pulseaudio issueshttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues2020-04-06T22:55:29Zhttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/532PulseAudio does not react properly to hardware changing its sample rate2020-04-06T22:55:29ZBugzilla Migration UserPulseAudio does not react properly to hardware changing its sample rate## Submitted by Adam Goode
Assigned to **pul..@..op.org**
**[Link to original bug (#107249)](https://bugs.freedesktop.org/show_bug.cgi?id=107249)**
## Description
I have a MOTU Ultralite AVB audio interface, connected via USB. The...## Submitted by Adam Goode
Assigned to **pul..@..op.org**
**[Link to original bug (#107249)](https://bugs.freedesktop.org/show_bug.cgi?id=107249)**
## Description
I have a MOTU Ultralite AVB audio interface, connected via USB. The main control interface is via an embedded web server, which allows me to set routing, clock source, and sample rate. If I change the sample rate, pulseaudio does not detect this change and audio comes out distorted.
This happens when I change the rate from 44100 to 48000 to allow the clock to sync to an external optical source running at 48000.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/19318.04 Headphones detected, but not switched on automatically after startup2021-10-29T16:23:35ZBugzilla Migration User18.04 Headphones detected, but not switched on automatically after startup## Submitted by tom..@..er.com
Assigned to **pul..@..op.org**
**[Link to original bug (#106834)](https://bugs.freedesktop.org/show_bug.cgi?id=106834)**
## Description
Been this way since at least Ubuntu 15.
Plug in your headphone...## Submitted by tom..@..er.com
Assigned to **pul..@..op.org**
**[Link to original bug (#106834)](https://bugs.freedesktop.org/show_bug.cgi?id=106834)**
## Description
Been this way since at least Ubuntu 15.
Plug in your headphones, boot up your laptop. No sound. Speakers and headphones are disabled. Super annoying, you have to unplug and re-plug, and go through the stupid dialogue of "what are you trying to plug in?" which has no memory, and no defaults.
### See also
* https://launchpad.net/bugs/1583801Russell TreleavenRussell Treleavenhttps://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/517pulseaudio-module-gconf auto-selects incorrect Profile when plugging headphones2018-07-30T10:34:59ZBugzilla Migration Userpulseaudio-module-gconf auto-selects incorrect Profile when plugging headphones## Submitted by Geroge T
Assigned to **pul..@..op.org**
**[Link to original bug (#106429)](https://bugs.freedesktop.org/show_bug.cgi?id=106429)**
## Description
Source:
https://bugs.kde.org/show_bug.cgi?id=393963
Disconnecting an...## Submitted by Geroge T
Assigned to **pul..@..op.org**
**[Link to original bug (#106429)](https://bugs.freedesktop.org/show_bug.cgi?id=106429)**
## Description
Source:
https://bugs.kde.org/show_bug.cgi?id=393963
Disconnecting and reconnecting the headphones, automatically switches between these two:
- "Analog Surround 4.0 Output" when disconnecting the headphones, which makes the speakers work.
- "Analog Stereo Output" when plugging the headphones to the computer jack port, BUT there is no sound on them because it should select "Digital Stereo (IEC958) Output" instead.
Running pulseaudio-module-gconf-11.1-18.fc28.x86_64https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/423Linux Mint No Sound HDMI Output2018-07-30T10:26:33ZBugzilla Migration UserLinux Mint No Sound HDMI Output## Submitted by Pavel Komarov
Assigned to **pul..@..op.org**
**[Link to original bug (#106361)](https://bugs.freedesktop.org/show_bug.cgi?id=106361)**
## Description
```
* Cinnamon version 3.6.7
* Distribution - Mint 17.3
* Int...## Submitted by Pavel Komarov
Assigned to **pul..@..op.org**
**[Link to original bug (#106361)](https://bugs.freedesktop.org/show_bug.cgi?id=106361)**
## Description
```
* Cinnamon version 3.6.7
* Distribution - Mint 17.3
* Intel HD Graphics 5500 with Intel Corporation Broadwell-U Integrated Graphics (rev 09) (prog-if 00 [VGA controller])
* 64 bit
```
**Issue**
I have been using Linux Mint since 2014 across four different computers and have never in all this time been able to output sound over HDMI. There are numerous issues on the linux mint forums about it (https://forums.linuxmint.com/viewtopic.php?t=242328, for example), but none have ever provided a solution that has worked for me.
**Steps to reproduce**
Install Linux Mint, plug your computer in to a TV via HDMI, and try to play a video. The picture will appear just fine, but audio will continue to come from the laptop.
**Expected behaviour**
Both audio and video should be output to the TV.
**Other information**
I've heard whispers of this occurring with other distros, though a friend who uses Ubuntu claims he has not had this problem.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/150Focusrite Saffire 26/IO Boot/Reset loop2018-07-30T09:59:11ZBugzilla Migration UserFocusrite Saffire 26/IO Boot/Reset loop## Submitted by Ramon Fried
Assigned to **pul..@..op.org**
**[Link to original bug (#105921)](https://bugs.freedesktop.org/show_bug.cgi?id=105921)**
## Description
Created attachment 138648
dmesg
Focusrite Saffire 26/IO, Get's to...## Submitted by Ramon Fried
Assigned to **pul..@..op.org**
**[Link to original bug (#105921)](https://bugs.freedesktop.org/show_bug.cgi?id=105921)**
## Description
Created attachment 138648
dmesg
Focusrite Saffire 26/IO, Get's to constant boot/reset loop.
Logs attached.
**Attachment 138648**, "dmesg":
[file_105921.txt](/uploads/fe12e1383d4bb5dc25f56c06c5bfc8cb/file_105921.txt)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/335Fast volume change (using keyboard key) causes volume difference between chan...2018-07-30T10:18:53ZBugzilla Migration UserFast volume change (using keyboard key) causes volume difference between channels## Submitted by Hilary Jendrasiak
Assigned to **pul..@..op.org**
**[Link to original bug (#105714)](https://bugs.freedesktop.org/show_bug.cgi?id=105714)**
## Description
Created attachment 138315
Screen capture with `pavucontrol` ...## Submitted by Hilary Jendrasiak
Assigned to **pul..@..op.org**
**[Link to original bug (#105714)](https://bugs.freedesktop.org/show_bug.cgi?id=105714)**
## Description
Created attachment 138315
Screen capture with `pavucontrol` and `screenkey`.
I've custom keybind in
/etc/xdg/openbox/rc.xml
It is:
`<keybind key="XF86AudioLowerVolume">`
`<action name="Execute">`
`<startupnotify>`
`<enabled>`true`</enabled>`
`<name>`LowerVolume`</name>`
`</startupnotify>`
`<command>`amixer -D pulse sset Master 1%-`</command>`
`</action>`
`</keybind>`
(Analogical 1%+)
In my keyboard the XF86AudioLowerVolume can be triggered by rotating keyboard's volume control winder. And the winder is ok.
As long as I'm rotating this winder slow – everything is ok. The problem occurs when I'm sending more than c.a. 10 XF86AudioLowerVolume "presses" per second. (That's not so many – less than 180° rotation! That winder is standard "infinite" rotation winder. Nothing special.)
I've prepared simple bash script that can simulate spinning of winder volume control. For me it's working just as bad as fast winder rotation.
https://pastebin.com/hR4UCjjJ
(Take care about that I've muted STDOUT and STDERR in loop)
**Attachment 138315**, "Screen capture with `pavucontrol` and `screenkey`.":
![2018-03-23_14-53-19](/uploads/2ee7c309fc4317d5d8ef76ae941052fc/2018-03-23_14-53-19.mp4)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/256Unable to properly switch between Speakers and Headphones (has "workaround")2021-06-03T14:54:41ZBugzilla Migration UserUnable to properly switch between Speakers and Headphones (has "workaround")## Submitted by gbr..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#105654)](https://bugs.freedesktop.org/show_bug.cgi?id=105654)**
## Description
Created attachment 138239
pactl list sinks
===
OS: Arch Linux ...## Submitted by gbr..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#105654)](https://bugs.freedesktop.org/show_bug.cgi?id=105654)**
## Description
Created attachment 138239
pactl list sinks
===
OS: Arch Linux x64
Kernel: 4.15.10
PulseAudio 11.1
Motherboard: B75M-DGS R2.0 (5.1 CH HD Audio (Realtek ALC662 Audio Codec)) [1]
===
Hi,
I was having trouble making PulseAudio switch the audio output between my headphones and speakers, and most importantly, play audio in them exclusively. My headphone is plugged into the front panel and my speakers are using the rear green connection, and both of them are 2 channel. I was able to find a workaround, which I will demonstrate later, but as of now I'll write like I didn't know of that workaround:
I'm using both ALSA and PulseAudio's default options (although I'm not sure if Arch Linux changes any configuration file).
I have two sinks (see attachment for the output of "pactl list sinks"), but the relevant one is the latter, which holds my headphones and speakers (they're split into two ports).
I'll do my best to reproduce the issue I'm having. I apologize in advance bacause English is not my first language. So, for now let's assume I have only my speakers plugged in (a port identified as "Line Out" in pavucontrol). When I decide to plug in my headphones, here's what happens in pavucontrol:
1. The "Line Out" port is set to unplugged and outputs no audio anymore
2. The "Headphones" port is set to plugged in and successfully outputs audio
The only problem I see here is that the "Line Out" port was set to "unplugged" (this is about semantics, but it really wasn't phisically unplugged), but no big deal since I'm getting audio exclusively in my headphones, which is exactly what I wanted.
Okay, now that I have both speakers and headphones plugged in, let's say I want to switch back to my speakers. All I would have to do is change the "Port" to "Line Out (unplugged)", but here's what really happens:
1. The "Headphones" port gets muted (ok)
2. The "Line Out" port DOES NOT get unmuted
3. I get no audio output in both of them
Strangely, the "Headphones" port WAS NOT set to "unplugged" like it happened before with "Line Out".
---
WORKAROUND:
Here's the "workaround" (not sure if you can call it that) I found while messing with alsamixer:
Assuming we're still are in the previous scenario where I had no audio output, if I open "alsamixer -c0" and set the "Auto-Mute Mode" control to "Disabled", I get exclusive audio output in my speakers (which is perfect and exactly what I wanted). Now, let's say I'd switch back to my headphones. The audio does output to my headphones, but not exclusively. So I still hear audio coming from my speakers as well (I don't want that). That's it.
Speaking of that "Auto-Mute Mode" control, the documentation for Linux v4.5 states [2]:
>When both headphone and line-out jacks are present, it gives "Disabled", "Speaker Only" and "Line-Out+Speaker".
Well, I do have both of them present, but I the only options I get are: "Disabled" and "Enabled". So there's that as well...
---
I will be honest to you guys, I've been searching for a solution to this problem for a long time. At the same time, I'm not an expert (not even close to that) about Linux / PulseAudio / ALSA, so there's a high possibility that I might have missed some PulseAudio configuration (I'm really sorry if that's the case).
But based on a lot of Google searches [3][4][5][6], I'm definitely not the only person dealing with that issue, and while I found some possible solutions, none of them really worked for me... There's even a guy saying he has given up [5]. That "workaround" I found was by accident, while I was fiddling with alsamixer. I mean, it does work perfectly for me, but it's very inconvenient having to disable "Auto-Mute Mode" in alsamixer when I want to use my speakers, and then enable it again when I want to use my headphones. If there isn't a solution within PulseAudio or if this is not a bug at all, let me know. I'm sure this could be automated with scripts, but I'm not a programmer, so any help on would be appreciated as well.
Thank you very much.
TL;DR: I wanted to switch between speakers and headphones and output the audio in them EXCLUSIVELY, like you do on Windows [7], but I couldn't do it by default; But I've found a workaround (which is a tad inconvenient when you do it by hand); And I was not sure if this was a bug or not, so I decided to report here just in case.
===
REFERENCES:
[1] http://www.asrock.com/mb/Intel/B75M-DGS%20R2.0/
[2] https://github.com/torvalds/linux/blob/v4.15/Documentation/sound/hd-audio/controls.rst#realtek-codecs
[3] https://wiki.archlinux.org/index.php/PulseAudio/Examples#Having_both_speakers_and_headphones_plugged_in_and_switching_in_software_on-the-fly (this was the most promising one, unfortunately it didn't work)
[4] https://bbs.archlinux.org/viewtopic.php?id=187828
[5] https://bbs.archlinux.org/viewtopic.php?id=226651
[6] https://bbs.archlinux.org/viewtopic.php?id=161296
[7] https://i.imgur.com/gfKkQcS.png
**Attachment 138239**, "pactl list sinks":
[pactl_list_sinks.txt](/uploads/bc0ce2315f2dba42c16b65e16a552387/pactl_list_sinks.txt)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/61Pulseaudio delays sound compared to activity on screen when using sound via HDMI2018-07-30T10:17:52ZBugzilla Migration UserPulseaudio delays sound compared to activity on screen when using sound via HDMI## Submitted by Freek de Kruijf
Assigned to **pul..@..op.org**
**[Link to original bug (#104501)](https://bugs.freedesktop.org/show_bug.cgi?id=104501)**
## Description
I use a HDMI connection to my monitor with speakers. When play...## Submitted by Freek de Kruijf
Assigned to **pul..@..op.org**
**[Link to original bug (#104501)](https://bugs.freedesktop.org/show_bug.cgi?id=104501)**
## Description
I use a HDMI connection to my monitor with speakers. When playing a score in Musescore the sound is delayed by about 1 second compared to the the activity on the screen. The note in the score which is played is highlighted.
When changing the sound device in the tab "I/O" in Musescore from PulseAudio to ALSA audio, sound and picture are in sync. This suggests that pulseaudio is delaying the sound. "I/O" is reached by selecting Edit->Preferences...
I noticed earlier also this type of delay when playing a video in Firefox, however currently this is no longer the case. It was most apparent when a person is speaking in the video.
Previously I did not use HDMI to produce sound, but an old fashioned sound device, which AFAIR also went through PulseAudio, but I did not see/hear this delay.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/34No system-wide sound normalization2018-07-30T09:35:03ZBugzilla Migration UserNo system-wide sound normalization## Submitted by Nikola
Assigned to **pul..@..op.org**
**[Link to original bug (#103893)](https://bugs.freedesktop.org/show_bug.cgi?id=103893)**
## Description
Why there is no system-wide sound normalization/loudness equalization o...## Submitted by Nikola
Assigned to **pul..@..op.org**
**[Link to original bug (#103893)](https://bugs.freedesktop.org/show_bug.cgi?id=103893)**
## Description
Why there is no system-wide sound normalization/loudness equalization on Ubuntu 16.04? I think it should come preinstalled because it's such a fantastic feature and it's ommision from Ubuntu is a shame. As of now, I have to adjust volume for every other video/audio I watch/listen. Please make this possible in coming builds.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/269Headset jack partially not recognized as connected at boot2018-07-30T10:12:40ZBugzilla Migration UserHeadset jack partially not recognized as connected at boot## Submitted by Carlo Caione
Assigned to **pul..@..op.org**
**[Link to original bug (#103482)](https://bugs.freedesktop.org/show_bug.cgi?id=103482)**
## Description
Codec is RT5651. The driver codec is exporting two jack controls:...## Submitted by Carlo Caione
Assigned to **pul..@..op.org**
**[Link to original bug (#103482)](https://bugs.freedesktop.org/show_bug.cgi?id=103482)**
## Description
Codec is RT5651. The driver codec is exporting two jack controls:
numid=173,iface=CARD,name='Headphone Jack'
; type=BOOLEAN,access=r-------,values=1
: values=off
numid=174,iface=CARD,name='Headset Mic Jack'
; type=BOOLEAN,access=r-------,values=1
: values=off
The driver correctly recognizes when the jack is connected / disconnected and in general it works fine (audio output and input are switched to the headset when the jack is connected).
The problem arises when we boot the machine with the headset jack already connected. In this case pulseaudio keeps setting the audio output to the speakers and the audio input to the headset mic. When the jack is then disconnected and connected again, everything works fine.
Please note that the kernel is exporting to the userspace the correct status for the headset jack, so I guess the problem is somewhere in the pulseaudio behavior.
In attachment the logs of when pulseaudio is started with and without headset jack connected.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/275Pulseaudio doesn't set Headphone channel to 100 on connect2018-07-30T10:13:07ZBugzilla Migration UserPulseaudio doesn't set Headphone channel to 100 on connect## Submitted by Yan Pas
Assigned to **pul..@..op.org**
**[Link to original bug (#99665)](https://bugs.freedesktop.org/show_bug.cgi?id=99665)**
## Description
I use Ubuntu 16.04, kernle 4.4, pulseaudio 8, motherboard Asus B150 gami...## Submitted by Yan Pas
Assigned to **pul..@..op.org**
**[Link to original bug (#99665)](https://bugs.freedesktop.org/show_bug.cgi?id=99665)**
## Description
I use Ubuntu 16.04, kernle 4.4, pulseaudio 8, motherboard Asus B150 gaming (HD audio is on)
I've speakers connected via rear 3.5 jack.
When I connect headphones in the front jack, I'm automatically switched to them, but the volume is too low, because the loudness of front headphones consists of Master + Headphones levels. Each time I disconnect headphnes, Headphones level is alsamixer resets to 0.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/463Lack of hardware volume no longer supported with HifimeDIY Tiny2018-07-30T10:29:41ZBugzilla Migration UserLack of hardware volume no longer supported with HifimeDIY Tiny## Submitted by Jerome
Assigned to **pul..@..op.org**
**[Link to original bug (#99607)](https://bugs.freedesktop.org/show_bug.cgi?id=99607)**
## Description
Hi,
I'm using a HifimeDIY Tiny USB DAC (http://hifimediy.com/DACs/ready-...## Submitted by Jerome
Assigned to **pul..@..op.org**
**[Link to original bug (#99607)](https://bugs.freedesktop.org/show_bug.cgi?id=99607)**
## Description
Hi,
I'm using a HifimeDIY Tiny USB DAC (http://hifimediy.com/DACs/ready-made-dacs/tiny-dac), which doesn't support hardware volume (fixed analog gain). This used to work on Debian Jessie, but no longer with Stretch.
This USB DAC is peculiar: it has a first DAC, the TI Burr-Brown, which is only used for its USB interface. Its analog interface is not connected nor used, but the Burr-Brown SPDIF output is connected to the input of a Sabre DAC. It's the Sabre that is actually used to generate the analog output. And the set-up has a fixed analog gain, so the volume control must be done in software.
In Debian Jessie, I could see the two outputs of the Burr-Brown under pulseaudio, analog and SPDIF (IEC958). Analog was the default and of course gave no output, but I just had to select the IEC958 output to get a working system, with proper volume control.
Now on Stretch I don't see the IEC958 output at all. The output is configured as analog stereo by default, and I do have audio output! So the SPDIF output of the burr-Brown must be used then? However pulseaudio/alsa believe there is hardware volume control, which is not the case: if the volume is at any value but zero, then the output level is full blast. It's really a step function: mute or max output.
I guess this could be solved in one of two ways?:
1) show back the IEC958 option
2) disable hardware volume control for this card
I've tried to look for ways to do this without success, but then Linux sound support is new to me so I may have missed something. Thanks for any help/pointer.
I add some info on the dongle as seen by the system:
== pacmd list-cards (just the dongle) ===========================
index: 2
name: <alsa_card.usb-Burr-Brown_from_TI_USB_Audio_DAC-00>
driver: <module-alsa-card.c>
owner module: 29
properties:
alsa.card = "2"
alsa.card_name = "USB Audio DAC"
alsa.long_card_name = "Burr-Brown from TI USB Audio DAC at usb-0000:00:14.0-1, full speed"
alsa.driver_name = "snd_usb_audio"
device.bus_path = "pci-0000:00:14.0-usb-0:1:1.0"
sysfs.path = "/devices/pci0000:00/0000:00:14.0/usb1/1-1/1-1:1.0/sound/card2"
udev.id = "usb-Burr-Brown_from_TI_USB_Audio_DAC-00"
device.bus = "usb"
device.vendor.id = "08bb"
device.vendor.name = "Texas Instruments"
device.product.id = "2706"
device.product.name = "PCM2706 Audio Codec"
device.serial = "Burr-Brown_from_TI_USB_Audio_DAC"
device.string = "2"
device.description = "PCM2706 Audio Codec"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-usb"
profiles:
output:analog-mono: Analog Mono Output (priority 200, available: unknown)
output:analog-stereo: Analog Stereo Output (priority 6000, available: unknown)
off: Off (priority 0, available: unknown)
active profile: <output:analog-stereo>
sinks:
alsa_output.usb-Burr-Brown_from_TI_USB_Audio_DAC-00.analog-stereo/#2: PCM2706 Audio Codec Analog Stereo
sources:
alsa_output.usb-Burr-Brown_from_TI_USB_Audio_DAC-00.analog-stereo.monitor/#3: Monitor of PCM2706 Audio Codec Analog Stereo
ports:
analog-output: Analog Output (priority 9900, latency offset 0 usec, available: unknown)
properties:
== pacmd list-sources (only USB DAC) =====================
index: 3
name: <alsa_output.usb-Burr-Brown_from_TI_USB_Audio_DAC-00.analog-stereo.monitor>
driver: <module-alsa-card.c>
flags: DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
state: SUSPENDED
suspend cause: IDLE
priority: 1040
volume: front-left: 65536 / 100% / 0.00 dB, front-right: 65536 / 100% / 0.00 dB
balance 0.00
base volume: 65536 / 100% / 0.00 dB
volume steps: 65537
muted: no
current latency: 0.00 ms
max rewind: 0 KiB
sample spec: s16le 2ch 44100Hz
channel map: front-left,front-right
Stereo
used by: 0
linked by: 0
configured latency: 0.00 ms; range is 0.50 .. 2000.00 ms
monitor_of: 2
card: 2 <alsa_card.usb-Burr-Brown_from_TI_USB_Audio_DAC-00>
module: 29
properties:
device.description = "Monitor of PCM2706 Audio Codec Analog Stereo"
device.class = "monitor"
alsa.card = "2"
alsa.card_name = "USB Audio DAC"
alsa.long_card_name = "Burr-Brown from TI USB Audio DAC at usb-0000:00:14.0-1, full speed"
alsa.driver_name = "snd_usb_audio"
device.bus_path = "pci-0000:00:14.0-usb-0:1:1.0"
sysfs.path = "/devices/pci0000:00/0000:00:14.0/usb1/1-1/1-1:1.0/sound/card2"
udev.id = "usb-Burr-Brown_from_TI_USB_Audio_DAC-00"
device.bus = "usb"
device.vendor.id = "08bb"
device.vendor.name = "Texas Instruments"
device.product.id = "2706"
device.product.name = "PCM2706 Audio Codec"
device.serial = "Burr-Brown_from_TI_USB_Audio_DAC"
device.string = "2"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-usb"
== lsusb -d 08bb:2706 -v ==========================
Bus 001 Device 005: ID 08bb:2706 Texas Instruments PCM2706 Audio Codec
Device Descriptor:
bLength 18
bDescriptorType 1
bcdUSB 1.10
bDeviceClass 0 (Defined at Interface level)
bDeviceSubClass 0
bDeviceProtocol 0
bMaxPacketSize0 8
idVendor 0x08bb Texas Instruments
idProduct 0x2706 PCM2706 Audio Codec
bcdDevice 1.00
iManufacturer 1 Burr-Brown from TI
iProduct 2 USB Audio DAC
iSerial 0
bNumConfigurations 1
Configuration Descriptor:
bLength 9
bDescriptorType 2
wTotalLength 190
bNumInterfaces 3
bConfigurationValue 1
iConfiguration 0
bmAttributes 0x80
(Bus Powered)
MaxPower 100mA
Interface Descriptor:
bLength 9
bDescriptorType 4
bInterfaceNumber 0
bAlternateSetting 0
bNumEndpoints 0
bInterfaceClass 1 Audio
bInterfaceSubClass 1 Control Device
bInterfaceProtocol 0
iInterface 0
AudioControl Interface Descriptor:
bLength 9
bDescriptorType 36
bDescriptorSubtype 1 (HEADER)
bcdADC 1.00
wTotalLength 40
bInCollection 1
baInterfaceNr( 0) 1
AudioControl Interface Descriptor:
bLength 12
bDescriptorType 36
bDescriptorSubtype 2 (INPUT_TERMINAL)
bTerminalID 1
wTerminalType 0x0101 USB Streaming
bAssocTerminal 0
bNrChannels 2
wChannelConfig 0x0003
Left Front (L)
Right Front (R)
iChannelNames 0
iTerminal 0
AudioControl Interface Descriptor:
bLength 9
bDescriptorType 36
bDescriptorSubtype 3 (OUTPUT_TERMINAL)
bTerminalID 2
wTerminalType 0x0301 Speaker
bAssocTerminal 0
bSourceID 3
iTerminal 0
AudioControl Interface Descriptor:
bLength 10
bDescriptorType 36
bDescriptorSubtype 6 (FEATURE_UNIT)
bUnitID 3
bSourceID 1
bControlSize 1
bmaControls( 0) 0x01
Mute Control
bmaControls( 1) 0x02
Volume Control
bmaControls( 2) 0x02
Volume Control
iFeature 0
Interface Descriptor:
bLength 9
bDescriptorType 4
bInterfaceNumber 1
bAlternateSetting 0
bNumEndpoints 0
bInterfaceClass 1 Audio
bInterfaceSubClass 2 Streaming
bInterfaceProtocol 0
iInterface 0
Interface Descriptor:
bLength 9
bDescriptorType 4
bInterfaceNumber 1
bAlternateSetting 1
bNumEndpoints 1
bInterfaceClass 1 Audio
bInterfaceSubClass 2 Streaming
bInterfaceProtocol 0
iInterface 0
AudioStreaming Interface Descriptor:
bLength 7
bDescriptorType 36
bDescriptorSubtype 1 (AS_GENERAL)
bTerminalLink 1
bDelay 0 frames
wFormatTag 1 PCM
AudioStreaming Interface Descriptor:
bLength 17
bDescriptorType 36
bDescriptorSubtype 2 (FORMAT_TYPE)
bFormatType 1 (FORMAT_TYPE_I)
bNrChannels 2
bSubframeSize 2
bBitResolution 16
bSamFreqType 3 Discrete
tSamFreq[ 0] 32000
tSamFreq[ 1] 44100
tSamFreq[ 2] 48000
Endpoint Descriptor:
bLength 9
bDescriptorType 5
bEndpointAddress 0x02 EP 2 OUT
bmAttributes 9
Transfer Type Isochronous
Synch Type Adaptive
Usage Type Data
wMaxPacketSize 0x00c0 1x 192 bytes
bInterval 1
bRefresh 0
bSynchAddress 0
AudioControl Endpoint Descriptor:
bLength 7
bDescriptorType 37
bDescriptorSubtype 1 (EP_GENERAL)
bmAttributes 0x00
bLockDelayUnits 2 Decoded PCM samples
wLockDelay 512 Decoded PCM samples
Interface Descriptor:
bLength 9
bDescriptorType 4
bInterfaceNumber 1
bAlternateSetting 2
bNumEndpoints 1
bInterfaceClass 1 Audio
bInterfaceSubClass 2 Streaming
bInterfaceProtocol 0
iInterface 0
AudioStreaming Interface Descriptor:
bLength 7
bDescriptorType 36
bDescriptorSubtype 1 (AS_GENERAL)
bTerminalLink 1
bDelay 0 frames
wFormatTag 1 PCM
AudioStreaming Interface Descriptor:
bLength 17
bDescriptorType 36
bDescriptorSubtype 2 (FORMAT_TYPE)
bFormatType 1 (FORMAT_TYPE_I)
bNrChannels 1
bSubframeSize 2
bBitResolution 16
bSamFreqType 3 Discrete
tSamFreq[ 0] 32000
tSamFreq[ 1] 44100
tSamFreq[ 2] 48000
Endpoint Descriptor:
bLength 9
bDescriptorType 5
bEndpointAddress 0x02 EP 2 OUT
bmAttributes 9
Transfer Type Isochronous
Synch Type Adaptive
Usage Type Data
wMaxPacketSize 0x0060 1x 96 bytes
bInterval 1
bRefresh 0
bSynchAddress 0
AudioControl Endpoint Descriptor:
bLength 7
bDescriptorType 37
bDescriptorSubtype 1 (EP_GENERAL)
bmAttributes 0x00
bLockDelayUnits 2 Decoded PCM samples
wLockDelay 512 Decoded PCM samples
Interface Descriptor:
bLength 9
bDescriptorType 4
bInterfaceNumber 2
bAlternateSetting 0
bNumEndpoints 1
bInterfaceClass 3 Human Interface Device
bInterfaceSubClass 0 No Subclass
bInterfaceProtocol 0 None
iInterface 0
HID Device Descriptor:
bLength 9
bDescriptorType 33
bcdHID 1.00
bCountryCode 0 Not supported
bNumDescriptors 1
bDescriptorType 34 Report
wDescriptorLength 36
Report Descriptors:
** UNAVAILABLE **
Endpoint Descriptor:
bLength 7
bDescriptorType 5
bEndpointAddress 0x85 EP 5 IN
bmAttributes 3
Transfer Type Interrupt
Synch Type None
Usage Type Data
wMaxPacketSize 0x0001 1x 1 bytes
bInterval 10
Device Status: 0x0000
(Bus Powered)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/531Front Panel switch on Xonar DX2022-05-03T13:14:34ZBugzilla Migration UserFront Panel switch on Xonar DX## Submitted by sla..@..hon.su
Assigned to **pul..@..op.org**
**[Link to original bug (#99479)](https://bugs.freedesktop.org/show_bug.cgi?id=99479)**
## Description
Created attachment 129078
amixer -c1
Card: Xonar DX
Pulse 9.0
N...## Submitted by sla..@..hon.su
Assigned to **pul..@..op.org**
**[Link to original bug (#99479)](https://bugs.freedesktop.org/show_bug.cgi?id=99479)**
## Description
Created attachment 129078
amixer -c1
Card: Xonar DX
Pulse 9.0
Now for changing front/back output i must did it only on alsamixer via `Front Panel` switch.
Also `Headphone FP` (or any other FP), `Headphone` profiles are absent
**Attachment 129078**, "amixer -c1":
[amixer.log](/uploads/f8f261117c88285746b3c79e4c796e40/amixer.log)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/373HDMI: whining noise (high frequent beep)2018-07-30T12:32:23ZBugzilla Migration UserHDMI: whining noise (high frequent beep)## Submitted by Dollinger Florian
Assigned to **pul..@..op.org**
**[Link to original bug (#99290)](https://bugs.freedesktop.org/show_bug.cgi?id=99290)**
## Description
When I plug in my TV's HDMI cable into my laptop (Acer Aspire ...## Submitted by Dollinger Florian
Assigned to **pul..@..op.org**
**[Link to original bug (#99290)](https://bugs.freedesktop.org/show_bug.cgi?id=99290)**
## Description
When I plug in my TV's HDMI cable into my laptop (Acer Aspire R3 131T, Intel Audio) i get a really annoying, high frequent whine from the TV speakers.
It doesn't matter how loud the volume is set, the whistle is always the same
until I playback some music or video or something else via HDMI, even if the music is super silent - the noise disappears as long as the music lasts.
After hitting the pause button, the debug log looks something like that:
--------------------------
D: [alsa-sink-HDMI 0] alsa-sink.c: Cutting sleep time for the initial iterations by half.
D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
D: [pulseaudio] module-udev-detect.c: Resuming all sinks and sources of card alsa_card.pci-0000_00_1b.0.
D: [alsa-sink-HDMI 0] sink-input.c: Requesting rewind due to corking
D: [alsa-sink-HDMI 0] alsa-sink.c: Requested to rewind 76688 bytes.
D: [alsa-sink-HDMI 0] alsa-sink.c: Limited to 6224 bytes.
D: [alsa-sink-HDMI 0] alsa-sink.c: before: 1556
D: [alsa-sink-HDMI 0] alsa-sink.c: after: 1556
D: [alsa-sink-HDMI 0] alsa-sink.c: Rewound 6224 bytes.
D: [alsa-sink-HDMI 0] sink.c: Processing rewind...
D: [alsa-sink-HDMI 0] sink-input.c: Have to rewind 6224 bytes on render memblockq.
D: [alsa-sink-HDMI 0] sink-input.c: Have to rewind 12448 bytes on implementor.
D: [alsa-sink-HDMI 0] source.c: Processing rewind...
--------------------------
The funny thing is, if I run a "speaker test" (HDMI) on gnomes audio settings, then the noise doesn't appear again (until i pause another video/music again), the debug log looks like that when the noise is gone:
--------------------------
D: [alsa-sink-HDMI 0] alsa-sink.c: after: 1632
D: [alsa-sink-HDMI 0] alsa-sink.c: Rewound 6528 bytes.
D: [alsa-sink-HDMI 0] sink.c: Processing rewind...
D: [alsa-sink-HDMI 0] sink-input.c: Have to rewind 6528 bytes on render memblockq.
D: [alsa-sink-HDMI 0] source.c: Processing rewind...
I: [pulseaudio] sink-input.c: Freeing input 24 "Vorne Rechts"
--------------------------
It also doesn't happen on the _internal_ speakers of my laptop, after hitting pause on VLC, the debug looks like that:
--------------------------
I: [pulseaudio] module-stream-restore.c: Synced.
D: [alsa-sink-ALC255 Analog] protocol-native.c: Requesting rewind due to end of underrun.
D: [alsa-sink-ALC255 Analog] sink-input.c: Requesting rewind due to uncorking
D: [alsa-sink-ALC255 Analog] alsa-sink.c: Requested to rewind 352800 bytes.
D: [alsa-sink-ALC255 Analog] alsa-sink.c: Limited to 6608 bytes.
D: [alsa-sink-ALC255 Analog] alsa-sink.c: before: 1652
D: [alsa-sink-ALC255 Analog] alsa-sink.c: after: 1652
D: [alsa-sink-ALC255 Analog] alsa-sink.c: Rewound 6608 bytes.
D: [alsa-sink-ALC255 Analog] sink.c: Processing rewind...
D: [alsa-sink-ALC255 Analog] sink.c: latency = 773
D: [alsa-sink-ALC255 Analog] source.c: Processing rewind...
D: [alsa-sink-ALC255 Analog] sink-input.c: Requesting rewind due to corking
D: [alsa-sink-ALC255 Analog] alsa-sink.c: Requested to rewind 101200 bytes.
D: [alsa-sink-ALC255 Analog] alsa-sink.c: Limited to 6224 bytes.
D: [alsa-sink-ALC255 Analog] alsa-sink.c: before: 1556
D: [alsa-sink-ALC255 Analog] alsa-sink.c: after: 1556
D: [alsa-sink-ALC255 Analog] alsa-sink.c: Rewound 6224 bytes.
D: [alsa-sink-ALC255 Analog] sink.c: Processing rewind...
D: [alsa-sink-ALC255 Analog] sink.c: latency = 729
D: [alsa-sink-ALC255 Analog] sink-input.c: Have to rewind 6224 bytes on render memblockq.
D: [alsa-sink-ALC255 Analog] sink-input.c: Have to rewind 12448 bytes on implementor.
D: [alsa-sink-ALC255 Analog] source.c: Processing rewind...
--------------------------
My System is:
Linux Apricity (12-2016)
based on Linux Arch
Pulseaudio Version 9https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/501no output device created for dock when dock has 2 outputs, Dell Dock TB152018-07-30T10:32:58ZBugzilla Migration Userno output device created for dock when dock has 2 outputs, Dell Dock TB15## Submitted by aap..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#98926)](https://bugs.freedesktop.org/show_bug.cgi?id=98926)**
## Description
Created attachment 128302
aplay and pactl list cards output
When...## Submitted by aap..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#98926)](https://bugs.freedesktop.org/show_bug.cgi?id=98926)**
## Description
Created attachment 128302
aplay and pactl list cards output
When using a TB15 dock on an XPS 13 (9350) the front output works fine but the output in the back of the dock is not accessible by pulse audio. alsa finds the card and the two devices fine. In the attached aplay -L and pactl list cards output you see USB Audio #1 listed correctly and playing a sound to that device via aplay plughw:1,1 works fine but pulseaudio does not seem to create an output for it.
pavucontrol lists the USB device but its only the front output (which is designed for a headset).
**Attachment 128302**, "aplay and pactl list cards output":
[notes.txt](/uploads/66798f794f5682c9800510247c3988ce/notes.txt)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/179Rewinding results in omitted samples2018-07-30T10:02:06ZBugzilla Migration UserRewinding results in omitted samples## Submitted by Kai Ruhnau
Assigned to **pul..@..op.org**
**[Link to original bug (#98674)](https://bugs.freedesktop.org/show_bug.cgi?id=98674)**
## Description
Created attachment 127891
Test tone
I'm running my Yocto distributio...## Submitted by Kai Ruhnau
Assigned to **pul..@..op.org**
**[Link to original bug (#98674)](https://bugs.freedesktop.org/show_bug.cgi?id=98674)**
## Description
Created attachment 127891
Test tone
I'm running my Yocto distribution (morty with PulseAudio 9.0) with Linux 4.7 on an i.MX6SX platform containing a Wolfson WM9715 codec driven by the wm9712 driver.
I'm trying to play a small test wav file with about 100ms duration (see attachment). I can successfully use aplay on the hardware device directly.
When I use PulseAudio to play that file (for example through aplay or by upload-sample/play-sample), sometimes samples at the beginning are omitted and don't reach the analog audio output (see following attachments).
It doesn't happen if this is the first time something is played after the server has started. It then happens every time (with varying number of samples) until the alsa card is suspended. It then again doesn't happen once after the card is unsuspended.
After a small IRC session, I already have disabled the rewinding code (https://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/alsa-sink.c?id=8887f256e0f1167e579a217387a4efc5397edf93#n1649) and this makes the problem go away, but I'd be surprised if this was even close to an actual solution.
**Attachment 127891**, "Test tone":
[Testton_Pegel_100Proz_Stereo.wav](/uploads/35644391fc36c50f9bdbc065e7f4f0cd/Testton_Pegel_100Proz_Stereo.wav)https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/442Audio is muted and mapped wrongly by default on Focusrite Scarlett 18i82018-07-30T10:28:07ZBugzilla Migration UserAudio is muted and mapped wrongly by default on Focusrite Scarlett 18i8## Submitted by Elias Hogstvedt `@CapsAdmin`
Assigned to **pul..@..op.org**
**[Link to original bug (#96648)](https://bugs.freedesktop.org/show_bug.cgi?id=96648)**
## Description
The first time I plugged this device in while being...## Submitted by Elias Hogstvedt `@CapsAdmin`
Assigned to **pul..@..op.org**
**[Link to original bug (#96648)](https://bugs.freedesktop.org/show_bug.cgi?id=96648)**
## Description
The first time I plugged this device in while being relatively new to linux I spent a lot of time figuring out why the audio wasn't working as opening a terminal and typing alsamixer, selecting the soundcard and pressing M on the output to unmute it wasn't something I had thought of. The left and right channel are also PCM 1/PCM 1 by default but they should be PCM 1/PCM 2. So after unmuting and setting the right channels there's audio at least.
But after that there's the issue that it's being treated as a 5.1 system so the bass is on some other channel. There's an existing issue for that though https://bugs.freedesktop.org/show_bug.cgi?id=93163
I don't know how you handle specific devices. If they don't conform do you do special case handling? Or should that happen somewhere else?https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/26Pulseaudio does not create a separate source for each input subdevice on a card2020-08-04T10:05:08ZBugzilla Migration UserPulseaudio does not create a separate source for each input subdevice on a card## Submitted by ror..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#96415)](https://bugs.freedesktop.org/show_bug.cgi?id=96415)**
## Description
I am using Pulseaudio 1:8.0-0ubuntu3 on ubuntu xenial.
My built i...## Submitted by ror..@..il.com
Assigned to **pul..@..op.org**
**[Link to original bug (#96415)](https://bugs.freedesktop.org/show_bug.cgi?id=96415)**
## Description
I am using Pulseaudio 1:8.0-0ubuntu3 on ubuntu xenial.
My built in sound card has two analog input devices, one with a single subdevice, and the other with two subdevices. Here is the output of arecord -l:
$ arecord -l
**** List of CAPTURE Hardware Devices ****
card 0: SB [HDA ATI SB], device 0: ALC889A Analog [ALC889A Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 0: SB [HDA ATI SB], device 1: ALC889A Digital [ALC889A Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: SB [HDA ATI SB], device 2: ALC889A Alt Analog [ALC889A Alt Analog]
Subdevices: 2/2
Subdevice #0: subdevice #0
Subdevice #1: subdevice #1
$ pactl list short sources
4 alsa_output.pci-0000_00_14.2.iec958-stereo.monitor module-alsa-card.c s16le 2ch 48000Hz RUNNING
5 alsa_input.pci-0000_00_14.2.analog-stereo module-alsa-card.c s16le 2ch 48000Hz RUNNING
I would expect Pulseaudio (perhaps module-udev-detect?) to detect and create 3 input sources to allow access to all three subdevices (hw:0,0,0; hw:0,2,0; hw:0,2,1). This would allow me to record from all three analog input ports at once. However, it seems that pulseaudio stops looking after it finds the first subdevice (hw:0,0,0)
alsamixer detects all three capture ports, and allows me to assign each capture port to a separate input source.
pavucontrol allows me to switch between capture ports on a single input source. I could understand this for devices with only a single capture subdevice, but it is severely limiting for a device with more than one capture subdevice.
I wish to use pulseaudio as a mixer for three loopback sources on my HTPC: a Chromecast audio, my TV, and a phono amp.https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/271[NEED]Try recover from snd_pcm_avail() -EPIPE2018-07-30T10:12:53ZBugzilla Migration User[NEED]Try recover from snd_pcm_avail() -EPIPE## Submitted by IvanXie
Assigned to **pul..@..op.org**
**[Link to original bug (#96361)](https://bugs.freedesktop.org/show_bug.cgi?id=96361)**
## Description
ALSA sink has been unloaded due to process_rewind() return -1.
"alsa-si...## Submitted by IvanXie
Assigned to **pul..@..op.org**
**[Link to original bug (#96361)](https://bugs.freedesktop.org/show_bug.cgi?id=96361)**
## Description
ALSA sink has been unloaded due to process_rewind() return -1.
"alsa-sink.c: process_rewind(1754) > [alsa-sink-Multimedia1 (*)] snd_pcm_avail() failed: Broken pipe"
"sink.c: sink_free(843) > [pulseaudio] Freeing sink 2 "alsa_output.0.analog-stereo""
"module.c pa_module_free(183) > [pulseaudio] unloaded "module-alsa-sink" (index: #15)"
Actually, sometimes we may meet snd_pcm_avail() -EPIPE from process_rewind, pa_alsa_safe_avail. But how about do recover for this error case instead of return error and unload sink?
I've tried make a patch for this issue. But I'm not sure about this is a proper solution. Please give me your opinion. Thank you.
if (PA_UNLIKELY((unused = pa_alsa_safe_avail(u->pcm_handle, u->hwbuf_size, &u->sink->sample_spec)) < 0)) {
pa_log("snd_pcm_avail() failed: %s", pa_alsa_strerror((int) unused));
#ifdef RECOVER_FROM_EPIPE
/* try recover if we got -EPIPE from snd_pcm_avail */
if (unsed == -EPIPE) {
pa_log("snd_pcm_avail() failed of -EPIPE. Try recover!!!");
if (try_recover(u, "process_rewind", unused) < 0)
return -1;
else {
pa_log_info("Tried rewind, but was apparently not possible.");
pa_sink_process_rewind(u->sink, 0);
return 0;
}
}
#endif
return -1;
}https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/343device reservation fails when a device string has seperator(,)2018-07-30T10:19:28ZBugzilla Migration Userdevice reservation fails when a device string has seperator(,)## Submitted by KimJeongYeon
Assigned to **pul..@..op.org**
**[Link to original bug (#94666)](https://bugs.freedesktop.org/show_bug.cgi?id=94666)**
## Description
For example, the device parameter of pa_alsa_get_reserve_name() is ...## Submitted by KimJeongYeon
Assigned to **pul..@..op.org**
**[Link to original bug (#94666)](https://bugs.freedesktop.org/show_bug.cgi?id=94666)**
## Description
For example, the device parameter of pa_alsa_get_reserve_name() is "hw:USB,0",
snd_card_get_index() always fails. Because seperator(,) wasn't regarded.
Therefore, JACK couldn't obtain audio device from Pulseaudio.