pulseaudio issueshttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues2019-04-26T10:59:04Zhttps://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/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/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/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/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/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/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/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/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/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/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/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/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/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/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.