I recently removed JACK via sudo apt purge --autoremove multimedia-jack. However, many of its files remain, including an entire GUI application for a system I don't use. It says it's automatically installed in apt, so how do I find what's "requiring" it?

Not sure if the topic is too audio-specific to be posted here, if so then do tell and I'd appreciate suggestions on proper forums. My LP recording setup is through an M-audio 2496 card on linux running jack, using a command line script to capture. I use the hardware input to capture and set the recording level using alsamixer; My usual capture level has been -3 dB for years. These days I'm testing a new phono preamp that has an higher output level than my other units so I lowered the capture level to -6dB but ran into clipping issues. Investigating this problem, it turns out that setting the capture level below -3 dB on hardware channels "H/W Multi" (left) and "H/W Multi 1" (right) clips the signal ! Perhaps some pics would help illustrate the phenomenon. Below are shots of the same high modulation passage of an LP record at different capture levels in alsamixer. Note that except for the 0 dB test, there were no Xruns or overruns during the recording process. At 0 dB I get overload peaks as can be expected; at -3 dB things seem ok like my usual setup; Note the recorded amplitude level. But as you can see, lowering capture level further not only reduces overall recorded amplitude but clips the peaks, even if they reach a lower amplitude than the recording at -3 dB. I always assumed the alsamixer capture level worked as an equivalent of the "recording level" on the cassette decks of my youth, e.g. simply attenuated the signal. Now it seems it both attenuates and limits somehow.... I'm confused! Please help me better understand what's going on here and thanks in advance for any insights. -Joe

I have a M-Audio fasttrack USB interface that has a mic input and an instrument input. Mic is channel 1 and instrument is channel 2, which is the opposite of most other interfaces that have this layout, including my Line 6 pod X3. So, to load my presets/patches in, let's say, carla, for the time being i have to either (1) manually connect the right JACK inputs to each carla instance when i'm using the interface that doesn't match, or (2) have several copies of each patch file, one for each interface. This is kind of annoying because i need to use one of multiple interfaces depending what i'm doing (recording, jamming with friends, etc) Is there a way to swap or remap the channels directly from JACK, like for example some undocumented commandline/config option - or through a proxy client that would then be used as input for carla/renoise/etc?

I'm running a debian 'testing' system: Operating System: Debian GNU/Linux 12 (bookworm) Kernel: Linux 6.1.0-7-amd64 Architecture: x86-64 Hardware Vendor: TOSHIBA Hardware Model: PORTEGE R30-A Firmware Version: Version 4.20 GNOME Shell: 43.4 I installed two audio applications for jack mixxx and guitarix, which both crash during startup with almost the same ERROR-Message: **Mixxx** *(a digital DJ system, where Wave, Ogg, FLAC and MP3 files can be mixed on a computer for use in live performances.):* mixxx: vendor/freeverb/revmodel.cpp:40: void revmodel::setrate(int): Assertion `rate <= TUNING_MAX_SAMPLE_RATE' failed. Aborted **Guitarix** *(a rock guitar amplifier for the JACK Audio Connection Kit with one input and two outputs.):* guitarix: vendor/freeverb/revmodel.cpp:40: void revmodel::setrate(int): Assertion `rate <= TUNING_MAX_SAMPLE_RATE' failed. Aborted I got jack running with 192000Hz, so I do not think the problem can be solved there: JACK compiled with System V SHM support. loading driver .. apparent rate = 192000 creating alsa driver ... hw:TR8S,0|hw:TR8S,0|256|2|192000|0|0|nomon|swmeter|-|32bit configuring for 192000Hz, period = 256 frames (1.3 ms), buffer = 2 periods ALSA: final selected sample format for capture: 32bit float little-endian ALSA: use 2 periods for capture ALSA: final selected sample format for playback: 32bit float little-endian ALSA: use 2 periods for playback Does anyone know what ***vendor/freeverb/revmodel.cpp*** is? And how I can fix this? *My first thought was that I had installed an incompatible plugin, but this does not seem to be the case.*

Linux systems have been having a historical (>5 yrs) problem in configuring audio devices especially commonplace headphones with combo jacks. Since many people want to use their favorite linux systems for video chatting, there are records of frustrating unresolved problems all across various forums. I get that the drivers for the external mic (in headphones with combo jacks) are not currently available (or not developed(?)). So given that, the user should be able to use the internal microphone for input and headphone for output. Along this line I went down the rabbit hole (digged up upto 5-6 yr old issues) and tried lot of things only to get no success in the end (I am using common combo jack headphones and hp laptop running ubuntu 16.04). Many people have variously reported this issue. Here's what commonly happens.. # When headphone is not connected, Internal microphone and speakers work well. PulseAudio shows: pacmd list-cards shows: ports: analog-input-internal-mic: Internal Microphone (priority 8900, latency offset 0 usec, available: unknown) properties: device.icon_name = "audio-input-microphone" analog-input-mic: Microphone (priority 8700, latency offset 0 usec, available: no) properties: device.icon_name = "audio-input-microphone" analog-output-speaker: Speakers (priority 10000, latency offset 0 usec, available: unknown) properties: device.icon_name = "audio-speakers" analog-output-headphones: Headphones (priority 9000, latency offset 0 usec, available: no) properties: device.icon_name = "audio-headphones" # When headphone is connected, Output through headphones work well. But external microphone (located on headphone) does not work (stuttering noise, drivers not present, alright), but then Internal microphone is 'unplugged' too. (So there is no way to record any sound.) PulseAudio shows: pacmd list-cards shows: ports: analog-input-internal-mic: Internal Microphone (priority 8900, latency offset 0 usec, available: no) properties: device.icon_name = "audio-input-microphone" analog-input-mic: Microphone (priority 8700, latency offset 0 usec, available: yes) properties: device.icon_name = "audio-input-microphone" analog-output-speaker: Speakers (priority 10000, latency offset 0 usec, available: no) properties: device.icon_name = "audio-speakers" analog-output-headphones: Headphones (priority 9000, latency offset 0 usec, available: yes) properties: device.icon_name = "audio-headphones" **So, headphones gives the output, that's great, but is there any way to force internal microphone for input? (somehow make available: yes)**

The only command line solution for gapless playback I found so far (working with ALSA and JACK) is moc (»music on console«). While I'm still searching for a simpler way I was wondering if it is possible to loop an audio file into a new file for a given number of times? Something like: loop-audio infile.flac --loop 32 outfile.flac for repeating infile.flac 32 times into outfile.flac

While using pipewire jack, I experience stuttering and other "artifacts" when playing complex projects in REAPER. My journal is being filled with messages of this kind: > pipewire: spa.audioadapter: 0x5623327d2e28: scheduling stopped node I looked online and couldn't find anything useful. Does anyone know what this might be referring too?

Distro: Debian Bullseye Audio interface: M-Audio M-Track 2X2M, Focusrite Scarlett 2i4 I can play any sound fine with vlc, mpv, quodlibet, firefox. I start jack and I can still plays sound by telling each app to use [Pulseaudio jack sink](https://askubuntu.com/a/1213554/419514) . When stopping jack, each software goes back from jack sink to audio interface. They all play fine (after a short transition glitch when stopping jack) except for vlc which has audio playback issues (image seems fine). The symptoms vary slightly between the two audio interfaces. When using the M-Audio M-Track 2X2M, the sounds keeps stopping every few seconds and the console is filled with those errors:

[0000559265fdea20] pulse audio output warning: starting late (-17920 us)
[0000559265fdea20] main audio output warning: playback way too early (-512867): playing silence
[0000559265fdea20] main audio output warning: playback too late (60253): up-sampling
[0000559265fdea20] main audio output warning: timing screwed (drift: 121333 us): stopping resampling
[0000559265fdea20] main audio output warning: playback too late (121292): up-sampling
[0000559265fdea20] main audio output warning: playback way too late (182033): flushing buffers
[0000559265fdea20] main audio output warning: playback way too early (-508497): playing silence
[0000559265fdea20] main audio output warning: playback too late (75859): up-sampling
[0000559265fdea20] main audio output warning: timing screwed (drift: 154957 us): stopping resampling
[0000559265fdea20] main audio output warning: playback too late (154937): up-sampling
[0000559265fdea20] main audio output warning: playback way too late (194625): flushing buffers
[0000559265fdea20] pulse audio output warning: starting late (-1470 us)
[0000559265fdea20] main audio output warning: playback way too early (-508119): playing silence
[0000559265fdea20] main audio output warning: playback too late (60012): up-sampling
[0000559265fdea20] main audio output warning: timing screwed (drift: 120048 us): stopping resampling
[0000559265fdea20] main audio output warning: playback too late (120331): up-sampling
[0000559265fdea20] main audio output warning: playback way too late (180143): flushing buffers
[0000559265fdea20] pulse audio output warning: starting late (-3654 us)
[0000559265fdea20] main audio output warning: playback way too early (-548874): playing silence
[0000559265fdea20] main audio output warning: playback too late (77824): up-sampling
[0000559265fdea20] main audio output warning: timing screwed (drift: 159831 us): stopping resampling
[0000559265fdea20] main audio output warning: playback too late (159790): up-sampling
[0000559265fdea20] main audio output warning: playback way too late (180711): flushing buffers
[0000559265fdea20] pulse audio output warning: starting late (-19826 us)
[0000559265fdea20] main audio output warning: playback way too early (-513173): playing silence
[0000559265fdea20] main audio output warning: playback too late (60207): up-sampling

When using the Scarlett, I don't get those errors and the sound doesn't stop but it sounds dirty, as if it clipped. vlc seems to be the only affected player. Restarting vlc doesn't help. Changing output to internal soundcard then back doesn't help either. Issue stops when selecting internal sound card then starts again when selecting audio interface. Restarting pulseaudio (then restarting vlc) does solve the problem (until next time).

systemctl --user restart pulseaudio

I'm clueless. I don't see anything relevant in syslog. The only log I can find that shows something related is the vlc log quoted above, but I'm afraid it only shows the consequence, not the root cause. -------------------------------------- I just noticed those errors in QjackCtl when stopping jack. They only appear when using pulseaudio jack sink.

Sat Dec  3 18:13:29 2022: Client 'qjackctl' with PID 5618 is out
Sat Dec  3 18:13:29 2022: Stopping jack server...
Sat Dec  3 18:13:29 2022: Client 'system' with PID 0 is out
Sat Dec  3 18:13:29 2022: Client 'PulseAudio JACK Sink' with PID 5393 is out
Sat Dec  3 18:13:29 2022: Client 'PulseAudio JACK Source' with PID 5393 is out
Sat Dec  3 18:13:29 2022: ERROR: Cannot write socket fd = 47 err = Broken pipe
Sat Dec  3 18:13:29 2022: ERROR: CheckRes error
Sat Dec  3 18:13:29 2022: ERROR: Could not write notification
Sat Dec  3 18:13:29 2022: ERROR: ClientNotify fails name = system notification = 1 val1 = 0 val2 = 0
Sat Dec  3 18:13:29 2022: ERROR: Cannot write socket fd = 51 err = Broken pipe
Sat Dec  3 18:13:29 2022: ERROR: CheckRes error
Sat Dec  3 18:13:29 2022: ERROR: Could not write notification
Sat Dec  3 18:13:29 2022: ERROR: ClientNotify fails name = system notification = 1 val1 = 0 val2 = 0
Sat Dec  3 18:13:29 2022: Released audio card Audio1
Sat Dec  3 18:13:29 2022: ERROR: Cannot write socket fd = 47 err = Broken pipe
Sat Dec  3 18:13:29 2022: ERROR: CheckRes error
Sat Dec  3 18:13:29 2022: ERROR: Could not write notification
Sat Dec  3 18:13:29 2022: ERROR: ClientNotify fails name = freewheel notification = 1 val1 = 0 val2 = 0
Sat Dec  3 18:13:29 2022: ERROR: Cannot write socket fd = 51 err = Broken pipe
Sat Dec  3 18:13:29 2022: ERROR: CheckRes error
Sat Dec  3 18:13:29 2022: ERROR: Could not write notification
Sat Dec  3 18:13:29 2022: ERROR: ClientNotify fails name = freewheel notification = 1 val1 = 0 val2 = 0

Is it possible that when stopping jack, pulseaudio jack sink is interrupted in a not so nice way and this screws up pulseaudio?

I see that with audio servers (in my case, pipewire) you can alter the "latency". (please forgive me, I am not very knowledgeable with these things.)

="128/48000"

The Arch Linux wiki described this as "request[ing] a custom buffer size". I was wondering, is there a "downside" to setting the latency really low. Is it simply more responsive audio a higher cost of resources?

I've played around with jack in the past, but I have never really been successful with it because it feels overwhelming and it doesn't allow my computer audio output to continue functioning while jack is running. Trying to play songs in my media player has them hang at 0 seconds attempting to play them; presumably they are writing PCM or something to pulseaudio or alsa and that write is never completing. Is there a way to have jack running most or all of the time in the background while still being able to hear computer audio output? In my case, I have a USB audio interface (a Focusrite Scarlett) which is connected to a pair of speakers which I use as my default audio output, but I'm not sure whether the output type (i.e. USB or an internal sound card) matters here. I'm on Ubuntu 20.04 LTS running the latest HWE kernel (actually elementary OS Odin, but based on Ubuntu 20.04). --- **EDIT:** I was able to completely bypass all of this by using PipeWire. PipeWire is a low-latency audio system which can speak PulseAudio, ALSA, and Jack protocols, and can easily replace both PulseAudio and Jack. I replaced my desktop's PulseAudio server with PipeWire, it isn't too hard, and enabled PipeWire for Jack as well, so now everything goes to one unified place and I don't need to do silly workarounds to make things work, they just do, and everything is none the wiser. I've written up how to do this for my distribution, elementary OS 6, in [this Gist](https://gist.github.com/naftulikay/1303526109bdffe8aaaa87f2ba908cac) .

On MacOs there is an app called Jack Pilot that can create virtual midi ports that can be used to route midi data between applications. Is it possible to configure JACK to do this on Linux? I have a single midi controller that I want to use to send data to Bitwig and to a serial device input. When I start Bitwig first, the midi controller is unavailable for the serial device and if I run the serial application first, Bitwig cannot receive data from the midi controller. Can anyone offer guidance or documentation on how to set up JACK to make single midi controller output available to multiple devices? I am using Arch Linux. Thanks!

I use OBS to stream into Google Meet so I can send music and a opening scene. I also use OBS option for create a virtual camera, once I use a DSRL camera and not a webcam. The issue I need to solve right now is that I can not use my Blue Yeti on OBS and on Google Meet at the same time. If I plug it into OBS, I can't connect it into Google Meet. Is there anyway I can plug my microphone into Google Meet and also connect it as a virtual microphone device into OBS? That way I could record on OBS, while I'm live on G-Meet.

Note: To ward off any of the inevitable confusion that can arise here it is important to distinguish the old series of Fender Mustang amps (has plenty of support and info on the internet) from the **GT** series. I am asking about the **GT** series here. TL;DR The Question: - Does anybody have specific knowledge or experience that can help me get the Mustang GT series of amps working in Linux? If so, please recommend a general procedure to get the features of this amp working with Linux. --- More context: I am transitioning from Windows to Linux (continually delighted and happy the FOSSier I get) and getting as much as I can working of my old setup as I can. One thing I could do in Windows was record audio using my Fender Mustang GT as an audio interface. It has a USB out. I want to attempt to configure it for recording in Linux. I am looking for high quality and low latency and will want some recommendations on what to use be it Jack / ALSA / PulseAudio, whatever. I currently have Ardour installed and Audacity. Please help, I'm using NixOS 18.03 and comfortable with getting into configs and compiling things and all that. I've used various distros I just have less experience with Linux audio since I always used to do my music production in other OSes. Think that is about to change. Thanks!

Debian Buster. I use Pulseaudio as sound server but sometimes launch Jack for MAO. When Jack is on, I can get the sound of Pulseaudio applications thanks to the pulseaudio-module-jack that adds a Pulseaudio sink to Jack (as I explained in https://askubuntu.com/a/1213554/419514) . Except I came to realize that not all applications work. I do get the sound of vlc. But when using Firefox, Quodlibet or Audacity, nothing comes out. In fact, when clicking the "play" button, the cursor on the time slider doesn't even move. The "play" button indicates the file is playing but it is not. The playback begins as soon as I stop Jack. I couldn't find any relevant log.

I'm looking for a solution to take the HTML5 audio from multiple tabs in a browser (I don't care which browser) and mix them into 1 multichannel audio stream. For example a 5.1 channel stream Browser tab 'A' --> Stream 1 - Front Left Channel Browser tab 'B' --> Stream 1 - Front Right Channel Browser tab 'C' --> Stream 1 - Rear Left Channel .... In another area I intend to split the stream to listen to different channels separately, but at this point I'm just trying to create the audio stream. I've read documentation for ALSA and JACK and am having trouble trying to find this exact scenario.

I'm running Arch, and I want to run my AKAI MPK mini mk II through Qsynth. I have gotten to the point where, when I press a key on the keyboard, Qsynth acknowledges it, however it makes no sound. Qsynth does have a soundfont installed (I've tried several, currently FluidR3_GM.sf2), and from qjackctl I can see that the keyboard is connected to the FLUID synth's input, and qsynth's outputs are connected to the system inputs. But no sound. Edit as of 08-Mar-2022: It just works now. A couple of days ago, I decided to try again. I opened Qsynth, tried to connect the keyboard to it, was told it was already connected, and then I pressed a key and it made sound. I have no idea what changed; perhaps something updated, and that's all.

I recently installed AVLinux for low-latency audio recording, which works very well with my chap USB sound card, provided I line in from an external amp, etc. However, since my bass amp line out is incredibly noisy, I have started looking into using Guitarix with either a USB Rocksmith cable or a 1/4"-3.5mm cable. I have read the AVLinux User Guide (especially re: PulseAudio source/sink), but no matter what I've tried, I don't get any output, even though Guitarix is picking up the guitar (confirmed with the tuner). I'd love to be able to get the Rocksmith cable working as well (same issue - Guitarix can "hear" from the USB but I have no way of hearing any output from Guitarix), but I'd settle for either working. I feel like I have to be missing something quite obvious. I DO get output from other sources (i.e. Firefox, Audacity).

first of all, here is the information about my system (Ubuntu Studio 21.10) - in advance: The sound output is completely distorted as soon as Jack is activated (regardless of whether via JackDBus or JackD), although it was still working without any problems until a few days ago:

silvermoon@ubuntupc:~$ lsb_release -d
Description:    Ubuntu 21.10
silvermoon@ubuntupc:~$ uname -r
5.13.0-20-lowlatency
silvermoon@ubuntupc:~$ cat /proc/asound/cards
 0 [sofhdadsp      ]: sof-hda-dsp - sof-hda-dsp
                      LENOVO-20QF0027GE-ThinkPadX1Yoga4th
 1 [AUDIO          ]: USB-Audio - CONEXANT USB AUDIO
                      Conexant CONEXANT USB AUDIO at usb-0000:0a:00.0-2.1.1.2, full speed
silvermoon@ubuntupc:~$ aplay -l
**** Liste der Hardware-Geräte (PLAYBACK) ****
Karte 0: sofhdadsp [sof-hda-dsp], Gerät 0: HDA Analog (*) []
  Sub-Geräte: 1/1
  Sub-Gerät #0: subdevice #0
Karte 0: sofhdadsp [sof-hda-dsp], Gerät 1: HDA Digital (*) []
  Sub-Geräte: 1/1
  Sub-Gerät #0: subdevice #0
Karte 0: sofhdadsp [sof-hda-dsp], Gerät 3: HDMI1 (*) []
  Sub-Geräte: 1/1
  Sub-Gerät #0: subdevice #0
Karte 0: sofhdadsp [sof-hda-dsp], Gerät 4: HDMI2 (*) []
  Sub-Geräte: 1/1
  Sub-Gerät #0: subdevice #0
Karte 0: sofhdadsp [sof-hda-dsp], Gerät 5: HDMI3 (*) []
  Sub-Geräte: 1/1
  Sub-Gerät #0: subdevice #0
Karte 1: AUDIO [CONEXANT USB AUDIO], Gerät 0: USB Audio [USB Audio]
  Sub-Geräte: 1/1
  Sub-Gerät #0: subdevice #0
silvermoon@ubuntupc:~$ aplay /usr/share/sounds/alsa/Noise.wav
Wiedergabe: WAVE '/usr/share/sounds/alsa/Noise.wav' : Signed 16 bit Little Endian, Rate: 48000 Hz, mono
silvermoon@ubuntupc:~$ lspci -nnk | grep -iA2 audio
00:1f.3 Audio device : Intel Corporation Cannon Point-LP High Definition Audio Controller [8086:9dc8] (rev 11)
        Subsystem: Lenovo Cannon Point-LP High Definition Audio Controller [17aa:2292]
        Kernel driver in use: sof-audio-pci-intel-cnl
        Kernel modules: snd_hda_intel, snd_sof_pci_intel_cnl
00:1f.4 SMBus [0c05]: Intel Corporation Cannon Point-LP SMBus Controller [8086:9da3] (rev 11)
silvermoon@ubuntupc:~$ ps -C esd
    PID TTY          TIME CMD
silvermoon@ubuntupc:~$ ps -C arts
    PID TTY          TIME CMD
silvermoon@ubuntupc:~$ ps -C pulseaudio
    PID TTY          TIME CMD
   3654 ?        00:00:08 pulseaudio
silvermoon@ubuntupc:~$ grep "^audio" /etc/group | grep "$USER" | wc -l
1
silvermoon@ubuntupc:~$ dpkg -l | tr -s " " | grep " alsa-"
ii alsa-base 1.0.25+dfsg-0ubuntu7 all ALSA driver configuration files
ii alsa-tools 1.2.2-1 amd64 Console based ALSA utilities for specific hardware
ii alsa-tools-gui 1.2.2-1 amd64 GUI based ALSA utilities for specific hardware
ii alsa-topology-conf 1.2.5.1-2 all ALSA topology configuration files
ii alsa-ucm-conf 1.2.4-2ubuntu4 all ALSA Use Case Manager configuration files
ii alsa-utils 1.2.4-1ubuntu4 amd64 Utilities for configuring and using ALSA
silvermoon@ubuntupc:~$ lsmod | grep "snd"
snd_seq_dummy          16384  0
snd_usb_audio         299008  0
snd_usbmidi_lib        36864  1 snd_usb_audio
mc                     57344  5 videodev,snd_usb_audio,videobuf2_v4l2,uvcvideo,videobuf2_common
snd_ctl_led            24576  0
snd_soc_skl_hda_dsp    24576  5
snd_soc_intel_hda_dsp_common    20480  1 snd_soc_skl_hda_dsp
snd_soc_hdac_hdmi      36864  1 snd_soc_skl_hda_dsp
snd_hda_codec_hdmi     61440  1
snd_hda_codec_realtek   147456  1
snd_hda_codec_generic    81920  1 snd_hda_codec_realtek
snd_soc_dmic           16384  1
snd_sof_pci_intel_cnl    16384  0
snd_sof_intel_hda_common    98304  1 snd_sof_pci_intel_cnl
soundwire_intel        40960  1 snd_sof_intel_hda_common
snd_sof_intel_hda      20480  1 snd_sof_intel_hda_common
snd_sof_pci            20480  2 snd_sof_intel_hda_common,snd_sof_pci_intel_cnl
snd_sof_xtensa_dsp     16384  1 snd_sof_intel_hda_common
snd_sof               135168  2 snd_sof_pci,snd_sof_intel_hda_common
snd_soc_hdac_hda       24576  1 snd_sof_intel_hda_common
snd_hda_ext_core       32768  4 snd_sof_intel_hda_common,snd_soc_hdac_hdmi,snd_soc_hdac_hda,snd_sof_intel_hda
snd_soc_acpi_intel_match    49152  2 snd_sof_intel_hda_common,snd_sof_pci_intel_cnl
snd_soc_acpi           16384  2 snd_soc_acpi_intel_match,snd_sof_intel_hda_common
snd_soc_core          294912  7 soundwire_intel,snd_sof,snd_sof_intel_hda_common,snd_soc_hdac_hdmi,snd_soc_hdac_hda,snd_soc_dmic,snd_soc_skl_hda_dsp
snd_compress           28672  1 snd_soc_core
ac97_bus               16384  1 snd_soc_core
snd_pcm_dmaengine      16384  1 snd_soc_core
snd_hda_intel          53248  0
snd_intel_dspcfg       28672  2 snd_hda_intel,snd_sof_intel_hda_common
snd_intel_sdw_acpi     20480  2 snd_sof_intel_hda_common,snd_intel_dspcfg
snd_hda_codec         147456  7 snd_hda_codec_generic,snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec_realtek,snd_soc_intel_hda_dsp_common,snd_soc_hdac_hda,snd_soc_skl_hda_dsp
snd_hda_core           94208  11 snd_hda_codec_generic,snd_hda_codec_hdmi,snd_hda_intel,snd_hda_ext_core,snd_hda_codec,snd_hda_codec_realtek,snd_soc_intel_hda_dsp_common,snd_sof_intel_hda_common,snd_soc_hdac_hdmi,snd_soc_hdac_hda,snd_sof_intel_hda
snd_hwdep              16384  2 snd_usb_audio,snd_hda_codec
snd_pcm               122880  12 snd_hda_codec_hdmi,snd_hda_intel,snd_usb_audio,snd_hda_codec,soundwire_intel,snd_sof,snd_sof_intel_hda_common,snd_soc_hdac_hdmi,snd_compress,snd_soc_core,snd_hda_core,snd_pcm_dmaengine
snd_seq_midi           20480  0
snd_seq_midi_event     16384  1 snd_seq_midi
ledtrig_audio          16384  4 snd_ctl_led,snd_hda_codec_generic,snd_sof,thinkpad_acpi
snd_rawmidi            36864  2 snd_seq_midi,snd_usbmidi_lib
snd_seq                73728  7 snd_seq_midi,snd_seq_midi_event,snd_seq_dummy
snd_seq_device         16384  3 snd_seq,snd_seq_midi,snd_rawmidi
snd_timer              40960  2 snd_seq,snd_pcm
snd                    94208  31 snd_ctl_led,snd_hda_codec_generic,snd_seq,snd_seq_device,snd_hda_codec_hdmi,snd_hwdep,snd_hda_intel,snd_usb_audio,snd_usbmidi_lib,snd_hda_codec,snd_hda_codec_realtek,snd_timer,snd_soc_hdac_hdmi,snd_compress,thinkpad_acpi,snd_soc_core,snd_pcm,snd_soc_skl_hda_dsp,snd_rawmidi
soundcore              16384  2 snd_ctl_led,snd
silvermoon@ubuntupc:~$ head -n 3 /proc/asound/card0/codec#0
Codec: Realtek ALC285
Address: 0
AFG Function Id: 0x1 (unsol 1)
silvermoon@ubuntupc:~$ head -n 3 /proc/asound/card0/codec97#0/ac97#0-0
head: '/proc/asound/card0/codec97#0/ac97#0-0' kann nicht zum Lesen geöffnet werden: Datei oder Verzeichnis nicht gefunden
silvermoon@ubuntupc:~$ head -n 3 /proc/asound/card0/codec97#0/ac97#0-0+regs
head: '/proc/asound/card0/codec97#0/ac97#0-0+regs' kann nicht zum Lesen geöffnet werden: Datei oder Verzeichnis nicht gefunden
silvermoon@ubuntupc:~$ cat ~/.asoundrc
cat: /home/silvermoon/.asoundrc: Datei oder Verzeichnis nicht gefunden
silvermoon@ubuntupc:~$ cat ~/.asoundrc.asoundconf
cat: /home/silvermoon/.asoundrc.asoundconf: Datei oder Verzeichnis nicht gefunden
silvermoon@ubuntupc:~$ cat /etc/asound.conf 
cat: /etc/asound.conf: Datei oder Verzeichnis nicht gefunden
silvermoon@ubuntupc:~$ cat .config/jack/conf.xml 




 
  alsa
  true
  true
  false
  a
 
 
  
  
  
  
  
   hw:0
   hw:0
   hw:0
   48000
   1008
   2
   true
   n
   raw
  
  
  
  
  
  
  
  
  
  
  
 
 
  
  
  
  
  
  
  
  
 

silvermoon@ubuntupc:~$

Note: In relation to the problem, it makes no difference whether frames / periods in Jack is set to this strange value of 1008 or 128 or 512 or 1024 or ... I've already tried a lot here. **Setting without Jack:** Jack is inactive, **L**oudness**L**evel (LL) 100%, aplay Front_Center.wav, output normal via pulse audio without any distortion or clipping, **Settings below are with active jack** - Setting 1: LL=100%, PulseAudioJackSink (PAJS)=100%, aplay via PulseAudioJackSink->system-playback, output is totaly distorted - Setting 2: LL=100%, PAJS=20% (shortly before distortion), aplay output not nice but understandable ;-) - Setting 3: LL=20%, Output MuseScore via Jack->system-playback, whereby the volume control in MuseScore is set to -79.5db (normal at approx. -25db): already distorted - Setting 4: same setting as in #3 but LL=60%: output totaly distorted. Same effect if volume control in MuseScore is set to more than approx. -78db. Not only the output of PAJS and MuseScore, everything is distorted, especially if the soundsource using jack couldn't be leveled down itself. Without knowing what I might have changed and where there could changes be made: A short time ago (in any case with 21.04, but I also mean under 21.10) everything worked without any problems. Do you have any ideas where to change appropriate settings? I have already done the following: - ALSA module reinstallation - Complete reset of alsa and trying different levels of the alsamixer settings, although what is strange (for me) here: I don't have a PCM controller - tried JackD instead of JackDBus - reset of pulseaudio configuration Thanks in advance

My pulse audio mixer looks like this: May be this qjackctl log can help: 16:22:20.805 Statistics reset. 16:22:20.806 ALSA connection change. 16:22:20.807 D-BUS: Service is available (org.jackaudio.service aka jackdbus). Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock 16:22:20.931 ALSA connection graph change. 16:22:29.037 D-BUS: JACK server is starting... 16:22:29.038 D-BUS: JACK server was started (org.jackaudio.service aka jackdbus). Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock Tue Nov 9 16:22:28 2021: Starting jack server... Tue Nov 9 16:22:28 2021: JACK server starting in non-realtime mode Tue Nov 9 16:22:28 2021: self-connect-mode is "Don't restrict self connect requests" Tue Nov 9 16:22:28 2021: ERROR: Cannot lock down 107341340 byte memory area (Cannot allocate memory) Tue Nov 9 16:22:29 2021: Acquired audio card Audio0 Tue Nov 9 16:22:29 2021: creating alsa driver ... hw:0|hw:0|1024|2|96000|0|0|nomon|swmeter|-|32bit Tue Nov 9 16:22:29 2021: configuring for 96000Hz, period = 1024 frames (10.7 ms), buffer = 2 periods Tue Nov 9 16:22:29 2021: ALSA: final selected sample format for capture: 32bit integer little-endian Tue Nov 9 16:22:29 2021: ALSA: use 2 periods for capture Tue Nov 9 16:22:29 2021: ALSA: final selected sample format for playback: 32bit integer little-endian Tue Nov 9 16:22:29 2021: ALSA: use 2 periods for playback Tue Nov 9 16:22:29 2021: graph reorder: new port 'system:capture_1' Tue Nov 9 16:22:29 2021: New client 'system' with PID 0 Tue Nov 9 16:22:29 2021: graph reorder: new port 'system:capture_2' Tue Nov 9 16:22:29 2021: graph reorder: new port 'system:playback_1' Tue Nov 9 16:22:29 2021: graph reorder: new port 'system:playback_2' Tue Nov 9 16:22:29 2021: New client 'PulseAudio JACK Sink' with PID 671 Tue Nov 9 16:22:29 2021: Connecting 'PulseAudio JACK Sink:front-left' to 'system:playback_1' Tue Nov 9 16:22:29 2021: Connecting 'PulseAudio JACK Sink:front-right' to 'system:playback_2' Tue Nov 9 16:22:29 2021: New client 'PulseAudio JACK Source' with PID 671 Tue Nov 9 16:22:29 2021: ERROR: JackEngine::XRun: client = PulseAudio JACK Source was not finished, state = Triggered Tue Nov 9 16:22:29 2021: ERROR: JackAudioDriver::ProcessGraphAsyncMaster: Process error Tue Nov 9 16:22:29 2021: Connecting 'system:capture_1' to 'PulseAudio JACK Source:front-left' Tue Nov 9 16:22:29 2021: Connecting 'system:capture_2' to 'PulseAudio JACK Source:front-right' Tue Nov 9 16:22:30 2021: Saving settings to "/home/human/.config/jack/conf.xml" ... 16:22:31.083 JACK connection change. 16:22:31.085 Statistics reset. 16:22:31.102 Client activated. 16:22:31.102 Patchbay deactivated. 16:22:31.115 JACK connection graph change. Cannot lock down 107341340 byte memory area (Cannot allocate memory) Tue Nov 9 16:22:31 2021: New client 'qjackctl' with PID 108656

At present I have pulseaudio configured with the jack sink and source modules:

load-module module-jack-sink 
load-module module-jack-source

I've recently discovered that the first capture input on my audio capture card is a little faulty, which has caused me to move to the second capture. Although this is problem is likely very uncommon, Discord seems to have problems with this arrangement. At present when in a VC normally, there's no issue. However when watching someone's stream, I'm unable to be heard. After opening QjackCtl, and navigating to the connection graph, I could see that capture_1 is mapped to front-left, and capture_2 is mapped to front-right. I can manually connect capture_2 to front-left, which solves my discord issue, but I'm wondering if there is a way to have this persist, since a pulseaudio -k would revert it back to the default connections. What would I need to configure, in order to have capture_2 mapped to both front-left and front-right on pulseaudio startup?