023-08-10T09:42:15ZBugzilla Migration UserCookie not created & PA may not start when XDG_CONFIG_HOME is set on Windows## Submitted by Michael DePaulo
Assigned to **pul..@..op.org**
**[Link to original bug (#75418)](https://bugs.freedesktop.org/show_bug.cgi?id=75418)**
## Description
Version: 4.99.3
OS: Windows XP 32-bit SP3 (joined to a domain)
...## Submitted by Michael DePaulo
Assigned to **pul..@..op.org**
**[Link to original bug (#75418)](https://bugs.freedesktop.org/show_bug.cgi?id=75418)**
## Description
Version: 4.99.3
OS: Windows XP 32-bit SP3 (joined to a domain)
I believe I've observed the same behavior on Pulseaudio 3.0 and 4.0 too.
I also believe I've observed the same behavior on other versions of Windows (such as Win7 64-bit Pro SP1.)
XDG_CONFIG_HOME appears to default to the path %USERPROFILE%\.config . So if I do not set XDG_CONFIG_HOME, the cookie is created at:
C:\Documents and Settings\mike.DEPAULO\.config\pulse
If I set XDG_CONFIG_HOME to C:\test\config, pulseaudio will use an existing cookie at either:
C:\Documents and Settings\mike.DEPAULO\.config\pulse\cookie
C:\test\config\pulse\cookie
However, the problem is that the cookie is not created at either path.
Furthermore, if the cookie cannot be found, pulseaudio.exe fails to start. It produces the following output, then quits:
C:\Program Files\x2goclient\pulseaudio-4.99.3-win32-bug_66962-bug_69712-test>pulseaudio.exe -n -F C:\config.pa
W: [(null)] pulsecore/core-util.c: Secure directory creation not supported on Win32.
W: [(null)] pulsecore/core-util.c: Secure directory creation not supported on Win32.
W: [(null)] pulsecore/core-util.c: Secure directory creation not supported on Win32.
W: [(null)] pulsecore/core.c: failed to allocate shared memory pool. Falling back to a normal memory pool.
W: [(null)] pulsecore/authkey.c: Failed to open cookie file 'C:\Documents and Settings\mike.DEPAULO\.config/pulse/cookie': No such file or directory
W: [(null)] pulsecore/authkey.c: Failed to load authorization key 'C:\Documents and Settings\mike.DEPAULO\.config/pulse/cookie': No error
W: [(null)] pulsecore/authkey.c: Failed to open cookie file 'C:\Documents and Settings\mike.DEPAULO\.pulse-cookie': No such file or directory
W: [(null)] pulsecore/authkey.c: Failed to load authorization key 'C:\Documents and Settings\mike.DEPAULO\.pulse-cookie': No error
W: [(null)] pulsecore/core-util.c: Secure directory creation not supported on Win32.
W: [(null)] pulsecore/authkey.c: Failed to open cookie file 'C:\Documents and Settings\mike.DEPAULO\.config/pulse/cookie': No such file or directory
W: [(null)] pulsecore/authkey.c: Failed to load authorization key 'C:\Documents and Settings\mike.DEPAULO\.config/pulse/cookie': No error
E: [(null)] pulsecore/module.c: Failed to load module "module-native-protocol-tcp" (argument: "port=4713"): initialization failed.
E: [(null)] daemon/main.c: Module load failed.
E: [(null)] daemon/main.c: Failed to initialize daemon.
W: [(null)] pulsecore/core-util.c: Secure directory creation not supported on Win32.
Note that my config.pa file contains the lines:
load-module module-native-protocol-tcp port=4713
load-module module-esound-protocol-tcp port=4714
load-module module-waveout
Note that this bug appears to be separate from [Bug 75006](https://bugs.freedesktop.org/show_bug.cgi?id=75006) - "neither XDG_CONFIG_HOME or PULSE_COOKIE is respected" because:
1. An existing cookie can be used relative to XDG_CONFIG_HOME.
2. When XDG_CONFIG_HOME is set, whether the cookie is found or not, the runtime dir is still created relative to XDG_CONFIG_HOME.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/23Add object name parsing to pa_config_parser2020-07-09T09:06:03ZBugzilla Migration UserAdd object name parsing to pa_config_parser## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#88830)](https://bugs.freedesktop.org/show_bug.cgi?id=88830)**
## Description
Some configuration files support the following pattern:
...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#88830)](https://bugs.freedesktop.org/show_bug.cgi?id=88830)**
## Description
Some configuration files support the following pattern:
[SomeObjectType foo]
somekey = somevalue
[SomeObjectType bar]
somekey = someothervalue
In the above example, "SomeObjectType" identifies a type for an object, and "foo" and "bar" identify the object. Currently pa_config_parser doesn't understand anything about this, however; it just sees that there are two section with names "SomeObjectType foo" and "SomeObjectType bar". This means that parsing the object names is pushed to the code that is using pa_config_parser. The object names are parsed every time a config value assignment is done, which is redundant work, and makes the parsing more complicated, and it also prevents us from printing the line number of the section header when a missing or invalid object name is encountered.
So, pa_config_parser should be extended so that it natively understands the concept of object names in section headers, so that the users of pa_config_parser can easily fetch the object name without having to parse it themselves.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/27Capture device handling on Acer Aspire One 531h2020-03-19T14:17:43ZBugzilla Migration UserCapture device handling on Acer Aspire One 531h## Submitted by Dmitrij D. Czarkoff
Assigned to **pul..@..op.org**
**[Link to original bug (#42643)](https://bugs.freedesktop.org/show_bug.cgi?id=42643)**
## Description
On Acer Aspire One 531h there are two microphones:
* intern...## Submitted by Dmitrij D. Czarkoff
Assigned to **pul..@..op.org**
**[Link to original bug (#42643)](https://bugs.freedesktop.org/show_bug.cgi?id=42643)**
## Description
On Acer Aspire One 531h there are two microphones:
* internal, works;
* mic jack, doesn't work, seems to be hardware fault. May be is aggregated with internal mic, I can't test it.
From now on I refere to internal micorophone as "mic".
Alsa gives three capture controls:
* Mic Boost - probably boosts something, but no audible effect (with my ears);
* Capture - controls the internal mic.
* Capture 1 - controls nothing.
Pulseaudio gives one two-channel control.
With alsa applications everything works as expected - the mic works, records in two-channel mode. Eg., in gnome-sound-recorder.
In pulsaudio the capture records nothing with capture level up on both channels. More interestingly, the actual volume seems to be the difference between the channels' volumes. Eg., if left channel is set to 0 in alsamixer and right channel is set to 100, sound is loud, in combination 0-50 - half of previous.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/29Missing form-factors in sink priorities2019-06-16T17:30:34ZBugzilla Migration UserMissing form-factors in sink priorities## Submitted by Christian Kellner `@gicmo`
Assigned to **pul..@..op.org**
**[Link to original bug (#100579)](https://bugs.freedesktop.org/show_bug.cgi?id=100579)**
## Description
Currently not all form-factors that are defined by ...## Submitted by Christian Kellner `@gicmo`
Assigned to **pul..@..op.org**
**[Link to original bug (#100579)](https://bugs.freedesktop.org/show_bug.cgi?id=100579)**
## Description
Currently not all form-factors that are defined by bluetooth are taken
into accout when priorities are assigned. Missing are:
- hansfree
- portable
- car
- hifi
- phone (might be a source only?)
If they are missing it means they don't get any contribution to the
priority from the form factor and will most likely be rated lower
then anything 'internal'.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/31Dre Beats cause major hang ups2019-02-01T18:30:04ZBugzilla Migration UserDre Beats cause major hang ups## Submitted by John C
Assigned to **pul..@..op.org**
**[Link to original bug (#55401)](https://bugs.freedesktop.org/show_bug.cgi?id=55401)**
## Description
I have an HP Envy 3040nr:
http://www.newegg.com/Product/Product.aspx?Item...## Submitted by John C
Assigned to **pul..@..op.org**
**[Link to original bug (#55401)](https://bugs.freedesktop.org/show_bug.cgi?id=55401)**
## Description
I have an HP Envy 3040nr:
http://www.newegg.com/Product/Product.aspx?Item=N82E16834158219
It has Dre "Beats" speaker. I do not know if they are the cause of the issue, but this computer suffer sever hang ups after log in.
Gnome-shell gdm cannot activate on this computer unless I use a workaround.
This happens with Arch, Ubuntu, Mint, and Fedora.
My workaround is to add "blacklist snd-usb-audio" to /etc/modprobe.d/alsa-base.conf
Without using this workaround, the computer borders on unsuable. The sound applets refuse to even work.
I was unsure under which category to put this.
My OS: Arch Linux x64https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/38Programmable dead time when starting playback on spdif or hdmi2020-01-28T00:09:19ZBugzilla Migration UserProgrammable dead time when starting playback on spdif or hdmi## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#47387)](https://bugs.freedesktop.org/show_bug.cgi?id=47387)**
## Description
Copied from http://lists.freedesktop.org/archives/pulseau...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#47387)](https://bugs.freedesktop.org/show_bug.cgi?id=47387)**
## Description
Copied from http://lists.freedesktop.org/archives/pulseaudio-discuss/2012-January/012647.html
David Hagood writes:
"I'd like to make a suggestion for Pulseaudio - programmable dead time
before uncorking a source when the output is SPDIF or HDMI.
Justification: I have my computer tied to a 5.1 receiver via SPDIF. When I
start playing music in applications such as Audacious, the first half
second of audio is lost as the stereo has to detect the data stream
starting, work out the encoding, and then start decoding. If I could tell
Pulse to insert some "dead air" to allow things to sync up first, then
start playing, this would be eliminated."https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/40Incredibly long internal audio name2018-07-30T09:35:45ZBugzilla Migration UserIncredibly long internal audio name## Submitted by Bastien Nocera
Assigned to **pul..@..op.org**
**[Link to original bug (#103263)](https://bugs.freedesktop.org/show_bug.cgi?id=103263)**
## Description
pulseaudio-11.1-3.fc26.x86_64
On my CherryTrail machine, the i...## Submitted by Bastien Nocera
Assigned to **pul..@..op.org**
**[Link to original bug (#103263)](https://bugs.freedesktop.org/show_bug.cgi?id=103263)**
## Description
pulseaudio-11.1-3.fc26.x86_64
On my CherryTrail machine, the internal audio name is suuuuper long. Is there a way of adding a quirk for this?
Sink #0
State: IDLE
Name: alsa_output.pci-0000_00_02.0-platform-hdmi-lpe-audio.analog-stereo
Description: Atom/Celeron/Pentium Processor x5-E8000/J3xxx/N3xxx Series PCI Configuration Registers Analog Stereo
Driver: module-alsa-card.c
Sample Specification: s16le 2ch 44100Hz
Channel Map: front-left,front-right
Owner Module: 6
Mute: no
Volume: front-left: 4681 / 7% / -68.77 dB, front-right: 4681 / 7% / -68.77 dB
balance 0.00
Base Volume: 65536 / 100% / 0.00 dB
Monitor Source: alsa_output.pci-0000_00_02.0-platform-hdmi-lpe-audio.analog-stereo.monitor
Latency: 17399 usec, configured 100136 usec
Flags: HARDWARE HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY
Properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "Intel HDMI/DP LPE Audio"
alsa.id = "HdmiLpeAudio"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "0"
alsa.card_name = "Intel HDMI/DP LPE Audio"
alsa.long_card_name = "Intel HDMI/DP LPE Audio"
alsa.driver_name = "snd_hdmi_lpe_audio"
device.bus_path = "pci-0000:00:02.0-platform-hdmi-lpe-audio"
sysfs.path = "/devices/pci0000:00/0000:00:02.0/hdmi-lpe-audio/sound/card0"
device.bus = "pci"
device.vendor.id = "8086"
device.vendor.name = "Intel Corporation"
device.product.id = "22b0"
device.product.name = "Atom/Celeron/Pentium Processor x5-E8000/J3xxx/N3xxx Series PCI Configuration Registers"
device.string = "front:0"
device.buffering.buffer_size = "17664"
device.buffering.fragment_size = "4416"
device.access_mode = "mmap"
device.profile.name = "analog-stereo"
device.profile.description = "Analog Stereo"
device.description = "Atom/Celeron/Pentium Processor x5-E8000/J3xxx/N3xxx Series PCI Configuration Registers Analog Stereo"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
Ports:
analog-output: Analog Output (priority: 9900)
Active Port: analog-output
Formats:
pcmhttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/46Add pa_config_parser_log_error()2019-04-26T10:59:04ZBugzilla Migration UserAdd pa_config_parser_log_error()## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#88832)](https://bugs.freedesktop.org/show_bug.cgi?id=88832)**
## Description
It's tedious to manually prefix log messages with the fil...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#88832)](https://bugs.freedesktop.org/show_bug.cgi?id=88832)**
## Description
It's tedious to manually prefix log messages with the file name and line number when logging errors or warnings when parsing configuration files. There should be pa_config_parser_log_error() and pa_config_parser_log_warn() that could be used by users of pa_config_parser to automatically add the file name and line number information to log messages.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/48flat volumes causes sound crackling if applications volume is not at same lev...2018-07-30T09:46:44ZBugzilla Migration Userflat volumes causes sound crackling if applications volume is not at same level of master volume## Submitted by Germano Massullo
Assigned to **pul..@..op.org**
**[Link to original bug (#92031)](https://bugs.freedesktop.org/show_bug.cgi?id=92031)**
## Description
Created attachment 118328
pulseaudio logs
Description of probl...## Submitted by Germano Massullo
Assigned to **pul..@..op.org**
**[Link to original bug (#92031)](https://bugs.freedesktop.org/show_bug.cgi?id=92031)**
## Description
Created attachment 118328
pulseaudio logs
Description of problem:
Let's assume that master audio volume level is at 50%.
Let's assume that Amarok player audio volume level is at 50%, so at the same level of master audio volume.
In this situation if you change the master audio volume level, Amarok player audio volume level will change too, at the same percentage. No problem till here.
Now you set a certain offset between Amarok volume level and master audio volume level, for example by decreasing Amarok by a 20%. Then try to change the master audio volume level, increasing or decreasing it quickly. You will ear audio crackling.
Disabling flat-volumes in /etc/pulse/daemon.conf solves the problem
Version-Release number of selected component (if applicable):
pulseaudio-6.0-8.fc22.x86_64
I attach output of
$ pulseaudio -vvv
retrieved while doing the test
Downstream bugreport: https://bugzilla.redhat.com/show_bug.cgi?id=1264177
**Attachment 118328**, "pulseaudio logs":
[pulseaudio.tar.xz](/uploads/6c2561007af682cb2c73493e5ca6d7ce/pulseaudio.tar.xz)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/49Add keyword to disable channels in profile mappings of multi-channel devices2018-07-30T09:36:35ZBugzilla Migration UserAdd keyword to disable channels in profile mappings of multi-channel devices## Submitted by Renne
Assigned to **pul..@..op.org**
**[Link to original bug (#91470)](https://bugs.freedesktop.org/show_bug.cgi?id=91470)**
## Description
Currently the only way to separate channels of multi-channel devices is mo...## Submitted by Renne
Assigned to **pul..@..op.org**
**[Link to original bug (#91470)](https://bugs.freedesktop.org/show_bug.cgi?id=91470)**
## Description
Currently the only way to separate channels of multi-channel devices is module-remap-{source|sink} which is complex to configure and tends to block the operation of Pulseaudio when devices are missing/not connected. It's not possible to publish device-specific profiles using module-remap-{source|sink}.
In profile-mappings it's possible to disable audio-routing by naming channels of a channel-map "aux[0-15]". But the source/sink is still handled as multi-channel device with e.g. 10 channels. Consider an audio-device with 10 output-channels and you want to use only 2 channels...
A channel-map would look like
channel-map = left, right, aux2, aux3, aux4, aux5, aux6, aux7, aux8, aux9
PAVUControl doesn't list 2 channels, but 10!
Please add a new keyword - like "none" - to disable a channel completely and to decrease the channel count.
Example:
[Mapping Stereo1]
channel-map = left, right, none, none, none, none, none, none, none, none
[Mapping Stereo2]
channel-map = none, none, left, right, none, none, none, none, none, none
[Mapping Stereo3]
channel-map = none, none, none, none, left, right, none, none, none, none
[Mapping Stereo4]
channel-map = none, none, none, none, none, none, left, right, none, none
[Mapping Stereo5]
channel-map = none, none, none, none, none, none, none, none, left, right
would allow to use one multi-channel sink with 10 channels as 5 stereo sinks.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/50Microphone port switch2018-07-30T09:36:38ZBugzilla Migration UserMicrophone port switch## Submitted by Pali Rohár
Assigned to **pul..@..op.org**
**[Link to original bug (#81580)](https://bugs.freedesktop.org/show_bug.cgi?id=81580)**
## Description
I have laptop with 4 pins combo jack (mic input + headphones output) ...## Submitted by Pali Rohár
Assigned to **pul..@..op.org**
**[Link to original bug (#81580)](https://bugs.freedesktop.org/show_bug.cgi?id=81580)**
## Description
I have laptop with 4 pins combo jack (mic input + headphones output) and when I plug (compatible) headphones then PA automatically switch output port from (internal) Speakers to Heaphones. But it does not switch input port from Internal Mic to (Jack) Mic automatically. If PA already switching audio output based on jack detection it should also automatically switch audio input when jack microphone is available. Reproducable with PA 3.0 and 5.0. With both versions only audio output is automatically switched bases on jack detection.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/53No playback2019-10-28T16:38:45ZBugzilla Migration UserNo playback## Submitted by Tim Koopman
Assigned to **pul..@..op.org**
**[Link to original bug (#50219)](https://bugs.freedesktop.org/show_bug.cgi?id=50219)**
## Description
Created attachment 61965
Debug output
After updating to pulseaudio ...## Submitted by Tim Koopman
Assigned to **pul..@..op.org**
**[Link to original bug (#50219)](https://bugs.freedesktop.org/show_bug.cgi?id=50219)**
## Description
Created attachment 61965
Debug output
After updating to pulseaudio 2.0, it refuses to play any audio. Clients can connect but nothing happens after that (playback does not advance). The deamon can be coaxed to output sound, but I have not found a reliable way to do so: sometimes changing the output device will fix it, sometimes restarting the daemon will help. After it starts working, it may suddenly stop working again as well (when another connection is made?).
Playing directly trough ALSA works fine.
I've attached the output of pulseaudio that appears when I try to play some music and it fails.
**Attachment 61965**, "Debug output":
[pulseaudio.txt](/uploads/8e35f320e19e648bc1868b895c12c6f2/pulseaudio.txt)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/54PulseAudio fix for sparc architecture2018-07-30T10:39:40ZBugzilla Migration UserPulseAudio fix for sparc architecture## Submitted by Brian Cameron `@bacameron`
Assigned to **pul..@..op.org**
**[Link to original bug (#50268)](https://bugs.freedesktop.org/show_bug.cgi?id=50268)**
## Description
Created attachment 62002
patch fixing issue
PulseAud...## Submitted by Brian Cameron `@bacameron`
Assigned to **pul..@..op.org**
**[Link to original bug (#50268)](https://bugs.freedesktop.org/show_bug.cgi?id=50268)**
## Description
Created attachment 62002
patch fixing issue
PulseAudio does not build properly on sparc because WORDS_BIGENDIAN does not get set if _BIG_ENDIAN is set. The attached patch fixes the ENDIAN so it works properly on sparc.
**Attachment 62002**, "patch fixing issue":
[pulseaudio-10-endian.diff](/uploads/818f6b416856175a4e196896a14a2f1e/pulseaudio-10-endian.diff)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/55resampler leftover buffer is not taken into account when rewinding2018-07-30T09:37:30ZBugzilla Migration Userresampler leftover buffer is not taken into account when rewinding## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#53911)](https://bugs.freedesktop.org/show_bug.cgi?id=53911)**
## Description
The resampler leftover buffer can contain data from a sin...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#53911)](https://bugs.freedesktop.org/show_bug.cgi?id=53911)**
## Description
The resampler leftover buffer can contain data from a sink input, data which has been popped from the sink input implementor but not yet pushed to the render_memblockq. That data is not currently taken into account when rewinding. When doing a rewind that affects the resampler, the leftover data is discarded, and that is not compensated in any way, which means that the data is completely lost and there's a skip in the audio.
In practice the leftover buffer is rarely used and even if it is used, it will contain a minimal amount of data, so the user-visible effect of this bug is minor.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/56Offers left/right channels for device with a single speaker2018-07-30T09:37:32ZBugzilla Migration UserOffers left/right channels for device with a single speaker## Submitted by Bastien Nocera
Assigned to **pul..@..op.org**
**[Link to original bug (#103941)](https://bugs.freedesktop.org/show_bug.cgi?id=103941)**
## Description
Using the "Iqua miniUFO" in a2dp mode, only the left speaker ca...## Submitted by Bastien Nocera
Assigned to **pul..@..op.org**
**[Link to original bug (#103941)](https://bugs.freedesktop.org/show_bug.cgi?id=103941)**
## Description
Using the "Iqua miniUFO" in a2dp mode, only the left speaker can play anything back in GNOME's Sound Settings. Playing back audio on the right channel doesn't play anything.
Bizarrely, pavucontrol only sees the "Headset" port, GNOME's Sound Settings can see both.
Is there a way to quirk the output to say that both channels need to be mixed into the left one for this device in a2dp mode?
Card #6
Name: bluez_card.00_13_04_83_5F_7A
Driver: module-bluez5-device.c
Owner Module: 27
Properties:
device.description = "Iqua miniUFO"
device.string = "00:13:04:83:5F:7A"
device.api = "bluez"
device.class = "sound"
device.bus = "bluetooth"
device.form_factor = "headset"
bluez.path = "/org/bluez/hci0/dev_00_13_04_83_5F_7A"
bluez.class = "0x240404"
bluez.alias = "Iqua miniUFO"
device.icon_name = "audio-headset-bluetooth"
device.intended_roles = "phone"
Profiles:
headset_head_unit: Headset Head Unit (HSP/HFP) (sinks: 1, sources: 1, priority: 20, available: yes)
a2dp_sink: High Fidelity Playback (A2DP Sink) (sinks: 1, sources: 0, priority: 10, available: yes)
off: Off (sinks: 0, sources: 0, priority: 0, available: yes)
Active Profile: a2dp_sink
Ports:
headset-output: Headset (priority: 0, latency offset: 0 usec)
Part of profile(s): headset_head_unit, a2dp_sink
headset-input: Headset (priority: 0, latency offset: 0 usec)
Part of profile(s): headset_head_unithttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/57pulseaudio crashed with SIGABRT in transport_speaker_gain_changed_cb()2018-07-30T09:37:36ZBugzilla Migration Userpulseaudio crashed with SIGABRT in transport_speaker_gain_changed_cb()## Submitted by Cristian Aravena
Assigned to **pul..@..op.org**
**[Link to original bug (#89093)](https://bugs.freedesktop.org/show_bug.cgi?id=89093)**
## Description
Open bug in launchpad.net:
https://bugs.launchpad.net/ubuntu/+s...## Submitted by Cristian Aravena
Assigned to **pul..@..op.org**
**[Link to original bug (#89093)](https://bugs.freedesktop.org/show_bug.cgi?id=89093)**
## Description
Open bug in launchpad.net:
https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/1421002
"Change Bluethoo."
```
#0 0x00007f69a77c6e37 in __GI_raise (sig=sig@entry=6) at ../nptl/sysdeps/unix/sysv/linux/raise.c:56
resultvar = 0
pid = 1793
selftid = 1793
#1 0x00007f69a77c8528 in __GI_abort () at abort.c:89
save_stage = 2
act = {__sigaction_handler = {sa_handler = 0x10c979e, sa_sigaction = 0x10c979e}, sa_mask = {__val = {140091773812220, 107491707232, 140734560817648, 140734560817640, 1, 206158430316, 140088948293633, 140733193388038, 17602462, 17602457, 57259365588238080, 16853379, 17044288, 16900064, 140734560818000, 17043876}}, sa_flags = -1753329825, sa_restorer = 0x7f69977ead80}
sigs = {__val = {32, 0 <repeats 15 times>}}
#2 0x00007f69977e4f68 in transport_speaker_gain_changed_cb (y=<optimized out>, t=<optimized out>, u=<optimized out>) at modules/bluetooth/module-bluetooth-device.c:1576
No locals.
#3 0x00007f69a8fbdffc in pa_hook_fire (hook=0x102f2b0, data=0xffae70) at pulsecore/hook-list.c:106
slot = 0x101bb00
next = <optimized out>
result = PA_HOOK_OK
__func__ = "pa_hook_fire"
__PRETTY_FUNCTION__ = "pa_hook_fire"
#4 0x00007f69a17a2da6 in parse_audio_property (d=0x101dfe0, interface=0x1041340 "org.bluez.Headset", i=0x6, is_property_change=255) at modules/bluetooth/bluetooth-util.c:628
value = 15
transport = 0xffae70
variant_i = {dummy1 = 0x10ae140, dummy2 = 0x7f6900600000, dummy3 = 108, dummy4 = 0, dummy5 = 17490320, dummy6 = 0, dummy7 = 17, dummy8 = 32767, dummy9 = 17490320, dummy10 = 0, dummy11 = 20, pad1 = 32767, pad2 = -1467155648, pad3 = 0x7fff518154b0}
p = PROFILE_HSP
tstamp_now = 0
tstamp_prev = 55407133202
m = 0x0
__func__ = "parse_audio_property"
__PRETTY_FUNCTION__ = "parse_audio_property"
#5 0x00007f69a17a66c3 in filter_cb (bus=0x701, m=0x10ae140, userdata=0x101dfe0) at modules/bluetooth/bluetooth-util.c:1023
arg_i = {dummy1 = 0x10ae140, dummy2 = 0x7f6900600000, dummy3 = 108, dummy4 = 0, dummy5 = 17490248, dummy6 = 0, dummy7 = 134, dummy8 = 32767, dummy9 = 17490320, dummy10 = 0, dummy11 = 16, pad1 = 32767, pad2 = -1467155136, pad3 = 0x7fff518154b0}
old_any_connected = true
d = 0x101dfe0
err = {name = 0x0, message = 0x0, dummy1 = 1, dummy2 = 0, dummy3 = 0, dummy4 = 0, dummy5 = 0, padding1 = 0x50}
__func__ = "filter_cb"
__PRETTY_FUNCTION__ = "filter_cb"
#6 0x00007f69a8699161 in dbus_connection_dispatch () from /tmp/apport_sandbox_M6XAT9/lib/x86_64-linux-gnu/libdbus-1.so.3
No symbol table info available.
#7 0x00007f69a8d71114 in dispatch_cb (ea=0xfd2168, ev=0x1046d20, userdata=<optimized out>) at pulsecore/dbus-util.c:55
conn = <optimized out>
#8 0x00007f69a8afedca in dispatch_defer (m=0xfd2110) at pulse/mainloop.c:682
e = 0x1046d20
r = 0
#9 pa_mainloop_dispatch (m=m@entry=0xfd2110) at pulse/mainloop.c:895
dispatched = 0
__func__ = "pa_mainloop_dispatch"
__PRETTY_FUNCTION__ = "pa_mainloop_dispatch"
#10 0x00007f69a8afefdc in pa_mainloop_iterate (m=0xfd2110, block=<optimized out>, retval=0x7fff51815658) at pulse/mainloop.c:935
r = 0
#11 0x00007f69a8aff080 in pa_mainloop_run (m=0xfd2110, retval=0x7fff51815658) at pulse/mainloop.c:950
r = <optimized out>
#12 0x0000000000407058 in main (argc=16589072, argv=0xff4a00) at daemon/main.c:1150
c = 0xfd6440
buf = 0xfdba80
conf = 0xfca090
mainloop = 0xfd2110
configured_address = 0xfd2110 "\220L\377"
retval = 0
d = 3
ltdl_init = false
passed_fd = 16556176
e = 0x0
daemon_pipe = {-1, -1}
daemon_pipe2 = {-1, -1}
pid_monitor = 0xfcfab0
server_lookup = 0xfdba80
server_bus = 0xff4a00
__func__ = "main"
__PRETTY_FUNCTION__ = "main"
```https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/62severe underruns with usb audio, works with pa 3.0 - buffer setup seems wrong2018-07-30T10:24:55ZBugzilla Migration Usersevere underruns with usb audio, works with pa 3.0 - buffer setup seems wrong## Submitted by David Mansfield
Assigned to **pul..@..op.org**
**[Link to original bug (#86262)](https://bugs.freedesktop.org/show_bug.cgi?id=86262)**
## Description
Created attachment 109436
log of running with PA 3.0 version fro...## Submitted by David Mansfield
Assigned to **pul..@..op.org**
**[Link to original bug (#86262)](https://bugs.freedesktop.org/show_bug.cgi?id=86262)**
## Description
Created attachment 109436
log of running with PA 3.0 version from Centos 7
Description of problem:
Constant and severe audio underruns when using Jitsi (voip softphone). When using the pulseaudio 3.0 version from Centos 7 (rhel) audio is perfectly smooth.
Version-Release number of selected component (if applicable):
pulseaudio-5.0-7.fc20.x86_64
How reproducible:
100%
Steps to Reproduce:
1. install jitsi, set up a sip account, use USB audio headset (e.g. plantronics)
2. make a call
3. audio (playback) is choppy and jitsi reports "discarded packets"
Actual results:
choppy audio, tons of underruns reported by PA
Expected results:
smooth audio
Additional info:
Installing the pulseaudio rpms from centos 7 "cures" the problem.
I have run PA in debug mode and captured the log from the startup of PA through hanging up the call both with the 3.0 PA version (current C7) and 5.0 PA version (current F20).
I'll attach the logs.
Diffing the two logs (old (-) = 3.0, new (+) = 5.0), it looks like the significant difference in tho logs has to do with buffering setup (as expected):
I: [pulseaudio] card.c: Created 1 "alsa_card.usb-Plantronics_Plantronics_.Audio_648_USB-00-USB"
+I: [pulseaudio] module-card-restore.c: Storing profile and port latency offsets for card alsa_card.usb-Plantronics_Plantronics_.Audio_648_USB-00-USB.
+D: [pulseaudio] module-alsa-card.c: Found 0 jacks.
D: [pulseaudio] reserve-wrap.c: Successfully create reservation lock monitor for device 'Audio3'
D: [pulseaudio] alsa-util.c: Trying front:3 with SND_PCM_NO_AUTO_FORMAT ...
D: [pulseaudio] alsa-util.c: Managed to open front:3
-I: [pulseaudio] alsa-util.c: cannot disable ALSA period wakeups
+I: [pulseaudio] alsa-util.c: Disabling tsched mode since BATCH flag is set
D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 5944 ms
-D: [pulseaudio] alsa-util.c: Set buffer size first (to 88200 samples), period size second (to 88200 samples).
-I: [pulseaudio] alsa-util.c: ALSA period wakeups were not disabled
+D: [pulseaudio] alsa-util.c: Set buffer size first (to 4408 samples), period size second (to 1102 samples).
I: [pulseaudio] alsa-sink.c: Successfully opened device front:3.
I: [pulseaudio] alsa-sink.c: Selected mapping 'Analog Stereo' (analog-stereo).
+I: [pulseaudio] alsa-sink.c: Cannot enable timer-based scheduling, falling back to sound IRQ scheduling.
I: [pulseaudio] alsa-sink.c: Successfully enabled mmap() mode.
-I: [pulseaudio] alsa-sink.c: Successfully enabled timer-based scheduling mode.
I: [pulseaudio] (alsa-lib)control.c: Invalid CTL front:3
I: [pulseaudio] alsa-util.c: Unable to attach to mixer front:3: No such file or directory
I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:3'
@@ -1055,9 +1173,9 @@
I: [pulseaudio] sink.c: device.product.name = "Plantronics .Audio 648 USB"
I: [pulseaudio] sink.c: device.serial = "Plantronics_Plantronics_.Audio_648_USB"
I: [pulseaudio] sink.c: device.string = "front:3"
-I: [pulseaudio] sink.c: device.buffering.buffer_size = "352800"
-I: [pulseaudio] sink.c: device.buffering.fragment_size = "176400"
-I: [pulseaudio] sink.c: device.access_mode = "mmap+timer"
+I: [pulseaudio] sink.c: device.buffering.buffer_size = "17632"
+I: [pulseaudio] sink.c: device.buffering.fragment_size = "4408"
+I: [pulseaudio] sink.c: device.access_mode = "mmap"
I: [pulseaudio] sink.c: device.profile.name = "analog-stereo"
I: [pulseaudio] sink.c: device.profile.description = "Analog Stereo"
I: [pulseaudio] sink.c: device.description = "Plantronics .Audio 648 USB Analog Stereo"
@@ -1085,16 +1203,16 @@
I: [pulseaudio] source.c: device.string = "3"
I: [pulseaudio] source.c: module-udev-detect.discovered = "1"
I: [pulseaudio] source.c: device.icon_name = "audio-card-usb"
-I: [pulseaudio] alsa-sink.c: Using 2.0 fragments of size 176400 bytes (1000.00ms), buffer size is 352800 bytes (2000.00ms)
-I: [pulseaudio] alsa-sink.c: Time scheduling watermark is 20.00ms
+I: [pulseaudio] alsa-sink.c: Using 4.0 fragments of size 4408 bytes (24.99ms), buffer size is 17632 bytes (99.95ms)
D: [pulseaudio] alsa-sink.c: hwbuf_unused=0
-D: [pulseaudio] alsa-sink.c: setting avail_min=87319
+D: [pulseaudio] alsa-sink.c: setting avail_min=1
Also, theres this difference in realtime prio:
-D: [alsa-sink] alsa-sink.c: Thread starting up
-I: [alsa-sink] core-util.c: Failed to acquire real-time scheduling: Permission denied
-I: [alsa-sink] alsa-sink.c: Starting playback.
-D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half.
-D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half.
+D: [pulseaudio] alsa-sink.c: Read hardware volume: front-left: 34131 / 52% / -17.00 dB, front-right: 34131 / 52% / -17.00 dB
+D: [alsa-sink-USB Audio] alsa-sink.c: Thread starting up
+D: [alsa-sink-USB Audio] core-util.c: RealtimeKit worked.
+I: [alsa-sink-USB Audio] core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5.
+I: [alsa-sink-USB Audio] alsa-sink.c: Starting playback.
**Attachment 109436**, "log of running with PA 3.0 version from Centos 7":
[palog.pa-3.0.txt](/uploads/424578b5c202ecf30b445d18ff44aff4/palog.pa-3.0.txt)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/65protocol-esound, protocol-simple: Rewind after underrun can be too large2018-07-30T09:38:13ZBugzilla Migration Userprotocol-esound, protocol-simple: Rewind after underrun can be too large## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#54251)](https://bugs.freedesktop.org/show_bug.cgi?id=54251)**
## Description
Rewinding after an underrun should be done like this:
pa...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#54251)](https://bugs.freedesktop.org/show_bug.cgi?id=54251)**
## Description
Rewinding after an underrun should be done like this:
pa_sink_input_request_rewind(s->sink_input, (size_t) (s->sink_input->thread_info.underrun_for == (uint64_t) -1 ? 0 : s->sink_input->thread_info.underrun_for), FALSE, TRUE, FALSE);
(The example is from protocol-native.c.)
That ensures that the received data won't overwrite valid data in case the underrun was so short that the sink still has data left from time before the underrun started.
protocol-esound.c currently does this:
pa_sink_input_request_rewind(c->sink_input, 0, FALSE, TRUE, FALSE);
Giving zero as the rewind amount means that a full rewind will be done, regardless of whether valid data might get overwritten. If valid data gets overwritten, there will be a skip in the audio.
protocol-simple.c has the same bug.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/68Digital and analog port for the same card2020-11-25T14:28:02ZBugzilla Migration UserDigital and analog port for the same card## Submitted by Laurent Bigonville
Assigned to **pul..@..op.org**
**[Link to original bug (#100555)](https://bugs.freedesktop.org/show_bug.cgi?id=100555)**
## Description
Created attachment 130668
pactl list
Hi,
I initially open...## Submitted by Laurent Bigonville
Assigned to **pul..@..op.org**
**[Link to original bug (#100555)](https://bugs.freedesktop.org/show_bug.cgi?id=100555)**
## Description
Created attachment 130668
pactl list
Hi,
I initially opened a bug in the GNOME bugzilla, but then I've been told to reopen it here.
https://bugzilla.gnome.org/show_bug.cgi?id=770909
Original report:
Hi,
In gnome-control-center, I see my Logitech G930 wireless headset twice, once with the name "Digital output (S/PDIF) G930" and once with "Analogic output G930"
I guess that should only be one entry?
====
Hadess later replied:
It's not a bug, the card has 2 ports, and we unroll the cards and ports in this list, to avoid having 2 drop-downs per hardware, and make it easier to switch between, for example, an internal speaker and the jack.
You could argue that switching between the analogue and digital outputs should be automatic, and I would certainly agree that this would clean up our interface, but that's a problem that can only be solved in PulseAudio and layers underneath it.
Please file a bug against PulseAudio mentioning the above, that's the best way to drain that swamp.
====
There is the same problem for my internal sound card
**Attachment 130668**, "pactl list":
[pactl_list.txt](/uploads/84b3b114fa72aa410fdd66f02e781b70/pactl_list.txt)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/71pulse-cli-syntax man page should describe suspend and kill in more detail2018-07-30T09:38:48ZBugzilla Migration Userpulse-cli-syntax man page should describe suspend and kill in more detail## Submitted by Frederick Eaton
Assigned to **pul..@..op.org**
**[Link to original bug (#105907)](https://bugs.freedesktop.org/show_bug.cgi?id=105907)**
## Description
I've been reading pulse-cli-syntax(5) and I don't understand w...## Submitted by Frederick Eaton
Assigned to **pul..@..op.org**
**[Link to original bug (#105907)](https://bugs.freedesktop.org/show_bug.cgi?id=105907)**
## Description
I've been reading pulse-cli-syntax(5) and I don't understand what is the difference between 'suspend-sink' and 'kill-sink-input' and how they are used. Is it true that I use the former when I want to free a hardware device so I can open it from another sound server, and that I would use the latter command to stop a client from playing sound without killing its process? Will Pulseaudio just ignore any further samples that are sent by a client or sink input which has been "killed", or will the client get an error of some kind?
Is there another manual page or document which describes these in more detail?
What is the meaning of the argument to 'suspend-sink' and 'suspend' which is boolean?
What is the difference between removing a client or sink, and removing it "forcibly" (as the manual page says is done by the "kill-" commands)?
I think all of this should be documented in a place which is easy for new users to locate, such as the pulse-cli-syntax(5) man page or some document which is referenced by it. As it is, I find a lot of examples on the web invoking things like 'pacmd suspend true' and each time I read this stuff I go to your man page and find it frustratingly incomplete.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/73Possible memory leak in resampler.c2018-07-30T09:38:55ZBugzilla Migration UserPossible memory leak in resampler.c## Submitted by Barun Kumar Singh
Assigned to **pul..@..op.org**
**[Link to original bug (#95377)](https://bugs.freedesktop.org/show_bug.cgi?id=95377)**
## Description## Submitted by Barun Kumar Singh
Assigned to **pul..@..op.org**
**[Link to original bug (#95377)](https://bugs.freedesktop.org/show_bug.cgi?id=95377)**
## Descriptionhttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/78pa_database_next() is inefficient for the "simple" pa_database implementation2018-07-30T09:39:23ZBugzilla Migration Userpa_database_next() is inefficient for the "simple" pa_database implementation## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#44532)](https://bugs.freedesktop.org/show_bug.cgi?id=44532)**
## Description
If I want to iterate through a database with pa_database_...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#44532)](https://bugs.freedesktop.org/show_bug.cgi?id=44532)**
## Description
If I want to iterate through a database with pa_database_next(), it will have O(n*n) time complexity with the "simple" pa_database implementation, because pa_database_next() will every time search for the next item by iterating the backing pa_hashmap from the beginning.
I'm not sure what would be the best way to fix this. Maybe the pa_database iteration interface could be changed to match pa_hashmap_iterate()?
In normal use cases I don't see this inefficiency causing any trouble, because the databases don't tend to be very big. It's just annoying to have this kind of stuff in the code...https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/83protocol-native: I get endless underruns when I playback when alsa buffer siz...2018-07-30T09:39:53ZBugzilla Migration Userprotocol-native: I get endless underruns when I playback when alsa buffer size is small## Submitted by Alban Browaeys
Assigned to **pul..@..op.org**
**[Link to original bug (#84878)](https://bugs.freedesktop.org/show_bug.cgi?id=84878)**
## Description
On my exynos4 arm board (odroid u2) I cannot playback with most "...## Submitted by Alban Browaeys
Assigned to **pul..@..op.org**
**[Link to original bug (#84878)](https://bugs.freedesktop.org/show_bug.cgi?id=84878)**
## Description
On my exynos4 arm board (odroid u2) I cannot playback with most "alsa/oss only" players (moc, alsaplayer, aplay when I set the buffer_size to a small value <= 5000).
Mind if I disable pulseaudio the playback is fine.
At startup underruns occurs and pile up fast. Mind with the workaround told below I also get underruns but they are recovered quite fast .
I found this to be related to the support or no support of dma residue reporting to alsa. It turns out that pl330 DMAC used on this board has not support for residue. Only granularity descriptor.
I could confirm as I have a set of patch that had this support and the issue vanish if I enable residue support (via granularity burst).
ie at :
https://github.com/prahal/linux/commit/6f6cfc5cc1f795526adb2b730154624639845117
and the three patches before this one.
Logs when granularity is BURST or SEGMENT but not DESCRIPTOR (ie odroid u2 with above patch applied, and all other platform I could get my hand on: two amd64 boxes -one intel the other amd64 - and an x86_32 all behaves the same: minreq is equal prebuf , except a few where minreq is below prebuf ).
PS: PULSE_LATENCY_MSEC=60 workaround the issue, though it looks
like it only does so as it forces the prebuf to tlength (which is always
higher than minreq).
Logs:
- vanilla pulseaudio: broken audio output:
E: [lt-pulseaudio] protocol-native.c: Client requested: maxlength=4194304 bytes tlength=32768 bytes minreq=2048 bytes prebuf=2048 bytes
E: [lt-pulseaudio] protocol-native.c: Client requested: maxlength=23777 ms tlength=185 ms minreq=11 ms prebuf=11 ms
I: [lt-pulseaudio] protocol-native.c: Requested tlength=185,76 ms, minreq=11,61 ms
D: [lt-pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
D: [lt-pulseaudio] protocol-native.c: Requested latency=11,61 ms, Received latency=99,95 ms
E: [lt-pulseaudio] protocol-native.c: Client accepted: maxlength=23777 ms tlength=299 ms minreq=99 ms prebuf=11 ms
D: [lt-pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=52896, base=4, prebuf=2048, minreq=17628 maxrewind=0
D: [lt-pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=52896, base=4, prebuf=2048, minreq=17628 maxrewind=0
- modified pulseaudio: correct audio out.
E: [lt-pulseaudio] protocol-native.c: Client requested: maxlength=4194304 bytes tlength=32768 bytes minreq=2048 bytes prebuf=2048 bytes
E: [lt-pulseaudio] protocol-native.c: Client requested: maxlength=23777 ms tlength=185 ms minreq=11 ms prebuf=11 ms
I: [lt-pulseaudio] protocol-native.c: Requested tlength=185,76 ms, minreq=11,61 ms
D: [lt-pulseaudio] protocol-native.c: Early requests mode enabled, configuring sink latency to minreq.
D: [lt-pulseaudio] protocol-native.c: Requested latency=11,61 ms, Received latency=99,95 ms
E: [lt-pulseaudio] protocol-native.c: Client accepted: maxlength=23777 ms tlength=299 ms minreq=99 ms prebuf=99 ms
D: [lt-pulseaudio] memblockq.c: memblockq requested: maxlength=4194304, tlength=52896, base=4, prebuf=17628, minreq=17628 maxrewind=0
D: [lt-pulseaudio] memblockq.c: memblockq sanitized: maxlength=4194304, tlength=52896, base=4, prebuf=17628, minreq=17628 maxrewind=0
I modified src/pulsecore/protocol-native.c fix_playback_buffer_attr ,I now set
prebuf to at least minreq (at the end of the function for now).
To sum up there is a way to workaround the issue in the kernel (in avoid no residue , ie granularity DESCRIPTOR).
Mind the client always request a prebuf equal or greater than minreq. Only when heuristics in fix__playback_buffer_attr increase minreq they do not change prebuf).https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/88pulseaudio 5.0 causes mplayer to freeze when pausing or seeking2018-07-30T09:40:34ZBugzilla Migration Userpulseaudio 5.0 causes mplayer to freeze when pausing or seeking## Submitted by lam..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#82166)](https://bugs.freedesktop.org/show_bug.cgi?id=82166)**
## Description
When using mplayer -ao pulse video.mp4, the video is reproduced c...## Submitted by lam..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#82166)](https://bugs.freedesktop.org/show_bug.cgi?id=82166)**
## Description
When using mplayer -ao pulse video.mp4, the video is reproduced correctly, but if I pause and unpause, or seek, mplayer will freeze. The only way to recover from this is killing mplayer.
There is no such problem when using alsa directly.
My system consists of:
pulseaudio 5.0
linux 3.15.0
alsa-lib 1.0.27
The sound card is HDA Nvidia.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/90Discontinuity in the interpolated delay after corking, flushing and uncorking.2018-07-30T09:40:43ZBugzilla Migration UserDiscontinuity in the interpolated delay after corking, flushing and uncorking.## Submitted by Niklas Haas
Assigned to **pul..@..op.org**
**[Link to original bug (#97799)](https://bugs.freedesktop.org/show_bug.cgi?id=97799)**
## Description
Created attachment 126489
regular playback
After a long session of ...## Submitted by Niklas Haas
Assigned to **pul..@..op.org**
**[Link to original bug (#97799)](https://bugs.freedesktop.org/show_bug.cgi?id=97799)**
## Description
Created attachment 126489
regular playback
After a long session of debugging in #pulseaudio, I got no further than this result, so I'm submitting it here for reference and to get more opinions:
Basically, I'm running into the issue where the reported delay (as measured by pa_stream_get_latency before and after writing data) suffers from a relatively big discontinuity (about 50-100ms) when seeking in mpv. Seeking in mpv is implemented by resetting the audio device, which in mpv's terms means:
1. corking the audio stream (pa_stream_cork true)
2. flushing the audio device (pa_stream_flush)
3. uncorking the audio stream (pa_stream_cork false)
4. continuing playback
I have attached a number of plots of mpv's internal timing data to help illustrate the problem. The first attachment (“regular playback”) establishes a baseline reading (10 seconds of uninterrupted playback). Some notes:
1. The green line (ao-delay) is the reported latency as measured directly by pa_stream_get_latency.
2. The blue line (ao-dev) is the difference between the latency and where mpv thinks it should be (assuming the audio device plays at a perfectly even rate). Ideally, this line should be exactly 0, and more importantly, this line should be as stable as possible (no jumps, no jittering and no spikes), because it is compared against the video stream's timing to detect A/V desynchronization.
3. While they're not as interesting for this test, the spikes at the top indicate events (rather than values). The curve going up represents an event starting, and the curve going down indicates an event stopping. It's somewhat harder to read the legend, so I'll repeat it here: green = ao-fill (inside “pa_stream_write”), blue = audio (“decoding more audio”), black = audio wait (“audio thread sleeping”), yellow = sleep (“playback thread sleeping”)
4. The latency values are measured with PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_NOT_MONOTONIC, and the target latency (tlength) is 2000ms. Decreasing the tlength to a lower value decreases the magnitude of the discontinuity.
The second attachment (“seeking”) indicates what happens when triggering a seek in mpv. I have triggered two seeks, one at about 3.3s and one at about 6.6s. We can observe:
1. The reported latency (ao-delay) suffers from an upwards discontinuity of about 50ms - going up from the baseline latency of about 1050ms to a value near 1100ms, and then gradually decreasing back down to 1000ms over the course of the next 1-2 seconds.
2. Similarly, the “apparent” position of mpv suffers from a downwards discontinuity from 0ms to about -35ms, which increases to +20ms over the course of the next 1-2 seconds.
The third attachment (“seeking zoom”) is a detail view of the discontinuity as it happens, in case it helps.
In case you're wondering why these discontinues affect playback in mpv, I've included a fourth attachment demonstrating the same effects during regular playback. I've included two graphs - one with a smaller tlength (~200ms iirc) and one with a larger tlength (2000ms). They're a bit more confusing because they also include video stats, but basically the effect that happens is as follows:
1. The user triggers a seek
2. The apparent audio position (ao-dev, green line at the bottom) shoots downwards (as observed earlier)
3. The measured A-V difference (avdiff, black line) spikes downwards as a result (by about 100ms in the tlength=2000ms case)
4. This exceeds mpv's thresholds for acceptable audio delay and triggers the video output to drop several frames to resynchronize audio and video. (Blue square at the top), thus restoring avdiff to a value around 0.
5. The apparent audio position rapidly approaches its “true” value again over the course of the next 1-2 seconds (again, as observed earlier)
6. This causes the measured av difference (black line) to very rapidly grow upwards again within these 1-2 seconds, triggering several frame drops a long the way (every time it exceeds the acceptable threshold) - these are the clearly visible spikes and corresponding blue frames in the few seconds after the seek.
The only known work-around in mpv is to decrease the pulse buffer size (tlength) to a smaller value, which makes the issue less severe (as seen on the left) but doesn't fully solve it.
It would be great if this could get fixed upstream in PulseAudio so we don't have to hack around it in mpv.
**Attachment 126489**, "regular playback":
![cork-demo](/uploads/2cf4b06fa04d75631973b3a2acb8a7df/cork-demo.png)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/9280% (512MB) of gnome-shell's memory mappings at start-up are due to PulseAudio2018-07-30T09:41:05ZBugzilla Migration User80% (512MB) of gnome-shell's memory mappings at start-up are due to PulseAudio## Submitted by Daniel van Vugt
Assigned to **pul..@..op.org**
**[Link to original bug (#105779)](https://bugs.freedesktop.org/show_bug.cgi?id=105779)**
## Description
Created attachment 138392
gnome-shell-all-mmaps-at-startup.pdf...## Submitted by Daniel van Vugt
Assigned to **pul..@..op.org**
**[Link to original bug (#105779)](https://bugs.freedesktop.org/show_bug.cgi?id=105779)**
## Description
Created attachment 138392
gnome-shell-all-mmaps-at-startup.pdf
80% (512MB) of gnome-shell's memory mappings at start-up are due to PulseAudio. This seems excessive.
Memory profile from Google heap profiler attached.
Although this may not be a bug. It might just be reserved address space with very little physical memory impact...?
**Attachment 138392**, "gnome-shell-all-mmaps-at-startup.pdf":
[gnome-shell-all-mmaps-at-startup.pdf](/uploads/8eb34fd08a6e3ecad053214bde4df84e/gnome-shell-all-mmaps-at-startup.pdf)
### See also
* https://launchpad.net/bugs/1759497https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/108pa_mainloop should report POLLNVAL when that occurs with IO events2018-07-30T09:54:03ZBugzilla Migration Userpa_mainloop should report POLLNVAL when that occurs with IO events## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#45044)](https://bugs.freedesktop.org/show_bug.cgi?id=45044)**
## Description
poll() may return POLLNVAL in revents, but pa_mainloop ig...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#45044)](https://bugs.freedesktop.org/show_bug.cgi?id=45044)**
## Description
poll() may return POLLNVAL in revents, but pa_mainloop ignores that. According to "man poll", POLLNVAL signals an "invalid request: fd not open (output only)". I'm not sure how that should be interpreted: if we never call poll() on an fd that we have closed ourselves, is it guaranteed that POLLNVAL will never occur? If so, we could have an assertion for that in map_flags_from_libc(). Or probably not: pa_mainloop can also be used by clients, and they may not be as careful with their file descriptors.
As proposed by Arun, I think we should add PA_IO_EVENT_INVALID to pa_io_event_flags_t.
Discussion:
http://lists.freedesktop.org/archives/pulseaudio-discuss/2011-December/012517.htmlhttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/117Stereo/surround detection2018-07-30T09:54:40ZBugzilla Migration UserStereo/surround detection## Submitted by fdo..@..ne.net
Assigned to **pul..@..op.org**
**[Link to original bug (#39360)](https://bugs.freedesktop.org/show_bug.cgi?id=39360)**
## Description
Currently it would seem that one has to choose between stereo or ...## Submitted by fdo..@..ne.net
Assigned to **pul..@..op.org**
**[Link to original bug (#39360)](https://bugs.freedesktop.org/show_bug.cgi?id=39360)**
## Description
Currently it would seem that one has to choose between stereo or surround output, e.g. an hdmi 2.0 sink or an hdmi 5.1 sink.
This means that either multichannel content such as DVD-A, movies or other video content will be downmixed to stereo or that stereo content will be outputted as multichannel (with several 'silent' channels or upmixed, depending on the 'enable-remixing' setting). The behaviour prevents receivers from upmixing the signal to surround themselves with arguably better algorithms such as DPL2, Neo:6 or Logic-7.
Therefore, I would like to request a feature to have detection for the max number of channels in the active source(s) and use the appropriate sink in the right mode. This will allow audio to be passed through 'natively' and leave the processing up to the receiver.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/138Replace dummy modules with a list of deprecated module aliases in the module ...2018-07-30T09:57:58ZBugzilla Migration UserReplace dummy modules with a list of deprecated module aliases in the module loading code## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#60146)](https://bugs.freedesktop.org/show_bug.cgi?id=60146)**
## Description
Sometimes we want to rename modules. That's no reason to ...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#60146)](https://bugs.freedesktop.org/show_bug.cgi?id=60146)**
## Description
Sometimes we want to rename modules. That's no reason to break user configuration. We have handled this sometimes by keeping a compatibility module that loads the new module, and sometimes we have simply broken configuration compatibility. Dummy modules take up compilation time and add clutter the source tree. I propose a lighter-weight solution: maintain a simple list of deprecated module aliases in the module loading code. If configuration contains a module that is on the list, the module loading code issues a warning and loads the modern version of the module.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/140pulseaudio won't run when home directory is mounted with CIFS2018-07-30T09:58:23ZBugzilla Migration Userpulseaudio won't run when home directory is mounted with CIFS## Submitted by Maarten Jacobs
Assigned to **pul..@..op.org**
**[Link to original bug (#71649)](https://bugs.freedesktop.org/show_bug.cgi?id=71649)**
## Description
I'm fairly sure this is a duplicate of an existing bug as I'm not...## Submitted by Maarten Jacobs
Assigned to **pul..@..op.org**
**[Link to original bug (#71649)](https://bugs.freedesktop.org/show_bug.cgi?id=71649)**
## Description
I'm fairly sure this is a duplicate of an existing bug as I'm not the first to experience this issue... However I have not been able to find a record of this bug on Bugzilla.
When a user's home directory is crossmounted on a linux machine from another host using CIFS, pulseaudio will not start as pulseaudio cannot set the necessary permissions on the .config/pulse directory (CIFS doesn't allow permissions).
I have put together a quick fix that tests whether the filesystem is a CIFS filesystem, and returns early from the procedure "pa_make_secure_dir" to prevent pulseaudio to fail:
--- core-util.c 2013-11-14 23:05:47.709911713 -0500
+++ core-util.c.orig 2013-11-14 23:10:28.314384314 -0500
@@ -37,7 +37,6 @@
#include <ctype.h>
#include <sys/types.h>
#include <sys/stat.h>
-#include <sys/vfs.h>
#include <dirent.h>
#ifdef HAVE_LANGINFO_H
@@ -224,7 +223,6 @@
* already exist, however. */
int pa_make_secure_dir(const char* dir, mode_t m, uid_t uid, gid_t gid, bool update_perms) {
struct stat st;
- struct statfs stfs;
int r, saved_errno;
bool retry = true;
@@ -283,14 +281,6 @@
if (!update_perms)
return 0;
- if (fstatfs(fd, &stfs) < 0) {
- pa_assert_se(pa_close(fd) >= 0);
- goto fail;
- }
-
- if (stfs.f_type == 0xFF534D42)
- return 0;
-
#ifdef HAVE_FCHOWN
if (uid == (uid_t) -1)
uid = getuid();
I have tested this on my installation (Linux Mint 14 32-bit) and it appears to work. I don't know how to submit the fix properly so I'm hoping somebody will see this and help guide me on this.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/142error: undefined reference to 'lt__PROGRAM__LTX_preloaded_symbols'2019-03-08T15:04:35ZBugzilla Migration Usererror: undefined reference to 'lt__PROGRAM__LTX_preloaded_symbols'## Submitted by Tomasz Paweł Gajc
Assigned to **pul..@..op.org**
**[Link to original bug (#99675)](https://bugs.freedesktop.org/show_bug.cgi?id=99675)**
## Description
Can't build PulseAudio 10.0 with gcc-6.3.1
make[3]: Leaving d...## Submitted by Tomasz Paweł Gajc
Assigned to **pul..@..op.org**
**[Link to original bug (#99675)](https://bugs.freedesktop.org/show_bug.cgi?id=99675)**
## Description
Can't build PulseAudio 10.0 with gcc-6.3.1
make[3]: Leaving directory '/builddir/build/BUILD/pulseaudio-10.0/src'
libtool: error: not configured to extract global symbols from dlpreopened files
daemon/main.c:528: error: undefined reference to 'lt__PROGRAM__LTX_preloaded_symbols'
daemon/dumpmodules.c:116: error: undefined reference to 'lt__PROGRAM__LTX_preloaded_symbols'
daemon/dumpmodules.c:129: error: undefined reference to 'lt__PROGRAM__LTX_preloaded_symbols'
collect2: error: ld returned 1 exit status
make[3]: *** [Makefile:6723: pulseaudio] Error 1
make[2]: *** [Makefile:4891: all] Error 2
make[1]: *** [Makefile:807: all-recursive] Error 1
make: *** [Makefile:622: all] Error 2
error: Bad exit status from /var/tmp/rpm-tmp.73086 (%build)
More logs can be found here
http://file-store.openmandriva.org/api/v1/file_stores/b735cedd3c73cf523804b2354a1dfde3641d22e3.log?show=truehttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/146Channels are swapped when mixing two stereo streams2018-07-30T09:58:51ZBugzilla Migration UserChannels are swapped when mixing two stereo streams## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#53206)](https://bugs.freedesktop.org/show_bug.cgi?id=53206)**
## Description
I played two songs (stereo audio) simultaneously, one wit...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#53206)](https://bugs.freedesktop.org/show_bug.cgi?id=53206)**
## Description
I played two songs (stereo audio) simultaneously, one with the left channel muted and one with the right channel muted. I listened with headphones, and the one that should have played to the left ear played to the right ear and vice versa. One of the songs was shorter than the other, and when the shorter one ended, the longer song jumped from the right ear to the left ear.
The symptoms suggest that the mixing code is broken: it swaps channels. With two streams active the channels are the wrong way around, but with only one stream the channels are right.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/151lag between analog and digital outputs using module-combine-sink2018-07-30T09:59:21ZBugzilla Migration Userlag between analog and digital outputs using module-combine-sink## Submitted by aso..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#64017)](https://bugs.freedesktop.org/show_bug.cgi?id=64017)**
## Description
Setting up simultaneous analog and digital output results in rela...## Submitted by aso..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#64017)](https://bugs.freedesktop.org/show_bug.cgi?id=64017)**
## Description
Setting up simultaneous analog and digital output results in relative lag between the analog and digital outputs on my Nforce2 motherboard with onboard sound.
This is using the example provided in the pulseaudio archlinux wiki, here:
https://wiki.archlinux.org/index.php/PulseAudio/Examples#Simultaneous_HDMI_and_Analog_Output
.. with the following "default.pa" entries:
### Load analog device
load-module module-alsa-sink device=hw:0,2
load-module module-combine-sink sink_name=combined
set-default-sink combined
I can be watching a movie on VLC and a relative delay between analog and digital outputs can develop, so that a sound will come out one set of speakers half a second after they come out the other, making movies unwatchable. A fast way to make the problem occur is to change tracks on Clementine, making use of the crossfade feature, resulting in an element of mixing. The relative lag is non-fixed and sporadic.
This problem does not occur using a custom .asoundrc to setup simultaneous digital and analog output using ALSA without Pulseaudio.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/162PA crash on Assertion 'pa_atomic_load(&b->pool->stat.n_imported) > 02018-07-30T10:00:11ZBugzilla Migration UserPA crash on Assertion 'pa_atomic_load(&b->pool->stat.n_imported) > 0## Submitted by Lode Cools
Assigned to **pul..@..op.org**
**[Link to original bug (#95352)](https://bugs.freedesktop.org/show_bug.cgi?id=95352)**
## Description
Created attachment 123624
PA backtraces
PA version 7.1.2
After a fe...## Submitted by Lode Cools
Assigned to **pul..@..op.org**
**[Link to original bug (#95352)](https://bugs.freedesktop.org/show_bug.cgi?id=95352)**
## Description
Created attachment 123624
PA backtraces
PA version 7.1.2
After a few days running, PA crashes with the attached backtrace and debug log.
In my setup, I use a lot of chromium & GST-pulsesink sink-inputs which are played on the same module-combine-sink which is than playing on a HDMI and analog output.
I can't really make up what is going wrong...
**Attachment 123624**, "PA backtraces":
[backtraces.txt](/uploads/46848553260b3655452c271ff65f123e/backtraces.txt)https://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)https://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/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/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/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/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/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/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/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/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/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/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/236No sound from mic through pulseaudio, however recording with alsa works fine2018-07-30T10:08:45ZBugzilla Migration UserNo sound from mic through pulseaudio, however recording with alsa works fine## Submitted by vxl..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#49311)](https://bugs.freedesktop.org/show_bug.cgi?id=49311)**
## Description
Have a Acer Aspire One D225E set up with Gnome 3 and PulseAudio (...## Submitted by vxl..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#49311)](https://bugs.freedesktop.org/show_bug.cgi?id=49311)**
## Description
Have a Acer Aspire One D225E set up with Gnome 3 and PulseAudio (Arch Linux). I can record sound through my mic just fine with arecord, audacity set to alsa, etc.
However with PulseAudio I get nothing. Input does not register in pavucontrol (though everything seems set up fine) and I can hear nothing through my mic when using Skype.
I've tried the solution here with no success:
https://getsatisfaction.com/jolicloud/topics/deaf_internal_mic_on_acer_aspire_one#reply_2108048https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/239Add a way to override priority of sinks - mostly for module-combine-sinks2018-07-30T10:08:58ZBugzilla Migration UserAdd a way to override priority of sinks - mostly for module-combine-sinks## Submitted by Myst Fox
Assigned to **pul..@..op.org**
**[Link to original bug (#101201)](https://bugs.freedesktop.org/show_bug.cgi?id=101201)**
## Description
This is a bit hard to describe. I'd really like to see a way to manua...## Submitted by Myst Fox
Assigned to **pul..@..op.org**
**[Link to original bug (#101201)](https://bugs.freedesktop.org/show_bug.cgi?id=101201)**
## Description
This is a bit hard to describe. I'd really like to see a way to manually set priorities on sinks. I recently got a portable DAC, and I wanted to use it when plugged in and fall back on module-combine-sink otherwise - mostly because I often use a program that cannot handle changing sinks. This however means I needed my priorities to be correctly ordered. After fixing the identification of my DAC on the udev side, and switching to running a git build to have recent fixes like https://bugs.freedesktop.org/show_bug.cgi?id=99222, I ran into that I wanted the module-combine-sink sink to be above my internal card but below my DAC. I solved this for now with `load-module module-combine-sink sink_properties='device.class="sound" device.form_factor="speaker"'
`, but that's faking out the priority setting to get the right order, not doing it any sort of "right".
I'm imagining adding a proplist entry of some naming that lets you outright set the priority, and there would be a check at the start of pa_device_init_priority in sink.c that would outright set the priority if it was a valid value. I'd like to know, though, how that sounds, what people think, ideas about the name, if there's anything big I might be missing, before I try to work on a patch.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/241libpulsecommon.so unaligned memory access issue2018-07-30T10:09:06ZBugzilla Migration Userlibpulsecommon.so unaligned memory access issue## Submitted by Yupeng Chang
Assigned to **pul..@..op.org**
**[Link to original bug (#65474)](https://bugs.freedesktop.org/show_bug.cgi?id=65474)**
## Description
I cross compiled PulseAudio 4.0 using codesourcery 2011.09 to ARM p...## Submitted by Yupeng Chang
Assigned to **pul..@..op.org**
**[Link to original bug (#65474)](https://bugs.freedesktop.org/show_bug.cgi?id=65474)**
## Description
I cross compiled PulseAudio 4.0 using codesourcery 2011.09 to ARM platform with the following CFLAGS
-march=armv6k -mtune=arm1136j-s -msoft-float -O3 -fPIC
pulseaudio daemon runs perfectly without any problem, but when I start to run client application like pactl info, the connection always fail with the following error messages:
# pactl info
I: [pulseaudio][pulsecore/client.c:80 pa_client_new()] Created 0 "Native client (UNIX socket client)"
D: [pulseaudio][pulsecore/protocol-dbus.c:773 pa_dbus_protocol_add_interface()] Interface org.PulseAudio.Core1.Client added for object /org/pulseaudio/core1/client0
D: [pulseaudio][pulsecore/protocol-native.c:2599 command_auth()] Protocol version: remote 28, local 28
I: [pulseaudio][pulsecore/protocol-native.c:2632 command_auth()] Got credentials: uid=0 gid=0 success=1
D: [pulseaudio][pulsecore/protocol-native.c:2662 command_auth()] SHM possible: yes
D: [pulseaudio][pulsecore/protocol-native.c:2680 command_auth()] Negotiated SHM: yes
E: [pulseaudio][pulsecore/protocol-native.c:1939 protocol_error()] protocol error, kicking client
I: [pulseaudio][pulsecore/client.c:102 pa_client_free()] Freed 0 "Native client (UNIX socket client)"
D: [pulseaudio][pulsecore/protocol-dbus.c:835 pa_dbus_protocol_remove_interface()] Interface org.PulseAudio.Core1.Client removed from object /org/pulseaudio/core1/client0
Connection failure: Connection terminated
I did some digging on this error, I found connection is terminated on function: command_set_client_name(), and the root failure is in function: pa_tagstruct_get_arbitrary(), on this check "if (t->data[t->rindex] != PA_TAG_ARBITRARY)".
Then I print out t->data[t->rindex] value to see if it's PA_TAG_ARBITRARY, what is strange is that this value is a randome value.
I go back to check the compiling log, and found the following warnings:
pulsecore/hashmap.c: In function 'remove_entry':
pulsecore/hashmap.c:94:9: warning: cast increases required alignment of target type [-Wcast-align]
pulsecore/hashmap.c: In function 'hash_scan':
pulsecore/hashmap.c:116:14: warning: cast increases required alignment of target type [-Wcast-align]
pulsecore/hashmap.c: In function 'pa_hashmap_put':
pulsecore/hashmap.c:141:22: warning: cast increases required alignment of target type [-Wcast-align]
pulsecore/hashmap.c:143:9: warning: cast increases required alignment of target type [-Wcast-align]
pulsecore/hashmap.c:144:9: warning: cast increases required alignment of target type [-Wcast-align]
pulsecore/hashmap.c:145:5: warning: cast increases required alignment of target type [-Wcast-align]
These files are included in libpulsecommon.so
Then I have to add -mno-unaligned-access to CLFAGS, then re-compile pulseaudio again. I replaced the original one with newly compiled libpulsecommon.so, leaving other libraries untouched, everything goes well without any issue.
So I think this must be an unaligned memory access issue in libpulsecommon.so
Please help to fix this!!https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/252[530U3C/530U4C, Realtek ALC269VC, Speaker, Internal] No sound at all2018-07-30T10:09:54ZBugzilla Migration User[530U3C/530U4C, Realtek ALC269VC, Speaker, Internal] No sound at all## Submitted by Cristian Aravena
Assigned to **pul..@..op.org**
**[Link to original bug (#81587)](https://bugs.freedesktop.org/show_bug.cgi?id=81587)**
## Description
https://bugzilla.kernel.org/show_bug.cgi?id=80771
0) Play musi...## Submitted by Cristian Aravena
Assigned to **pul..@..op.org**
**[Link to original bug (#81587)](https://bugs.freedesktop.org/show_bug.cgi?id=81587)**
## Description
https://bugzilla.kernel.org/show_bug.cgi?id=80771
0) Play music with Rhytmbox => OK!
1) Crash gnome-shell
2) Restore system automatic gnome-shell
3) Not sound (chromium (youtube); rhytmbox, etc.)
Not go rhytmbox, not play music.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/255Add support for the dmalloc memory checking tool.2018-07-30T10:11:18ZBugzilla Migration UserAdd support for the dmalloc memory checking tool.## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#46989)](https://bugs.freedesktop.org/show_bug.cgi?id=46989)**
## Description
The dmalloc (http://dmalloc.com/) tool is somewhat useful...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#46989)](https://bugs.freedesktop.org/show_bug.cgi?id=46989)**
## Description
The dmalloc (http://dmalloc.com/) tool is somewhat useful tool for checking memory issues. Valgrind is probably "better", but it slows things down too much in some environments.
The level of "support" I'd like to have in Pulseaudio is just including dmalloc.h in pulse/xmalloc.c if HAVE_DMALLOC is defined. This is pretty useless for tracking down memory leaks, because dmalloc only reports the immediate caller of malloc(), which in case of Pulseaudio will always be pa_xmalloc(). There are checks for other errors than memory leaks, so dmalloc can still be useful.
So, what needs to be done is adding this to pulse/xmalloc.c:
#ifdef HAVE_DMALLOC
#include <dmalloc.h>
#endif
In addition to that, Makefile.am of course needs some adjustment, and I'd like to have --enable-dmalloc switch in the configure script.
Note that linking libdmallocth into libpulse requires dmalloc to be built with -fPIC. That isn't enabled by default in dmalloc's build system, so this needs to be mentioned in "configure --help" for the --enable-dmalloc switch. It would be nice to check in the configure script whether the installed libdmallocth.a is built with -fPIC, but I don't know how that can be done, or whether it's even possible.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/270TV audio output device not remembered2018-07-30T10:12:49ZBugzilla Migration UserTV audio output device not remembered## Submitted by ami..@..is.net
Assigned to **pul..@..op.org**
**[Link to original bug (#88823)](https://bugs.freedesktop.org/show_bug.cgi?id=88823)**
## Description
I've recently upgraded to Kubuntu 14.10 (KDE 4.14.2), and started...## Submitted by ami..@..is.net
Assigned to **pul..@..op.org**
**[Link to original bug (#88823)](https://bugs.freedesktop.org/show_bug.cgi?id=88823)**
## Description
I've recently upgraded to Kubuntu 14.10 (KDE 4.14.2), and started getting this issue: Whenever my monitors enter energy savings mode (when pc is idle for a while - the pc itself is not suspended, only the monitors are), then the TV (connected as third display to PC) is no longer remembered as an audio output device.
So every time I want to use the TV I must go to K Menu -> System Settings -> Multimedia -> Audio and Video Settings -> Audio Hardware Setup -> Profile and re-select the proper profile. Sound Card is "Built-in Audio", and there are 3 different profiles in the dopdown list all identically named "Digital Stereo (HDMI) Output", however selecting each of them gives a different Connector name below: "HDMI / DisplayPort", "HDMI / DisplayPort 2 " and "HDMI / DisplayPort 3".
In my case the second one is the correct one, but when a screen energy savings mode is activated it reverts back to the third, which is incorrect.
This issue did not occur before the upgrade, and there were no other hardware changes.
See https://bugs.kde.org/show_bug.cgi?id=340979 for the previous history and discussion of this issue.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/272When in idle, PulseAudio uses from 2% to 7% CPU core load2018-07-30T10:12:58ZBugzilla Migration UserWhen in idle, PulseAudio uses from 2% to 7% CPU core load## Submitted by Germano Massullo
Assigned to **pul..@..op.org**
**[Link to original bug (#105840)](https://bugs.freedesktop.org/show_bug.cgi?id=105840)**
## Description
Created attachment 138481
oprofile_data
Operating system: Fe...## Submitted by Germano Massullo
Assigned to **pul..@..op.org**
**[Link to original bug (#105840)](https://bugs.freedesktop.org/show_bug.cgi?id=105840)**
## Description
Created attachment 138481
oprofile_data
Operating system: Fedora 27
pulseaudio-11.1-15.fc27.x86_64
KDE Frameworks 5.44.0
Qt 5.9.4
plasma-desktop-5.12.3-2.fc27.x86_64
I noticed that when I log into Plasma and my system is fully in idle, without any user application started, pulseaudio process load is in the range of [2, 5]%
I attach operf log
**Attachment 138481**, "oprofile_data":
[oprofile_data.tar.xz](/uploads/7dcc30b37f8b2cec4bce5cd7c0afdc3e/oprofile_data.tar.xz)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/278pa_sink_input_new() assumes that data->sample_spec is valid.2020-01-27T22:49:43ZBugzilla Migration Userpa_sink_input_new() assumes that data->sample_spec is valid.## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#51719)](https://bugs.freedesktop.org/show_bug.cgi?id=51719)**
## Description
If the sink input new data has all PA_SINK_INPUT_FIX_* fl...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#51719)](https://bugs.freedesktop.org/show_bug.cgi?id=51719)**
## Description
If the sink input new data has all PA_SINK_INPUT_FIX_* flags set, then pa_sink_input_new() should ignore the sample spec and channel map of the sink input new data. That's not currently the case: at least pa_format_info_from_sample_spec() is called with data->sample_spec as the parameter, and that will crash if the sample spec is not valid.
One part of this problem is that it's not possible to set only one of sample format and sample rate in pa_sink_input_new_data. This is a problem at least with module-loopback: it accepts the "format" and "rate" module arguments, and it's a perfectly reasonable thing for the user to set only one of them. If the user sets only the sample rate, for example, then it should be possible for module-loopback to only set the rate in pa_sink_input_new_data and leave the format unspecified.
There's a FIXME item related to this, search for "FIXME" in this commit diff: http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/?id=c6d8d1d7c19a105b224eac393e44bae319897b6b
(The same issue exists for source outputs too.)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/289Daemon can hang after [null-sink] asyncq.c: q overrun, queuing locally2018-07-30T10:14:13ZBugzilla Migration UserDaemon can hang after [null-sink] asyncq.c: q overrun, queuing locally## Submitted by Marcin Lewandowski
Assigned to **pul..@..op.org**
**[Link to original bug (#96749)](https://bugs.freedesktop.org/show_bug.cgi?id=96749)**
## Description
Created attachment 124804
gdb dump
Version 9.0.
I have conn...## Submitted by Marcin Lewandowski
Assigned to **pul..@..op.org**
**[Link to original bug (#96749)](https://bugs.freedesktop.org/show_bug.cgi?id=96749)**
## Description
Created attachment 124804
gdb dump
Version 9.0.
I have connected many (+/- 100) GStreamer based clients over native protocol with disabled SHM and realtime scheduling.
Each client that has GStreamer's pulsesink loads module-null-sink and sends data to this sink.
Each client that has GStreamer's pulsesrc loads module-null-sink and fetches data from this sink's monitor.
Sporadically I load loopback module to "connect" these null sinks and their monitors.
It works fine for a while and then loads of
[null-sink] asyncq.c: q overrun, queuing locally
appear in the syslog.
All operations of PA are stopped. It's not desired behaviour, even if there's something wrong with some of the clients it should not cause whole daemon to hang.
Stack trace is attached.
**Attachment 124804**, "gdb dump":
[pa-gdb.txt](/uploads/e8e6ca76bb81e86237edf9ccfebd466d/pa-gdb.txt)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/293Implement pa_mainloop_api around pa_rtpoll2018-07-30T10:14:39ZBugzilla Migration UserImplement pa_mainloop_api around pa_rtpoll## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#73429)](https://bugs.freedesktop.org/show_bug.cgi?id=73429)**
## Description
Most modules implement the IO thread event loop with pa_r...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#73429)](https://bugs.freedesktop.org/show_bug.cgi?id=73429)**
## Description
Most modules implement the IO thread event loop with pa_rtpoll, but module-tunnel-sink/source-new implement the event loop with pa_mainloop. module-rtp-recv assumes that the IO thread event loop is implemented with pa_rtpoll, so using module-rtp-recv with module-tunnel-sink-new results in a crash.
I think the interaction with the event loop in module-rtp-recv and elsewhere should be done through pa_mainloop_api instead of accessing the underlying implementation directly. This requires a pa_mainloop_api implementation based on pa_rtpoll.
### Blocking
* [Bug 73426](https://bugs.freedesktop.org/show_bug.cgi?id=73426)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/294Without tsched=0 for module-udev-detect, sound is distorted2019-11-27T18:20:33ZBugzilla Migration UserWithout tsched=0 for module-udev-detect, sound is distorted## Submitted by Josef Hahn
Assigned to **pul..@..op.org**
**[Link to original bug (#50510)](https://bugs.freedesktop.org/show_bug.cgi?id=50510)**
## Description
I've had sound distortions at least with many versions of VLC player ...## Submitted by Josef Hahn
Assigned to **pul..@..op.org**
**[Link to original bug (#50510)](https://bugs.freedesktop.org/show_bug.cgi?id=50510)**
## Description
I've had sound distortions at least with many versions of VLC player for a long time. Although mostly VLC was affected, i think other programs randomly had the same issues. The sound is distorted and has an echo component. Fiddling at the pulseaudio volume control for some seconds fixes this problem until the player opens a new stream.
Adding "tsched=0" to the options for the module-udev-detect module prevents the issue here.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/313"Headphone" audio output missing on Lenovo T440s after upgrade to commit aec8112018-07-30T10:16:52ZBugzilla Migration User"Headphone" audio output missing on Lenovo T440s after upgrade to commit aec811## Submitted by Andreas Fleig
Assigned to **pul..@..op.org**
**[Link to original bug (#86032)](https://bugs.freedesktop.org/show_bug.cgi?id=86032)**
## Description
Upgrading pulseaudio to commit aec811 on my Lenovo T440s with Fedo...## Submitted by Andreas Fleig
Assigned to **pul..@..op.org**
**[Link to original bug (#86032)](https://bugs.freedesktop.org/show_bug.cgi?id=86032)**
## Description
Upgrading pulseaudio to commit aec811 on my Lenovo T440s with Fedora 21 made the headphone audio output disappear.
Repeatable: always
Steps to reproduce:
1. Install pulseaudio-5.0-24.20141103gitaec81.fc21.x86_64 on Lenovo T440s
2. Reboot
3. Play audio, open gnome-control-center.
4. Plug in some headphones
Expected results:
gnome-control-center should show a new audio output device "Headphones", there
should be sound from the headphones
Actual results:
No audio output device displayed any more, and no sound from headphones
$ rpm -q pulseaudio
pulseaudio-5.0-24.20141103gitaec81.fc21.x86_64
alsa-info:
http://www.alsa-project.org/db/?f=503cc4ba2634c1098053dddf70051f65c717a7f5
The latest known working version of the package, pulseaudio-5.0-10.fc21, appears to be based on the 5.0 release:
http://pkgs.fedoraproject.org/cgit/pulseaudio.git/tree/sources?h=f21&id=7167b336b7f6e7457f6dfd60b055625560ca7532https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/324Unnecessary copying in pa_sink_get_formats()2018-07-30T10:17:42ZBugzilla Migration UserUnnecessary copying in pa_sink_get_formats()## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#71924)](https://bugs.freedesktop.org/show_bug.cgi?id=71924)**
## Description
pa_sink_get_formats() returns a freshly allocated idxset ...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#71924)](https://bugs.freedesktop.org/show_bug.cgi?id=71924)**
## Description
pa_sink_get_formats() returns a freshly allocated idxset with freshly allocated format info objects every time it's called. In my opinion this is waste of CPU cycles, and also makes life harder for the callers of the function, because they have to free the returned formats.
Currently the sink formats are stored inside the sink backend code. The formats could also be stored in the pa_sink struct, in which case the whole pa_sink_get_format() function could be removed (there's some point in keeping the getter function for encapsulation reasons, though, but I'd prefer not to keep it). If the formats are stored in pa_sink, then that raises a question how pa_sink_set_formats() would work. The sink backend should still handle that operation, but would the backend write directly to pa_sink.formats, or should it be done in some other way? I personally would prefer to do it in some other way, because while I slightly dislike getter functions, I do like setter functions, so the backend shouldn't write directly to fields in the pa_sink struct.
What would the "setter function" be called? pa_sink_set_formats() is a no-go, because that's the function from which the setter function would be called. I propose that pa_sink_formats_changed() is added. It would be called by the sink backend when the formats change in the backend, similarly how pa_sink_volume_changed() and pa_sink_mute_changed() are called when the volume/mute changes in the backend.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/326Crackling with snd-hda-intel / VIA VT2020 in VLC, Google Voice2018-07-30T10:17:52ZBugzilla Migration UserCrackling with snd-hda-intel / VIA VT2020 in VLC, Google Voice## Submitted by Edward Liaw
Assigned to **pul..@..op.org**
**[Link to original bug (#62604)](https://bugs.freedesktop.org/show_bug.cgi?id=62604)**
## Description
Further described here:
https://bugs.launchpad.net/ubuntu/+source/pu...## Submitted by Edward Liaw
Assigned to **pul..@..op.org**
**[Link to original bug (#62604)](https://bugs.freedesktop.org/show_bug.cgi?id=62604)**
## Description
Further described here:
https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/751265
My hardware:
$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: SB [HDA ATI SB], device 0: VT2020 Analog [VT2020 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: SB [HDA ATI SB], device 1: VT2020 Digital [VT2020 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: SB [HDA ATI SB], device 2: VT2020 HP [VT2020 HP]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 2: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
$ lspci -v | grep -A7 -i "audio"
00:14.2 Audio device: Advanced Micro Devices [AMD] nee ATI SBx00 Azalia (Intel HDA) (rev 40)
Subsystem: ASUSTeK Computer Inc. M4A89GTD PRO/USB3 Motherboard
Flags: bus master, slow devsel, latency 64, IRQ 16
Memory at fe4f8000 (64-bit, non-prefetchable) [size=16K]
Capabilities: `<access denied>`
Kernel driver in use: snd_hda_intel
00:14.3 ISA bridge: Advanced Micro Devices [AMD] nee ATI SB7x0/SB8x0/SB9x0 LPC host controller (rev 40)
--
06:00.1 Audio device: Advanced Micro Devices [AMD] nee ATI Cayman/Antilles HDMI Audio [Radeon HD 6900 Series]
Subsystem: XFX Pine Group Inc. Device aa80
Flags: bus master, fast devsel, latency 0, IRQ 86
Memory at fe9bc000 (64-bit, non-prefetchable) [size=16K]
Capabilities: `<access denied>`
Kernel driver in use: snd_hda_intel
#23 in the linked bug report fixed the crackling for me, but it was recommended that a bug report be filed.
In /etc/pulse/default.pa : I added tsched=0 to the end of load-module module-udev-detect
#61 gives a more technical description that may be of use:
It's all very confusing with many people experiencing similar symptoms. But the original logs look like a simple case of PulseAudio selecting a far too short latency:
I: protocol-native.c: Final latency 201.00 ms = 0.50 ms + 2*100.00 ms + 0.50 ms
D: alsa-sink.c: Cutting sleep time for the initial iterations by half.
D: alsa-sink.c: Latency set to 0.50ms
And indeed, VLC version 1.1.8 requested tlength=200ms and minreq=100ms. A similar problem affects more recent VLC versions with tlength=40ms and minreq=-1 (overriden by PA to 20ms). It seems that when tlength is exactly twice minreq, PulseAudio settles on an insanely small latency. That causes frequent underrun in hardware buffers, accounting for the distorsion.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/330pulseaudio will continue using high cpu after hibernate2018-07-30T10:18:08ZBugzilla Migration Userpulseaudio will continue using high cpu after hibernate## Submitted by TOM_Harrison
Assigned to **pul..@..op.org**
**[Link to original bug (#71078)](https://bugs.freedesktop.org/show_bug.cgi?id=71078)**
## Description
pulseaudio will using high cpu after wakeup from hibernate.
the onl...## Submitted by TOM_Harrison
Assigned to **pul..@..op.org**
**[Link to original bug (#71078)](https://bugs.freedesktop.org/show_bug.cgi?id=71078)**
## Description
pulseaudio will using high cpu after wakeup from hibernate.
the only solution is to kill the pulseaudio daemon and restart.
environment
kubuntu 13.10 AMD64
kde 4.11.2
libpulse0:amd64 1:4.0-0ubuntu6
pulseaudio with alsa dmix and dsnoop.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/332microphone not working, strange fix2018-07-30T10:18:20ZBugzilla Migration Usermicrophone not working, strange fix## Submitted by mattia.b89
Assigned to **pul..@..op.org**
**[Link to original bug (#99241)](https://bugs.freedesktop.org/show_bug.cgi?id=99241)**
## Description
I have a Dell XPS 13 (2015) with Broadwell hardware.
I am on Arch Lin...## Submitted by mattia.b89
Assigned to **pul..@..op.org**
**[Link to original bug (#99241)](https://bugs.freedesktop.org/show_bug.cgi?id=99241)**
## Description
I have a Dell XPS 13 (2015) with Broadwell hardware.
I am on Arch Linux x86_64, latest version of Pulseaudio and pavucontrol.
When I power on my laptop, microphone is not working.
In order to enable it I have to use that instructions (https://wiki.archlinux.org/index.php/Dell_XPS_13_(9343)#Enabling_the_microphone).
Issue n.1: Why my microphone is not working "automatically" ?
As you note, there are two ways in order to achieve the goal; the second one is using `pavucontrol`, simply switching the two entries in the menu (full instructions here again: https://wiki.archlinux.org/index.php/Dell_XPS_13_(9343)#Enabling_the_microphone)
Issue n.2: Why I have to switch twice in order to get the working microphone ?
If you need further information I am herehttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/346Glitching when adjusting volume2018-07-30T10:19:42ZBugzilla Migration UserGlitching when adjusting volume## Submitted by tro..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#79991)](https://bugs.freedesktop.org/show_bug.cgi?id=79991)**
## Description
Created attachment 100986
glitch reproduced on 220 hz sine wave, ...## Submitted by tro..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#79991)](https://bugs.freedesktop.org/show_bug.cgi?id=79991)**
## Description
Created attachment 100986
glitch reproduced on 220 hz sine wave, adjusted output volume from indicator applet
When adjusting the volume with pulse audio's volume control window the sound has small glitches.
This is sometimes not possible to hear, sometimes really apparent, depending on the type of sound played, and which volume control is changed.
Can be clearly heard with a test tone, generate a sine wave of 220 hz in audacity and set it to play.
Adjusting the output device slider (analog output) gives the most glitching (note: this is not just the pops that are supposed to be there!).
Adjusting the volume from the taskbar applet also adjust this same slider, without the pops that are supposed to be there. The glitches are still there.
Adjusting the program specific control under playback also causes glitching, but much less.
Confirmed on two computers.
**Attachment 100986**, "glitch reproduced on 220 hz sine wave, adjusted output volume from indicator applet":
[_pulseverbose.log](/uploads/1dd66909cf619eb3465d290278252ea5/_pulseverbose.log)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/347Feature request: always refilter LFE on laptops, even if already present2019-09-29T17:04:00ZBugzilla Migration UserFeature request: always refilter LFE on laptops, even if already present## Submitted by Alexander E. Patrakov
Assigned to **pul..@..op.org**
**[Link to original bug (#95506)](https://bugs.freedesktop.org/show_bug.cgi?id=95506)**
## Description
Some laptops have an internal subwoofer. The correct corne...## Submitted by Alexander E. Patrakov
Assigned to **pul..@..op.org**
**[Link to original bug (#95506)](https://bugs.freedesktop.org/show_bug.cgi?id=95506)**
## Description
Some laptops have an internal subwoofer. The correct corner frequency for the crossover filter to be used with such subwoofer is, let's say, 700 Hz. Definitely not 120 Hz.
Let's see what happens if the user sets lfe-crossover-freq to 700, and then listens to (2.0) music and watches a film witha 5.1 soundtrack.
Music doesn't have the LFE channel, so PulseAudio will synthesize it (correctly, except for the overall gain, which is tracked as bug #95021). Frequencies below 700 Hz will be sent to the subwoofer, frequencies above that will be sent to the main channels. So all is good.
A film soundtrack does, however, have an LFE channel. But it is intended to be played on home cinema hardware, where the crossover frequency is 120 Hz. So, the original soundtrack has some content between 120 and 700 Hz in main channels. PulseAudio will leave it there, because it refuses to remix LFE if it is already there. Result: tinny sound, because the subwoofer is essentially unused.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/348Volume changes can be delayed a lot2021-04-22T22:41:22ZBugzilla Migration UserVolume changes can be delayed a lot## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#48307)](https://bugs.freedesktop.org/show_bug.cgi?id=48307)**
## Description
Created attachment 59487
Log of delayed rewind
This was ...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#48307)](https://bugs.freedesktop.org/show_bug.cgi?id=48307)**
## Description
Created attachment 59487
Log of delayed rewind
This was reported in IRC by "uau". For him, sink volume changes happen with a large delay (like after a second). I have spotted one obvious bug in the source code: when the sink volume is changed, but the software volume doesn't need adjustment, the hardware volume is changed after whatever latency there happens to be at the moment of the volume change request, which tends to be large. If the software volume needs to be changed also, the sink is rewound, which will make the hw volume change happen earlier. So, the missing rewind is messing things up.
That is not the only problem, however. uau is having delays also when the rewind is happening. When I try changing the sink volume on my machine (with the current git master), the rewind happens before the volume change is applied, as expected. But for uau, the rewind appears to happen after the hw volume has been changed, i.e. about a second late. I haven't really investigated yet how that could be possible. uau is using Pulseaudio 1.1, the "current debian unstable version". Here's a log sample: http://fpaste.org/DVTW/
I'll also attach that log, in case the pastebin url becomes stale.
**Attachment 59487**, "Log of delayed rewind":
[log.txt](/uploads/0ab249bbe4a9e8250ebb91e2e812e52f/log.txt)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/352Recording a monitor when nothing is playing results in 2 seconds of extra sil...2018-07-30T10:20:03ZBugzilla Migration UserRecording a monitor when nothing is playing results in 2 seconds of extra silence## Submitted by Maarten Baert
Assigned to **pul..@..op.org**
**[Link to original bug (#66426)](https://bugs.freedesktop.org/show_bug.cgi?id=66426)**
## Description
When I record a monitor when nothing is playing, my application al...## Submitted by Maarten Baert
Assigned to **pul..@..op.org**
**[Link to original bug (#66426)](https://bugs.freedesktop.org/show_bug.cgi?id=66426)**
## Description
When I record a monitor when nothing is playing, my application always receives 2 seconds of silence during the first few milliseconds. I assume this is sound from the past, but I can't tell because it's just silence. This behaviour creates problems in my application, and I suppose in other applications as well.
My application deals with both audio and video, and the unexpected extra two seconds of audio are more than enough to desynchronize the audio and the video. I am now working around this by dropping all audio frames that I receive within the first 100ms, but this is clearly not ideal.
This does not happen when there is anything playing. It also doesn't happen when I record the microphone.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/355Rename pa_hashmap_remove() to pa_hashmap_steal()2018-07-30T10:20:15ZBugzilla Migration UserRename pa_hashmap_remove() to pa_hashmap_steal()## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#88828)](https://bugs.freedesktop.org/show_bug.cgi?id=88828)**
## Description
Some functions for removing items from a container use th...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#88828)](https://bugs.freedesktop.org/show_bug.cgi?id=88828)**
## Description
Some functions for removing items from a container use the word "remove", and some functions use the word "steal". Let's try to be consistent, and only use "remove" when the function also frees the removed item (if the free callback is set). So, pa_hashmap_remove() should be renamed to pa_hashmap_steal().https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/365Underlying h/w volumes are still adjusted due to flat volume logic even when ...2018-07-30T10:21:00ZBugzilla Migration UserUnderlying h/w volumes are still adjusted due to flat volume logic even when the stream is muted.## Submitted by Colin Guthrie
Assigned to **pul..@..op.org**
**[Link to original bug (#48284)](https://bugs.freedesktop.org/show_bug.cgi?id=48284)**
## Description
Reproduce:
(with flat volumes enabled)
1. Set sink volume to somet...## Submitted by Colin Guthrie
Assigned to **pul..@..op.org**
**[Link to original bug (#48284)](https://bugs.freedesktop.org/show_bug.cgi?id=48284)**
## Description
Reproduce:
(with flat volumes enabled)
1. Set sink volume to something sensible.
2. Set event sound volume to something less than that, e.g. half
3. Mute event sounds
4. Run alsamixer -c0
5. Trigger an event sound.
Watch as the volumes adjust due to flat volume logic!
This can be an issue as if you have event sounds set to 100% and muted, but sink volume to, say 60%, simply playing an event sound (which is muted) will push the h/w volume up to 100% and leave it there.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/368Rewinding when moving streams isn't done in an optimal way2018-07-30T10:21:08ZBugzilla Migration UserRewinding when moving streams isn't done in an optimal way## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#54182)](https://bugs.freedesktop.org/show_bug.cgi?id=54182)**
## Description
Code in the PA_SINK_MESSAGE_START_MOVE handler in sink.c:...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#54182)](https://bugs.freedesktop.org/show_bug.cgi?id=54182)**
## Description
Code in the PA_SINK_MESSAGE_START_MOVE handler in sink.c:
if (i->thread_info.state != PA_SINK_INPUT_CORKED) {
pa_usec_t usec = 0;
size_t sink_nbytes, total_nbytes;
/* The old sink probably has some audio from this
* stream in its buffer. We want to "take it back" as
* much as possible and play it to the new sink. We
* don't know at this point how much the old sink can
* rewind. We have to pick something, and that
* something is the full latency of the old sink here.
* So we rewind the stream buffer by the sink latency
* amount, which may be more than what we should
* rewind. This can result in a chunk of audio being
* played both to the old sink and the new sink.
*
* FIXME: Fix this code so that we don't have to make
* guesses about how much the sink will actually be
* able to rewind. If someone comes up with a solution
* for this, something to note is that the part of the
* latency that the old sink couldn't rewind should
* ideally be compensated after the stream has moved
* to the new sink by adding silence. The new sink
* most likely can't start playing the moved stream
* immediately, and that gap should be removed from
* the "compensation silence" (at least at the time of
* writing this, the move finish code will actually
* already take care of dropping the new sink's
* unrewindable latency, so taking into account the
* unrewindable latency of the old sink is the only
* problem).
*
* The render_memblockq contents are discarded,
* because when the sink changes, the format of the
* audio stored in the render_memblockq may change
* too, making the stored audio invalid. FIXME:
* However, the read and write indices are moved back
* the same amount, so if they are not the same now,
* they won't be the same after the rewind either. If
* the write index of the render_memblockq is ahead of
* the read index, then the render_memblockq will feed
* the new sink some silence first, which it shouldn't
* do. The write index should be flushed to be the
* same as the read index. */
/* Get the latency of the sink */
usec = pa_sink_get_latency_within_thread(s);
sink_nbytes = pa_usec_to_bytes(usec, &s->sample_spec);
total_nbytes = sink_nbytes + pa_memblockq_get_length(i->thread_info.render_memblockq);
if (total_nbytes > 0) {
i->thread_info.rewrite_nbytes = i->thread_info.resampler ? pa_resampler_request(i->thread_info.resampler, total_nbytes) : total_nbytes;
i->thread_info.rewrite_flush = TRUE;
pa_sink_input_process_rewind(i, sink_nbytes);
}
}
Code in the PA_SINK_MESSAGE_FINISH_MOVE handler in sink.c:
if (i->thread_info.state != PA_SINK_INPUT_CORKED) {
pa_usec_t usec = 0;
size_t nbytes;
/* In the ideal case the new sink would start playing
* the stream immediately. That requires the sink to
* be able to rewind all of its latency, which usually
* isn't possible, so there will probably be some gap
* before the moved stream becomes audible. We then
* have two possibilities: 1) start playing the stream
* from where it is now, or 2) drop the unrewindable
* latency of the sink from the stream. With option 1
* we won't lose any audio but the stream will have a
* pause. With option 2 we may lose some audio but the
* stream time will be somewhat in sync with the wall
* clock. Lennart seems to have chosen option 2 (one
* of the reasons might have been that option 1 is
* actually much harder to implement), so we drop the
* latency of the new sink from the moved stream and
* hope that the sink will undo most of that in the
* rewind. */
/* Get the latency of the sink */
usec = pa_sink_get_latency_within_thread(s);
nbytes = pa_usec_to_bytes(usec, &s->sample_spec);
if (nbytes > 0)
pa_sink_input_drop(i, nbytes);
pa_log_debug("Requesting rewind due to finished move");
pa_sink_request_rewind(s, nbytes);
}
So, there are two problems in the move start phase: the sink input is rewound more than it should be, and there may be unnecessary silence left in the render_memblockq.
It can be argued that the move finish phase is ok, but I think it would be better not to lose any audio.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/375Don't stop parsing configuration files when errors are detected2018-07-30T10:21:36ZBugzilla Migration UserDon't stop parsing configuration files when errors are detected## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#88953)](https://bugs.freedesktop.org/show_bug.cgi?id=88953)**
## Description
If there are errors in configuration files, pa_config_par...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#88953)](https://bugs.freedesktop.org/show_bug.cgi?id=88953)**
## Description
If there are errors in configuration files, pa_config_parser currently stops parsing immediately. It would be better to just log an error and continue parsing.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/384"Simultaneous output" is too low2018-07-30T10:22:33ZBugzilla Migration User"Simultaneous output" is too low## Submitted by Gabriel Corona
Assigned to **pul..@..op.org**
**[Link to original bug (#96922)](https://bugs.freedesktop.org/show_bug.cgi?id=96922)**
## Description
Created attachment 125060
pactl list
Hi,
I've updated Pulseaudi...## Submitted by Gabriel Corona
Assigned to **pul..@..op.org**
**[Link to original bug (#96922)](https://bugs.freedesktop.org/show_bug.cgi?id=96922)**
## Description
Created attachment 125060
pactl list
Hi,
I've updated Pulseaudio from 8.0-2+b2 to 9.0-1.1 (Debian packages) and with the new version the sound is way too low when using when using the "Simultaneous output". I have to put all the sliders at 150% as well as my speaker at near max to get some sound (and then I get a lot of noise).
It used to work fine before the update. Switching to "Builtin Audio Analog Stereo" gives me a much louder/normal audio output.
I attached the output of "pactl list".
Cheers.
**Attachment 125060**, "pactl list":
[pactl-list.txt](/uploads/87f0f8d2428c12e89f25720a4358d718/pactl-list.txt)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/386Make the UI handle translations2018-07-30T10:22:40ZBugzilla Migration UserMake the UI handle translations## Submitted by Bastien Nocera
Assigned to **pul..@..op.org**
**[Link to original bug (#66492)](https://bugs.freedesktop.org/show_bug.cgi?id=66492)**
## Description
Currently, when PulseAudio is running on a remote machine, or wit...## Submitted by Bastien Nocera
Assigned to **pul..@..op.org**
**[Link to original bug (#66492)](https://bugs.freedesktop.org/show_bug.cgi?id=66492)**
## Description
Currently, when PulseAudio is running on a remote machine, or with different envvars, the name of inputs, outputs, devices, etc. will be translated by the daemon and passed on translated to the UIs.
It would be nicer if the translations were handled in the PA library, so that running:
LC_ALL=fr_FR.UTF-8 gnome-control-center sound
would show all the translatable strings in the requested language, rather than a mish-mash of strings translated in 2 different languages.
Original request: https://bugzilla.gnome.org/show_bug.cgi?id=682866https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/393Mistreats headphone port with dysfunctional ALSA Master volume on ICH72018-07-30T10:23:34ZBugzilla Migration UserMistreats headphone port with dysfunctional ALSA Master volume on ICH7## Submitted by Tobias Wolf
Assigned to **Maarten Bosmans**
**[Link to original bug (#42353)](https://bugs.freedesktop.org/show_bug.cgi?id=42353)**
## Description
Created attachment 52868
ALSA info gzipped
I basically have a gene...## Submitted by Tobias Wolf
Assigned to **Maarten Bosmans**
**[Link to original bug (#42353)](https://bugs.freedesktop.org/show_bug.cgi?id=42353)**
## Description
Created attachment 52868
ALSA info gzipped
I basically have a generic Dell box with integrated ICH7 audio
0 [ICH7 ]: ICH4 - Intel ICH7
Intel ICH7 with AD1981B at irq 23
When I plug my headphones into the front, in alsamixer only the Headphone slider has an effect, Master does nothing to change volume.
Now Pulseaudio somehow merges these (even if flatvol is off) and the whole range of Headphone is compressed into the low range and it gets painful very quickly.
No idea if this is ALSA bug or not. Please advise and forward accordingly.
ALSA info and pulseaudio verbose output attached
~~**Attachment 52868**~~, "ALSA info gzipped":
[alsa-info.txt.6z8BY3PfQN.gz](/uploads/f5acbe19e00817ccf7040dfa9916a509/alsa-info.txt.6z8BY3PfQN.gz)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/395Single channel audio is really loud2018-07-30T10:23:55ZBugzilla Migration UserSingle channel audio is really loud## Submitted by Yuxuan Shui
Assigned to **pul..@..op.org**
**[Link to original bug (#88266)](https://bugs.freedesktop.org/show_bug.cgi?id=88266)**
## Description
When playing single channel audios, the resulting sound is way loude...## Submitted by Yuxuan Shui
Assigned to **pul..@..op.org**
**[Link to original bug (#88266)](https://bugs.freedesktop.org/show_bug.cgi?id=88266)**
## Description
When playing single channel audios, the resulting sound is way louder than playing dual channel audio with same volume settings.
If I convert the audio to dual channel before sending to pulseaudio, the result is normal.
If I create a channel map to map the channel to front left or front right, although this result in sound only play on one side (as expected), but the volume is correct.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/397LFE channel not synthesized properly2018-11-28T09:26:26ZBugzilla Migration UserLFE channel not synthesized properly## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#95021)](https://bugs.freedesktop.org/show_bug.cgi?id=95021)**
## Description
According to a report on the mailing list, upmixing a ste...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#95021)](https://bugs.freedesktop.org/show_bug.cgi?id=95021)**
## Description
According to a report on the mailing list, upmixing a stereo stream to 5.1 results in bass being removed from the audio. LFE remixing should move bass to the LFE channel, but reportedly that doesn't happen.
I'll mark this as a release blocker. I plan to try to reproduce the bug. If I can't reproduce, the blocker status can be removed, and maybe this bug can be closed too.
Here's the original report:
http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/25574/focus=25705https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/401pulseaudio crashed with SIGSEGV in set_profile_cb()2018-07-30T10:24:47ZBugzilla Migration Userpulseaudio crashed with SIGSEGV in set_profile_cb()## Submitted by Cristian Aravena
Assigned to **pul..@..op.org**
**[Link to original bug (#92107)](https://bugs.freedesktop.org/show_bug.cgi?id=92107)**
## Description
Open bug in launchpat.net :
https://bugs.launchpad.net/bugs/149...## Submitted by Cristian Aravena
Assigned to **pul..@..op.org**
**[Link to original bug (#92107)](https://bugs.freedesktop.org/show_bug.cgi?id=92107)**
## Description
Open bug in launchpat.net :
https://bugs.launchpad.net/bugs/1499113
"Connect two devices, celu movil samsung and headphone"
Package: pulseaudio 1:6.0-0ubuntu12
Backtrace:
```
#0 0x00007f1042baad12 in set_profile_cb (c=0x143ae90, new_profile=0x141f380) at modules/bluetooth/module-bluez5-device.c:1857
d = 0x1359d00
u = 0x143a680
p = 0x141f3b0
__func__ = "set_profile_cb"
__PRETTY_FUNCTION__ = "set_profile_cb"
#1 0x00007f1059232e06 in pa_card_set_profile (c=c@entry=0x143ae90, profile=profile@entry=0x141f380, save=save@entry=true) at pulsecore/card.c:281
r = <optimized out>
__func__ = "pa_card_set_profile"
__PRETTY_FUNCTION__ = "pa_card_set_profile"
#2 0x00007f10537acd51 in command_set_card_profile (pd=0x13db480, command=90, tag=15, t=0x141f120, userdata=0x1407830) at pulsecore/protocol-native.c:4907
c = 0x1407830
idx = 1
name = 0x0
profile_name = 0x13fc1b9 "a2dp_sink"
card = <optimized out>
profile = 0x141f380
ret = <optimized out>
__func__ = "command_set_card_profile"
__PRETTY_FUNCTION__ = "command_set_card_profile"
#3 0x00007f1058fcf88a in pa_pdispatch_run (pd=0x13db480, packet=packet@entry=0x13fc190, ancil_data=ancil_data@entry=0x13df3c0, userdata=userdata@entry=0x1407830) at pulsecore/pdispatch.c:341
cb = <optimized out>
tag = 15
command = 90
ts = 0x141f120
ret = -1
__func__ = "pa_pdispatch_run"
__PRETTY_FUNCTION__ = "pa_pdispatch_run"
#4 0x00007f10537b5ad5 in pstream_packet_callback (p=0x13df130, packet=0x13fc190, ancil_data=0x13df3c0, userdata=0x1407830) at pulsecore/protocol-native.c:5027
__func__ = "pstream_packet_callback"
__PRETTY_FUNCTION__ = "pstream_packet_callback"
#5 0x00007f1058fd23f2 in do_read (p=p@entry=0x13df130, re=re@entry=0x13df2b0) at pulsecore/pstream.c:880
d = <optimized out>
l = <optimized out>
r = <optimized out>
release_memblock = 0x0
__func__ = "do_read"
__PRETTY_FUNCTION__ = "do_read"
#6 0x00007f1058fd49c4 in do_pstream_read_write (p=0x13df130) at pulsecore/pstream.c:193
__func__ = "do_pstream_read_write"
__PRETTY_FUNCTION__ = "do_pstream_read_write"
#7 0x00007f1058d6ae87 in dispatch_pollfds (m=0x134b2a0) at pulse/mainloop.c:655
e = 0x144cf70
k = 1
r = 0
#8 pa_mainloop_dispatch (m=m@entry=0x134b2a0) at pulse/mainloop.c:898
dispatched = 0
__func__ = "pa_mainloop_dispatch"
__PRETTY_FUNCTION__ = "pa_mainloop_dispatch"
#9 0x00007f1058d6b28c in pa_mainloop_iterate (m=0x134b2a0, block=<optimized out>, retval=0x7ffd91e76b68) at pulse/mainloop.c:929
r = 1
#10 0x00007f1058d6b330 in pa_mainloop_run (m=m@entry=0x134b2a0, retval=retval@entry=0x7ffd91e76b68) at pulse/mainloop.c:944
r = <optimized out>
#11 0x0000000000406f49 in main (argc=<optimized out>, argv=<optimized out>) at daemon/main.c:1148
c = 0x134f930
buf = <optimized out>
conf = 0x13440e0
mainloop = 0x134b2a0
s = <optimized out>
configured_address = 0x13a8800 "\005"
r = <optimized out>
retval = 0
d = 3
ltdl_init = true
n_fds = <optimized out>
passed_fds = <optimized out>
e = <optimized out>
daemon_pipe = {-1, -1}
daemon_pipe2 = {-1, -1}
pid_monitor = 0x13594d0
autospawn_fd = -1
autospawn_locked = false
server_lookup = <optimized out>
lookup_service_bus = <optimized out>
server_bus = <optimized out>
start_server = <optimized out>
__func__ = "main"
__PRETTY_FUNCTION__ = "main"
```https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/416Adjusting volume speeds up audio2022-07-19T20:06:00ZBugzilla Migration UserAdjusting volume speeds up audio## Submitted by freedesktop38
Assigned to **pul..@..op.org**
**[Link to original bug (#101034)](https://bugs.freedesktop.org/show_bug.cgi?id=101034)**
## Description
1. Start a YouTube video in Firefox 53.
2. Open pavucontrol and ...## Submitted by freedesktop38
Assigned to **pul..@..op.org**
**[Link to original bug (#101034)](https://bugs.freedesktop.org/show_bug.cgi?id=101034)**
## Description
1. Start a YouTube video in Firefox 53.
2. Open pavucontrol and slide the output/master volume fader under the tab "Output Devices".
Result: No speed up of the audio or video
1. Start a YouTube video in Firefox 53.
2. Open pavucontrol and slide the volume fader of Firefox program under the "Playback" tab.
Result: Audio and video speed up and have glitches.
The same happens in Clementine and also if I use KDE's own audio configuration GUI under settings instead of pavucontrol.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/419PulseAudio crashes with Digigram VX222 card2018-07-30T10:26:18ZBugzilla Migration UserPulseAudio crashes with Digigram VX222 card## Submitted by Marcin Lewandowski
Assigned to **pul..@..op.org**
**[Link to original bug (#99062)](https://bugs.freedesktop.org/show_bug.cgi?id=99062)**
## Description
I cannot reproduce the exact behaviour but I have issues with...## Submitted by Marcin Lewandowski
Assigned to **pul..@..op.org**
**[Link to original bug (#99062)](https://bugs.freedesktop.org/show_bug.cgi?id=99062)**
## Description
I cannot reproduce the exact behaviour but I have issues with PulseAudio 8.0 shipped with ubuntu 16.04.
I was listening to music from Firefox, and then I started to hear only white noise.
In syslog I had plenty of
Dec 12 13:37:59 aksolotl pulseaudio[3292]: message repeated 15 times: [ [alsa-sink-VX PCM] alsa-sink.c: Failed to set hardware parameters: Zły argument]
Dec 12 13:37:59 aksolotl pulseaudio[3292]: [alsa-sink-VX PCM] alsa-sink.c: Failed to set hardware parameters: Zły argument
Dec 12 13:37:59 aksolotl kernel: [ 2758.081130] vx: cannot set different clock 48000 from the current 44100
Dec 12 13:37:59 aksolotl kernel: [ 2758.081174] vx: cannot set different clock 48000 from the current 44100
Dec 12 13:37:59 aksolotl kernel: [ 2758.081212] vx: cannot set different clock 48000 from the current 44100
Dec 12 13:37:59 aksolotl kernel: [ 2758.081250] vx: cannot set different clock 48000 from the current 44100https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/426analog-output-speaker not working anymore when headphone is plugged2018-07-30T10:26:47ZBugzilla Migration Useranalog-output-speaker not working anymore when headphone is plugged## Submitted by Knuth Posern
Assigned to **pul..@..op.org**
**[Link to original bug (#70865)](https://bugs.freedesktop.org/show_bug.cgi?id=70865)**
## Description
HP desktop pc with audio line-out / headphone jack in front and rea...## Submitted by Knuth Posern
Assigned to **pul..@..op.org**
**[Link to original bug (#70865)](https://bugs.freedesktop.org/show_bug.cgi?id=70865)**
## Description
HP desktop pc with audio line-out / headphone jack in front and rear AND a built-in mono speaker.
Hardware: 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 02)
Debian wheezy stable system with pulseaudio 2.0-6.1
With alsa there is a ""Simple mixer control 'Independent HP'" if set to
* "Enable", sound is /only/ audible through the built-in speaker
* "Disable", sound is /only/ audible through the connected headphone/line-out
With pulseaudio (thanks a lot to poljar #pulseaudio IRC !):
(1.) If I unplug any line-out audio cable aka headphone cables, then
* pactl set-sink-port 0 analog-output-headphones
Makes sound only be audible through the built-in internal mono speaker
* pactl set-sink-port 0 analog-output-speaker
Makes no sound audible (which seems to be exactly how it should be :)
(2.) If I plug an headphone cable either rear or front, then:
* pactl set-sink-port 0 analog-output-headphones
Makes sound /only/ be audible through the plugged headphone/line-out
BUT:
* pactl set-sink-port 0 analog-output-speaker
Makes sound /only/ be audible through the plugged headphone/line-out TOO
`<poljar>` statement: that sounds wrong, analog-output-speaker should either tagged as not available if the hardware prevents it to output to the speaker while something is connected or it should work, so feel free to report a bug
I created for both cases (1.) and (2.) the pulseaudio-output and the client-output. I hope you find them useful. Please let me know if you need anything else.
The system is a headless server system.
pulseaudio is running as user pulse
To get the pulseaudio-output it ran like this (debian default with changed --log-target and added -vvvvvv):
/usr/bin/pulseaudio -vvvvvv --system --disallow-exit --daemonize --log-target=file:/tmp/pulseaudio.log --high-priority --disallow-module-loading=1
So unter pulse I did not find any way yet to use the built-in speaker for sound-output when I have a headphone/line-out plugged.
Even though under alsa I am able to toggle the output to come through the built-in speaker and with or without a headphone/line-out plugged!
This is bugging me, because I have my stereo plugged all the time to the system, but it ONLY gets turned on, when I need it. So important system notifications should be played through speaker.
Please help :)
Thanks a lot in advance !
Tormen.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/430Cannot use internal Mic on Lenovo T5002018-07-30T10:27:04ZBugzilla Migration UserCannot use internal Mic on Lenovo T500## Submitted by Russel Winder
Assigned to **pul..@..op.org**
**[Link to original bug (#39982)](https://bugs.freedesktop.org/show_bug.cgi?id=39982)**
## Description
The PulseAudio Sound Preferences appears to give no way of using t...## Submitted by Russel Winder
Assigned to **pul..@..op.org**
**[Link to original bug (#39982)](https://bugs.freedesktop.org/show_bug.cgi?id=39982)**
## Description
The PulseAudio Sound Preferences appears to give no way of using the internal microphone on a Lenovo T500 running Debian Testing. The Input volume control appears to be for the mic input socket, there is no control for the internal mic. Using the Gnome ALSA Mixer there are extra controls over and above what the PulseAudio Sound Preferences offers. In particular there is a control Internal, which is required to set the input level of the internal microphone -- the power on default is zero which means the microphone is not providing source signal.
It isn't actually clear where the problem resides, it may or may not be a direct PulseAudio issue, but I was advised to submit a report here anyway since PulseAudio is the default sound system control and it should be able to control all the sound sources.
In a sense this is a blocking bug, but there is a simple workaround so it is not actually blocking.
I appreciate this may not be a suitably complete bug report, but I can provide any appropriate information needed as required.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/436sound is choppy and high-pitched after suspend+resume2018-07-30T10:27:46ZBugzilla Migration Usersound is choppy and high-pitched after suspend+resume## Submitted by Adam Borowski
Assigned to **pul..@..op.org**
**[Link to original bug (#84667)](https://bugs.freedesktop.org/show_bug.cgi?id=84667)**
## Description
Once my computer is suspended then resumed, pulseaudio makes all s...## Submitted by Adam Borowski
Assigned to **pul..@..op.org**
**[Link to original bug (#84667)](https://bugs.freedesktop.org/show_bug.cgi?id=84667)**
## Description
Once my computer is suspended then resumed, pulseaudio makes all sound
high-pitched and choppy. It's hard to diagnose this by ear, but it
-appears- to me that a few times per second a buffer is played at a
multiple of proper speed, with a pause until the buffer fills again.
Issuing 'killall pulseaudio' and restarting clients solves this issue
until next resume.
This worked correctly with the same sound card with a 5.1 speaker setup,
the problem started only after I replaced them by regular two speakers.
[this is forwarded https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=763985, that bug report includes some debug info which I did not copy here to reduce spam]https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/437USB Headset not set up correctly after resume2018-07-30T10:27:49ZBugzilla Migration UserUSB Headset not set up correctly after resume## Submitted by Marcus Moeller
Assigned to **pul..@..op.org**
**[Link to original bug (#80712)](https://bugs.freedesktop.org/show_bug.cgi?id=80712)**
## Description
Microsoft LifeChat LX-6000 is not set up correctly after resume. ...## Submitted by Marcus Moeller
Assigned to **pul..@..op.org**
**[Link to original bug (#80712)](https://bugs.freedesktop.org/show_bug.cgi?id=80712)**
## Description
Microsoft LifeChat LX-6000 is not set up correctly after resume. If I plug the headset, the device is listed in pavucontrol, but no sink/source is created.
A similar topic has been described on the mailinglist a while back: http://lists.freedesktop.org/archives/pulseaudio-discuss/2012-February/012907.htmlhttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/438Spurious creation of runtime directory by pa_context_connect()2018-07-30T10:27:53ZBugzilla Migration UserSpurious creation of runtime directory by pa_context_connect()## Submitted by hor..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#84669)](https://bugs.freedesktop.org/show_bug.cgi?id=84669)**
## Description
Under some conditions, pa_context_connect(), during the process o...## Submitted by hor..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#84669)](https://bugs.freedesktop.org/show_bug.cgi?id=84669)**
## Description
Under some conditions, pa_context_connect(), during the process of PA server discovery, winds up creating an empty runtime directory, even if 'autospawn' is disabled. This seems like it is probably a mistake.
Further details in this thread:
http://lists.freedesktop.org/archives/pulseaudio-discuss/2014-October/021889.htmlhttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/439Audio skips and repeats when increasing volume in applications2018-07-30T10:27:55ZBugzilla Migration UserAudio skips and repeats when increasing volume in applications## Submitted by Yomi
Assigned to **pul..@..op.org**
**[Link to original bug (#86676)](https://bugs.freedesktop.org/show_bug.cgi?id=86676)**
## Description
Whenever I increase the volume in any application sometimes the audio skips...## Submitted by Yomi
Assigned to **pul..@..op.org**
**[Link to original bug (#86676)](https://bugs.freedesktop.org/show_bug.cgi?id=86676)**
## Description
Whenever I increase the volume in any application sometimes the audio skips and/or repeats until the volume reaches the new percentage. Interfaces related to volume end up ignoring input until the new volume is reached. Any video playing at the time will also pause until the new volume is reached.
It's intermittent. There doesn't seem to be a way for me to reliably reproduce it.
Output from pulseaudio -vvvv is here : https://gist.github.com/Yomi0/91e3fc04d9744a790020
Is there any more information I need to provide?
Arch Linux 3.17.3-1-ARCH
Name : pulseaudio
Version : 5.0-1
Build Date : Mon 03 Mar 2014 03:31:31 PM EST
Install Date : Sun 24 Aug 2014 04:46:54 PM EDT
Name : alsa-lib
Version : 1.0.28-1
Build Date : Wed 18 Jun 2014 02:00:01 AM EDT
Install Date : Fri 22 Aug 2014 03:22:22 PM EDT
Name : alsa-plugins
Version : 1.0.28-2
Build Date : Mon 15 Sep 2014 02:54:49 PM EDT
Install Date : Tue 23 Sep 2014 01:12:48 PM EDT
Name : alsa-utils
Version : 1.0.28-1
Build Date : Wed 18 Jun 2014 02:08:32 AM EDT
Install Date : Sun 24 Aug 2014 04:44:00 PM EDThttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/443Server crashes if a client tells it to update stream rate when the stream doe...2018-07-30T10:28:10ZBugzilla Migration UserServer crashes if a client tells it to update stream rate when the stream doesn't support that## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#83591)](https://bugs.freedesktop.org/show_bug.cgi?id=83591)**
## Description
I haven't tried this in practice, but it seems that the s...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#83591)](https://bugs.freedesktop.org/show_bug.cgi?id=83591)**
## Description
I haven't tried this in practice, but it seems that the server doesn't check whether the sink input's or source output's resampler supports rate changes when a client sends a request to change the stream's rate. When using libpulse, invalid rate changes are filtered at client side, but a malicious client can talk to the server without using libpulse.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/458VERY loud hiss from speakers2018-07-30T10:29:18ZBugzilla Migration UserVERY loud hiss from speakers## Submitted by John Schmitt
Assigned to **Maarten Bosmans**
**[Link to original bug (#42351)](https://bugs.freedesktop.org/show_bug.cgi?id=42351)**
## Description
I get a frequent very, very, very loud hiss from mythfrontend when...## Submitted by John Schmitt
Assigned to **Maarten Bosmans**
**[Link to original bug (#42351)](https://bugs.freedesktop.org/show_bug.cgi?id=42351)**
## Description
I get a frequent very, very, very loud hiss from mythfrontend when any of the
following occur:
1. fast-forward
2. rewind
3. commercial skip
I can fix it but pressing fast-forward or rewind, sometimes 2 or 3 times, but
the next time there's a commercial skip, the hiss will be back.
This is a recent phenomenon (started happening about 2 weeks ago) that I cannot
correlate with an exact upgrade.
Version-Release number of selected component (if applicable):
0.9.22-5.fc15
How reproducible:
intermittent but often
Steps to Reproduce:
1. Play back a recording with mythfrontend
2. Press the right or left arrow
Actual results:
After a commercial skip or after the first or second or third press, a very
loud hiss will replace the audio that should accompany the recording.
In the debug output attached, the "Pool Full" and "events suppressed" messages
only roughly correlate in time to the noise. There doesn't seem to be an exact
message from the output of pulseaudio -vvv that can be correlated exactly to
the noise.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/460PulseAudio gets reliably killed upon a big number of client connections2018-07-30T10:29:26ZBugzilla Migration UserPulseAudio gets reliably killed upon a big number of client connections## Submitted by Ahmed S. Darwish
Assigned to **pul..@..op.org**
**[Link to original bug (#94629)](https://bugs.freedesktop.org/show_bug.cgi?id=94629)**
## Description
On master branch [1], connections from a high number of clients...## Submitted by Ahmed S. Darwish
Assigned to **pul..@..op.org**
**[Link to original bug (#94629)](https://bugs.freedesktop.org/show_bug.cgi?id=94629)**
## Description
On master branch [1], connections from a high number of clients
_reliably_ kills the PulseAudio daemon.
Here is a minimal script that triggers the bug:
LARGE_WAVE_FILE=...
for i in {1..60}; do
echo "Client #$i";
./src/pacat $LARGE_WAVE_FILE &
sleep 1
done
Here are some important factors:
1. The bug is always triggered in regular -O2 build, ALSA sink.
This usually happens after client #45. [2] [3]
2. Disabling SHM (--disable-shm) makes triggering the bug much
quicker. Only after connection from client #27
3. Compiling at -O0 also makes the bug triggered much quicker.
Only after connection from client #28
4. The bug _disappears_ when choosing the NULL sink as default
This is the case even after leaving the pacat clients run
for an hour
5. In the point #4 above, resetting ALSA sink back as the
default sink triggers the bug again.
6. This bug affects all versions of PulseAudiob back to v5.0!
I could not test older versions (v4.0, v3.0, ..) as they
always fail at runtime with my current toolchain. [4] [5]
Any thoughts on how to track this issue further?
Thanks,
==> footnotes:
[1] As of 19 March 2016, 4731690a21edc59acfd0bd27f810d5c895ac7629
[2] No logs are produced from the Linux kernel in this case.
Check Arun's work at http://goo.gl/0mq3ym for context.
[3] This is an old "AMD Athlon(tm) II X2 260" dual-core desktop
machine
[4] Arch Linux, GCC 5.3.0 with target x86_64-unknown-linux-gnu,
glibc version 2.23
[5] Runtime failure for v4.0 is:
hashmap.c: Assertion 'h->iterate_list_head' failed at
pulsecore/hashmap.c:151, function pa_hashmap_put(). Aborting.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/462PulseAudio does not set up all outputs on a soundcard as separate output devi...2018-07-30T10:29:35ZBugzilla Migration UserPulseAudio does not set up all outputs on a soundcard as separate output devices by default## Submitted by N. W.
Assigned to **pul..@..op.org**
**[Link to original bug (#96272)](https://bugs.freedesktop.org/show_bug.cgi?id=96272)**
## Description
Hi,
PulseAudio unfortunately does not set up all outputs on a soundcard a...## Submitted by N. W.
Assigned to **pul..@..op.org**
**[Link to original bug (#96272)](https://bugs.freedesktop.org/show_bug.cgi?id=96272)**
## Description
Hi,
PulseAudio unfortunately does not set up all outputs on a soundcard as separate output devices by default.
By default, each soundcard is only listed as one output device in the pavucontrol "Output Devices" tab, even if the soundcard has more than one output.
So, you have to switch between each output via the "Configuration" tab in pavucontrol, as shown in the following screenshot for example:
https://www.maketecheasier.com/assets/uploads/2010/12/hdmi-pavucontrol.png
And it is not possible to tell one app to use the "Analog Stereo Output" and another app to use the "Digital Stereo (HDMI) Output".
Why aren't "Analog Stereo Output" and "Digital Stereo (HDMI) Output" listed as separate output devices on the "Output Devices" tab in pavucontrol by default?
As far as I understand, the user has to add each output himself via:
load-module module-alsa-sink device=hw:x,x
?
But why?
Why isn't this being done by default?
And could you please change it?
Regardshttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/466pulseaudio[19858]: [alsa-sink] memblock.c: Assertion 'pa_atomic_load(&(b)->_r...2018-07-30T10:30:02ZBugzilla Migration Userpulseaudio[19858]: [alsa-sink] memblock.c: Assertion 'pa_atomic_load(&(b)->_ref) > 0' failed at pulsecore/memblock.c:590, function pa_memblock_unref(). Aborting.## Submitted by John Schmitt
Assigned to **pul..@..op.org**
**[Link to original bug (#59115)](https://bugs.freedesktop.org/show_bug.cgi?id=59115)**
## Description
Created attachment 72663
pulseaudio extracted from /var/log/message...## Submitted by John Schmitt
Assigned to **pul..@..op.org**
**[Link to original bug (#59115)](https://bugs.freedesktop.org/show_bug.cgi?id=59115)**
## Description
Created attachment 72663
pulseaudio extracted from /var/log/messages for the time that pulseaudio ran with log-level=4
About once a day pulseaudio crashes with that message. The machine is usually idle when this happens except for skype.
$ rpm -qa | grep -i pulse
wine-pulseaudio-1.5.13-1.fc17.x86_64
pulseaudio-module-gconf-1.1-9.fc17.x86_64
pulseaudio-1.1-9.fc17.x86_64
pulseaudio-libs-1.1-9.fc17.i686
wine-pulseaudio-1.5.13-1.fc17.i686
alsa-plugins-pulseaudio-1.0.26-1.fc17.x86_64
kde-settings-pulseaudio-4.8-19.fc17.noarch
gvncpulse-0.5.1-4.fc17.x86_64
pulseaudio-libs-glib2-1.1-9.fc17.x86_64
pulseaudio-module-x11-1.1-9.fc17.x86_64
pulseaudio-module-zeroconf-1.1-9.fc17.x86_64
pulseaudio-libs-1.1-9.fc17.x86_64
pulseaudio-utils-1.1-9.fc17.x86_64
Steps to Reproduce:
Install skype leave it running
use a USB headset (Logitch G35) as well as speakers on-board sound
run xfce4-mixer and pavucontrol, leave them running
Actual results:
pulseaudio[19858]: [alsa-sink] memblock.c: Assertion 'pa_atomic_load(&(b)->_ref) > 0' failed at pulsecore/memblock.c:590, function pa_memblock_unref(). Aborting.
$ cat ~/.pulse/daemon.conf
log-level = 4
; default-fragment-size-msec = 1
; default-sample-rate = 48000
; log-target = auto
; log-level = notice
; log-meta = no
; log-time = no
; log-backtrace = 0
**Attachment 72663**, "pulseaudio extracted from /var/log/messages for the time that pulseaudio ran with log-level=4":
[pulse.out](/uploads/387daa6d79da1785b71dc4e0361cf12f/pulse.out)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/469[cleanup] When lowering sink latency, the core could request a rewind2018-07-30T10:30:11ZBugzilla Migration User[cleanup] When lowering sink latency, the core could request a rewind## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#54006)](https://bugs.freedesktop.org/show_bug.cgi?id=54006)**
## Description
If a sink supports dynamic latency and the latency is adj...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#54006)](https://bugs.freedesktop.org/show_bug.cgi?id=54006)**
## Description
If a sink supports dynamic latency and the latency is adjusted downwards, there needs to be a rewind to ensure that the sink buffer doesn't contain more data than what is allowed by the new latency. This is because rewind requests can't be larger than the configured latency, and if there's more data than that in the buffer, a later rewind request may end up being too small.
The responsibility for issuing the rewind request is currently on the sink implementor. I don't think that's necessary, so I think it would be better to issue the rewind request in the core code.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/471Possible memory leak in memtrap.c2018-07-30T10:30:16ZBugzilla Migration UserPossible memory leak in memtrap.c## Submitted by Sachin Kumar Chauhan
Assigned to **pul..@..op.org**
**[Link to original bug (#95348)](https://bugs.freedesktop.org/show_bug.cgi?id=95348)**
## Description
In src\pulsecore\memtrap.c, a pointer aupdate is allocated ...## Submitted by Sachin Kumar Chauhan
Assigned to **pul..@..op.org**
**[Link to original bug (#95348)](https://bugs.freedesktop.org/show_bug.cgi?id=95348)**
## Description
In src\pulsecore\memtrap.c, a pointer aupdate is allocated memory using pa_aupdate_new() but pa_aupdate_free() is never called to free the memory.
Is the alloted memory being leaked ?https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/482[alsa-sink] memblockq.c: Assertion '!bq->blocks || (bq->...2018-07-30T10:31:03ZBugzilla Migration User[alsa-sink] memblockq.c: Assertion '!bq->blocks || (bq->write_index + (int64_t)chunk.length <= bq->blocks->index)' failed at pulsecore/memblockq.c:408, function pa_memblockq_push(). Aborting.## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#45373)](https://bugs.freedesktop.org/show_bug.cgi?id=45373)**
## Description
Created attachment 56298
Backtrace
Reported in IRC:
19:...## Submitted by Tanu Kaskinen `@tanuk`
Assigned to **pul..@..op.org**
**[Link to original bug (#45373)](https://bugs.freedesktop.org/show_bug.cgi?id=45373)**
## Description
Created attachment 56298
Backtrace
Reported in IRC:
19:28 -!- jelly-home [~jelly@homestuck.iskon.hr] has joined #pulseaudio
19:30 `< jelly-home>` Hello, I've caught pulseaudio killing itself with a SIGABRT and "Jan 29 15:44:43 kanta-linux pulseaudio[11171]: [alsa-sink] memblockq.c: Assertion '!bq->blocks ||
(bq->write_index + (int64_t)chunk.length `<= bq->`blocks->index)' failed at pulsecore/memblockq.c:408, function pa_memblockq_push(). Aborting."
19:30 `< jelly-home>` attempt at getting a backtrace: http://paste.debian.net/153989/
19:30 <@pulsator> Title: debian Pastezone (at paste.debian.net)
19:34 `< jelly-home>` this is PA 1.1, package 1.1-2 from Debian testing, i386 arch
I'll attach the backtrace here in case the pastebin link gets stale.
**Attachment 56298**, "Backtrace":
[backtrace.txt](/uploads/3867610ee8210c3a0bf32f4f81417ba1/backtrace.txt)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/483Implement opus audio compression2018-07-30T10:31:17ZBugzilla Migration UserImplement opus audio compression## Submitted by Jonas Heinrich
Assigned to **pul..@..op.org**
**[Link to original bug (#56993)](https://bugs.freedesktop.org/show_bug.cgi?id=56993)**
## Description
Hello,
I would like to see audio compression for streaming audio ...## Submitted by Jonas Heinrich
Assigned to **pul..@..op.org**
**[Link to original bug (#56993)](https://bugs.freedesktop.org/show_bug.cgi?id=56993)**
## Description
Hello,
I would like to see audio compression for streaming audio in the network. Especially the new codec Opus could be usefull, because it's very fast (low-latency) and efficent (good sound quality with low bitrates).
Such a feature would allow me to stream audio via wifi, what currently doesn't work with a normal tcp.sink.
Best regards,
Jonas