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Hello there! I apologize for bothering, but I feel totally unable to solve the issue by my own. Please, help me.
I've got a PreSonus AudioBox 22VSL USB sound card for about a decade already, and it has two input channels. During all that time I have been using it without any great issues except for one: when I plug in the microphone to the first channel, the sound is being recorded only via the left channel, and if I would re-plug my microphone to the second -- only the right channel works.
I had never experienced any troubles because of that. For instance, I could record an audio stream to the left channel only and then mix it down to mono (or to digital stereo) with a piece of software like "Audacity" or something like that, so I simply did not consider it to be a problem at all.
A couple of days ago I decided to completely remove all my proprietary software from the system and replaced it with FOS analogues. That's when the problem started to get nasty.
First of all, I had a lot of troubles connecting to the qTox app. I had to manually re-plug the USB cable twice, three times or even more in a row each session until it started to work properly: sometimes I was hearing my mates like if their lunges were full of helium, sometimes I could not hear them at all, sometimes they couldn't hear me, etc. So I have done some research on the Internet and found a solution to make this thing work at least somehow. I've created a file ~/.asoundrc looking like this:
pcm.pulse { type pulse }
ctl.pulse { type pulse }
pcm.!default { type pulse }
ctl.!default { type pulse }
It partly solved an issue, but only if run qTox as the first application which uses an input-output sources. I am able to start other programs to work with my soundcard, but only after qTox is started. So it's quite suitable for me (at least because I couldn't find a better option to solve this).
But one great issue remains: my mates can hear me only in one of their output channels (left or right -- depending to which channel my mic is plugged in to). So I have done another research, but this time found... nothing. 90% of similar cases are about mixing down stereo to mono in the output channels, not about the mic. The remaining 10% made me completely messed up. I've been trying a lot of different ways to fix it, but ended up with no sound at all. Fortunately, I've managed to bring things back to the default (previous) state by reinstalling the pulseaudio with getting rid of all its config files.
However, I couldn't find a solution for the problem: my mates are still angry due to single-channel output of my voice.
So now I decided to ask the Community for help. I hope that my experience would be useful to someone with similar problem.
Thank you in advantage and I am sorry for my bad English. I hope you got the point.
Last edited by neostanniy (2021-05-19 10:10:42)
"Life is what happens to you while you're busy making other plans" © John Winston Ono Lennon.
Offline
It works the same way as with outputs but instead of a remap sink you'd use a remap source.
You'd define e.g. a
load-module module-remap-source source_name=mono_Mix master=$NAMEOFMASTERSOURCE master_channel_map=left channel_map=mono
at the end of your /etc/pulse/default.pa and then configure qtox or whatever in pavucontrol or so to record from that new source.
You'll find the name of the source you have to put in master= from pacmd list-sources
The description of how audio breaks when qtox is started or not makes me doubt on whether your pulse is normally started properly, what's your output of
systemctl --user status pulseaudio
arecord -lL
sudo fuser -v /dev/snd/*
pacmd list-cards
pacmd list-sources
pacmd list-source-outputs
while having such a recording session in qTox active?
Last edited by V1del (2021-05-18 13:41:47)
Online
OK, my output of
systemctl --user status pulseaudio
is
● pulseaudio.service - Sound Service
Loaded: loaded (/usr/lib/systemd/user/pulseaudio.service; disabled; vendor preset: enabled)
Active: active (running) since Tue 2021-05-18 14:19:39 EEST; 3h 8min ago
TriggeredBy: ● pulseaudio.socket
Main PID: 1024 (pulseaudio)
Tasks: 7 (limit: 19118)
Memory: 9.7M
CPU: 5min 3.402s
CGroup: /user.slice/user-1000.slice/user@1000.service/app.slice/pulseaudio.service
├─1024 /usr/bin/pulseaudio --daemonize=no --log-target=journal
└─1050 /usr/lib/pulse/gsettings-helper
тра 18 14:19:38 arch systemd[953]: Starting Sound Service...
тра 18 14:19:39 arch systemd[953]: Started Sound Service.
тра 18 14:19:39 arch pulseaudio[1024]: GetManagedObjects() failed: org.freedesktop.systemd1.NoSuchUnit: Unit dbus-org.bluez.service not found.
тра 18 15:38:37 arch pulseaudio[1024]: The 'master' module argument is deprecated and may be removed in the future, please use the 'sink_master' argument instead.
тра 18 16:43:39 arch pulseaudio[1024]: ALSA викликала службу запису нових даних на пристрій, але насправді ніяких даних для запису не виявлено!
тра 18 16:43:39 arch pulseaudio[1024]: Ймовірно, це пов’язано з вадою у драйвері ALSA «snd_usb_audio». Будь ласка, повідомте про цю ваду розробникам ALSA.
тра 18 16:43:39 arch pulseaudio[1024]: Службу було викликано зі встановленим POLLOUT, але наступний виклик snd_pcm_avail() повернув 0 або інше значення < min_avail.
~
~
~
~
[bohdan@arch ~]$ systemctl --user status pulseaudio
● pulseaudio.service - Sound Service
Loaded: loaded (/usr/lib/systemd/user/pulseaudio.service; disabled; vendor preset: enabled)
Active: active (running) since Tue 2021-05-18 14:19:39 EEST; 3h 10min ago
TriggeredBy: ● pulseaudio.socket
Main PID: 1024 (pulseaudio)
Tasks: 7 (limit: 19118)
Memory: 9.7M
CPU: 5min 4.658s
CGroup: /user.slice/user-1000.slice/user@1000.service/app.slice/pulseaudio.service
├─1024 /usr/bin/pulseaudio --daemonize=no --log-target=journal
└─1050 /usr/lib/pulse/gsettings-helper
тра 18 14:19:38 arch systemd[953]: Starting Sound Service...
тра 18 14:19:39 arch systemd[953]: Started Sound Service.
тра 18 14:19:39 arch pulseaudio[1024]: GetManagedObjects() failed: org.freedesktop.systemd1.NoSuchUnit: Unit dbus-org.bluez.service not found.
тра 18 15:38:37 arch pulseaudio[1024]: The 'master' module argument is deprecated and may be removed in the future, please use the 'sink_master' argument instead.
тра 18 16:43:39 arch pulseaudio[1024]: ALSA викликала службу запису нових даних на пристрій, але насправді ніяких даних для запису не виявлено!
тра 18 16:43:39 arch pulseaudio[1024]: Ймовірно, це пов’язано з вадою у драйвері ALSA «snd_usb_audio». Будь ласка, повідомте про цю ваду розробникам ALSA.
тра 18 16:43:39 arch pulseaudio[1024]: Службу було викликано зі встановленим POLLOUT, але наступний виклик snd_pcm_avail() повернув 0 або інше значення < min_avail.
The output of
arecord -lL
is:
null
Discard all samples (playback) or generate zero samples (capture)
sysdefault
Default Audio Device
samplerate
Rate Converter Plugin Using Samplerate Library
speexrate
Rate Converter Plugin Using Speex Resampler
jack
JACK Audio Connection Kit
oss
Open Sound System
pulse
PulseAudio Sound Server
speex
Plugin using Speex DSP (resample, agc, denoise, echo, dereverb)
upmix
Plugin for channel upmix (4,6,8)
vdownmix
Plugin for channel downmix (stereo) with a simple spacialization
sysdefault:CARD=VSL
AudioBox 22 VSL, USB Audio
Default Audio Device
front:CARD=VSL,DEV=0
AudioBox 22 VSL, USB Audio
Front output / input
usbstream:CARD=VSL
AudioBox 22 VSL
USB Stream Output
usbstream:CARD=NVidia
HDA NVidia
USB Stream Output
**** List of CAPTURE Hardware Devices ****
card 0: VSL [AudioBox 22 VSL], device 0: USB Audio [USB Audio]
Subdevices: 0/1
Subdevice #0: subdevice #0
The output of
sudo fuser -v /dev/snd/*
looks like this:
/dev/snd/controlC0: bohdan 1024 F.... pulseaudio
/dev/snd/controlC1: bohdan 1024 F.... pulseaudio
/dev/snd/pcmC0D0c: bohdan 1024 F...m pulseaudio
/dev/snd/pcmC0D0p: bohdan 1024 F...m pulseaudio
Those are the outputs for the last three commands:
2 card(s) available.
index: 0
name: <alsa_card.pci-0000_09_00.1>
driver: <module-alsa-card.c>
owner module: 6
properties:
alsa.card = "1"
alsa.card_name = "HDA NVidia"
alsa.long_card_name = "HDA NVidia at 0xf7080000 irq 84"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:09:00.1"
sysfs.path = "/devices/pci0000:00/0000:00:03.1/0000:09:00.1/sound/card1"
device.bus = "pci"
device.vendor.id = "10de"
device.vendor.name = "NVIDIA Corporation"
device.product.id = "1aeb"
device.product.name = "TU116 High Definition Audio Controller"
device.string = "1"
device.description = "TU116 High Definition Audio Controller"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
profiles:
output:hdmi-stereo: вихід Digital Stereo (HDMI) (priority 5900, available: no)
output:hdmi-surround: вихід Digital Surround 5.1 (HDMI) (priority 800, available: no)
output:hdmi-surround71: вихід Digital Surround 7.1 (HDMI) (priority 800, available: no)
output:hdmi-stereo-extra1: вихід Digital Stereo (HDMI 2) (priority 5700, available: unknown)
output:hdmi-stereo-extra2: вихід Digital Stereo (HDMI 3) (priority 5700, available: no)
output:hdmi-surround-extra2: вихід Digital Surround 5.1 (HDMI 3) (priority 600, available: no)
output:hdmi-surround71-extra2: вихід Digital Surround 7.1 (HDMI 3) (priority 600, available: no)
output:hdmi-stereo-extra3: вихід Digital Stereo (HDMI 4) (priority 5700, available: no)
output:hdmi-surround-extra3: вихід Digital Surround 5.1 (HDMI 4) (priority 600, available: no)
output:hdmi-surround71-extra3: вихід Digital Surround 7.1 (HDMI 4) (priority 600, available: no)
output:hdmi-stereo-extra4: вихід Digital Stereo (HDMI 5) (priority 5700, available: no)
output:hdmi-surround-extra4: вихід Digital Surround 5.1 (HDMI 5) (priority 600, available: no)
output:hdmi-surround71-extra4: вихід Digital Surround 7.1 (HDMI 5) (priority 600, available: no)
output:hdmi-stereo-extra5: вихід Digital Stereo (HDMI 6) (priority 5700, available: no)
output:hdmi-surround-extra5: вихід Digital Surround 5.1 (HDMI 6) (priority 600, available: no)
output:hdmi-surround71-extra5: вихід Digital Surround 7.1 (HDMI 6) (priority 600, available: no)
output:hdmi-stereo-extra6: вихід Digital Stereo (HDMI 7) (priority 5700, available: no)
output:hdmi-surround-extra6: вихід Digital Surround 5.1 (HDMI 7) (priority 600, available: no)
output:hdmi-surround71-extra6: вихід Digital Surround 7.1 (HDMI 7) (priority 600, available: no)
off: Вимкнено (priority 0, available: unknown)
active profile: <off>
ports:
hdmi-output-0: HDMI / DisplayPort (priority 5900, latency offset 0 usec, available: no)
properties:
device.icon_name = "video-display"
hdmi-output-1: HDMI / DisplayPort 2 (priority 5800, latency offset 0 usec, available: yes)
properties:
device.icon_name = "video-display"
device.product.name = "C27F390
"
hdmi-output-2: HDMI / DisplayPort 3 (priority 5700, latency offset 0 usec, available: no)
properties:
device.icon_name = "video-display"
hdmi-output-3: HDMI / DisplayPort 4 (priority 5600, latency offset 0 usec, available: no)
properties:
device.icon_name = "video-display"
hdmi-output-4: HDMI / DisplayPort 5 (priority 5500, latency offset 0 usec, available: no)
properties:
device.icon_name = "video-display"
hdmi-output-5: HDMI / DisplayPort 6 (priority 5400, latency offset 0 usec, available: no)
properties:
device.icon_name = "video-display"
hdmi-output-6: HDMI / DisplayPort 7 (priority 5300, latency offset 0 usec, available: no)
properties:
device.icon_name = "video-display"
index: 1
name: <alsa_card.usb-PreSonus_AudioBox_22_VSL_2214-00>
driver: <module-alsa-card.c>
owner module: 7
properties:
alsa.card = "0"
alsa.card_name = "AudioBox 22 VSL"
alsa.long_card_name = "PreSonus AudioBox 22 VSL at usb-0000:0a:00.3-2, high speed"
alsa.driver_name = "snd_usb_audio"
device.bus_path = "pci-0000:0a:00.3-usb-0:2:1.0"
sysfs.path = "/devices/pci0000:00/0000:00:07.1/0000:0a:00.3/usb5/5-2/5-2:1.0/sound/card0"
udev.id = "usb-PreSonus_AudioBox_22_VSL_2214-00"
device.bus = "usb"
device.vendor.id = "194f"
device.vendor.name = "PreSonus Audio Electronics, Inc."
device.product.id = "0101"
device.product.name = "AudioBox 22 VSL"
device.serial = "PreSonus_AudioBox_22_VSL_2214"
device.string = "0"
device.description = "AudioBox 22 VSL"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-usb"
profiles:
input:analog-stereo: вхід Аналогове стерео (priority 65, available: unknown)
input:iec958-stereo: вхід Цифрове стерео (IEC958) (priority 55, available: unknown)
output:analog-stereo: вихід Аналогове стерео (priority 6500, available: unknown)
output:analog-stereo+input:analog-stereo: Аналогове двобічне стерео (priority 6565, available: unknown)
output:analog-stereo+input:iec958-stereo: вихід Аналогове стерео + вхід Цифрове стерео (IEC958) (priority 6555, available: unknown)
output:iec958-stereo: вихід Цифрове стерео (IEC958) (priority 5500, available: unknown)
output:iec958-stereo+input:analog-stereo: вихід Цифрове стерео (IEC958) + вхід Аналогове стерео (priority 5565, available: unknown)
output:iec958-stereo+input:iec958-stereo: Цифрове двобічне стерео (IEC958) (priority 5555, available: unknown)
off: Вимкнено (priority 0, available: unknown)
active profile: <output:analog-stereo+input:analog-stereo>
sinks:
alsa_output.usb-PreSonus_AudioBox_22_VSL_2214-00.analog-stereo/#0: AudioBox 22 VSL Аналогове стерео
sources:
alsa_output.usb-PreSonus_AudioBox_22_VSL_2214-00.analog-stereo.monitor/#0: Monitor of AudioBox 22 VSL Аналогове стерео
alsa_input.usb-PreSonus_AudioBox_22_VSL_2214-00.analog-stereo/#1: AudioBox 22 VSL Аналогове стерео
ports:
analog-input-mic: Microphone (priority 8700, latency offset 0 usec, available: unknown)
properties:
device.icon_name = "audio-input-microphone"
iec958-stereo-input: Digital Input (S/PDIF) (priority 0, latency offset 0 usec, available: unknown)
properties:
analog-output: Analog Output (priority 9900, latency offset 0 usec, available: unknown)
properties:
iec958-stereo-output: Digital Output (S/PDIF) (priority 0, latency offset 0 usec, available: unknown)
properties:
2 source(s) available.
index: 0
name: <alsa_output.usb-PreSonus_AudioBox_22_VSL_2214-00.analog-stereo.monitor>
driver: <module-alsa-card.c>
flags: DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
state: IDLE
suspend cause: (none)
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: 6 KiB
sample spec: s32le 2кан. 44100Гц
channel map: front-left,front-right
Стерео
used by: 0
linked by: 0
configured latency: 2000,00 ms; range is 0,50 .. 2000,00 ms
monitor_of: 0
card: 1 <alsa_card.usb-PreSonus_AudioBox_22_VSL_2214-00>
module: 7
properties:
device.description = "Monitor of AudioBox 22 VSL Аналогове стерео"
device.class = "monitor"
alsa.card = "0"
alsa.card_name = "AudioBox 22 VSL"
alsa.long_card_name = "PreSonus AudioBox 22 VSL at usb-0000:0a:00.3-2, high speed"
alsa.driver_name = "snd_usb_audio"
device.bus_path = "pci-0000:0a:00.3-usb-0:2:1.0"
sysfs.path = "/devices/pci0000:00/0000:00:07.1/0000:0a:00.3/usb5/5-2/5-2:1.0/sound/card0"
udev.id = "usb-PreSonus_AudioBox_22_VSL_2214-00"
device.bus = "usb"
device.vendor.id = "194f"
device.vendor.name = "PreSonus Audio Electronics, Inc."
device.product.id = "0101"
device.product.name = "AudioBox 22 VSL"
device.serial = "PreSonus_AudioBox_22_VSL_2214"
device.string = "0"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-usb"
* index: 1
name: <alsa_input.usb-PreSonus_AudioBox_22_VSL_2214-00.analog-stereo>
driver: <module-alsa-card.c>
flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
state: RUNNING
suspend cause: (none)
priority: 9049
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: 6,78 ms
max rewind: 0 KiB
sample spec: s32le 2кан. 48000Гц
channel map: front-left,front-right
Стерео
used by: 1
linked by: 1
configured latency: 50,00 ms; range is 2,00 .. 1837,50 ms
card: 1 <alsa_card.usb-PreSonus_AudioBox_22_VSL_2214-00>
module: 7
properties:
alsa.resolution_bits = "32"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "USB Audio"
alsa.id = "USB Audio"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "0"
alsa.card_name = "AudioBox 22 VSL"
alsa.long_card_name = "PreSonus AudioBox 22 VSL at usb-0000:0a:00.3-2, high speed"
alsa.driver_name = "snd_usb_audio"
device.bus_path = "pci-0000:0a:00.3-usb-0:2:1.0"
sysfs.path = "/devices/pci0000:00/0000:00:07.1/0000:0a:00.3/usb5/5-2/5-2:1.0/sound/card0"
udev.id = "usb-PreSonus_AudioBox_22_VSL_2214-00"
device.bus = "usb"
device.vendor.id = "194f"
device.vendor.name = "PreSonus Audio Electronics, Inc."
device.product.id = "0101"
device.product.name = "AudioBox 22 VSL"
device.serial = "PreSonus_AudioBox_22_VSL_2214"
device.string = "front:0"
device.buffering.buffer_size = "705600"
device.buffering.fragment_size = "352800"
device.access_mode = "mmap+timer"
device.profile.name = "analog-stereo"
device.profile.description = "Аналогове стерео"
device.description = "AudioBox 22 VSL Аналогове стерео"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-usb"
ports:
analog-input-mic: Microphone (priority 8700, latency offset 0 usec, available: unknown)
properties:
device.icon_name = "audio-input-microphone"
active port: <analog-input-mic>
1 source output(s) available.
index: 6
driver: <protocol-native.c>
flags: START_CORKED
state: RUNNING
source: 1 <alsa_input.usb-PreSonus_AudioBox_22_VSL_2214-00.analog-stereo>
volume: front-left: 65536 / 100% / 0,00 dB, front-right: 65536 / 100% / 0,00 dB
balance 0,00
muted: no
current latency: 0,00 ms
requested latency: 50,00 ms
sample spec: s16le 2кан. 48000Гц
channel map: front-left,front-right
Стерео
resample method: copy
owner module: 12
client: 26 <qtox>
properties:
media.name = "Capture Stream"
application.name = "qtox"
native-protocol.peer = "UNIX socket client"
native-protocol.version = "34"
application.process.id = "1918"
application.process.user = "bohdan"
application.process.host = "arch"
application.process.binary = "qtox"
application.language = "uk_UA.utf8"
window.x11.display = ":1"
application.process.machine_id = "eb3257aeb8014161bc68faa77e9ca63e"
module-stream-restore.id = "source-output-by-application-name:qtox"
I runned all those commands while qtox was active. I am still confused, unfornunately.
"Life is what happens to you while you're busy making other plans" © John Winston Ono Lennon.
Offline
No the sidethings I wanted to check seem fine. qtox is using pulse and pulse is started normally.
So to test whether this will work at all try to do the remapping live:
pacmd load-module module-remap-source source_name=mono_Mix master=alsa_input.usb-PreSonus_AudioBox_22_VSL_2214-00.analog-stereo master_channel_map=front-left channel_map=mono
Replace the front-left with front-right depending on which one is the active audio channel. Once that is done you should see a new mono_Mix in pacmd list-sources/pavucontrol that you can move the qtox recorder to, in this case that would be
pacmd move-source-output 6 mono_Mix #That 6 changes everytime you start a new recording in qtox doublecheck with list-source-outputs
If you have something that works, you can add to the end of /etc/pulse/default.pa
.nofail
load-module module-remap-source source_name=mono_Mix master=alsa_input.usb-PreSonus_AudioBox_22_VSL_2214-00.analog-stereo master_channel_map=front-left channel_map=mono
and that should be considered on future boots.
Online
The command
pacmd move-source-output 6 mono_Mix
ended up with this:
No source output found with this index.
even despite I wasn't restarting the qTox session. So I tried to add
load-module module-remap-source source_name=mono_Mix master=alsa_input.usb-PreSonus_AudioBox_22_VSL_2214-00.analog-stereo master_channel_map=front-left channel_map=mono
and restarted the pulsed, but that didn't worked either. I even restarted the whole machine, tried to use qTox, then the "Audacity": still the same result. So I temporary commented those two lines.
Last edited by neostanniy (2021-05-18 20:16:19)
"Life is what happens to you while you're busy making other plans" © John Winston Ono Lennon.
Offline
the command failed because you didn't double check the index nr. You can also just have pavucontrol open and switch manually there. you should see a mono_mix sink on the output devices tab after loading the remap-module and be able to switch what qtox records from on the recording tab.
Online
the command failed because you didn't double check the index nr. You can also just have pavucontrol open and switch manually there. you should see a mono_mix sink on the output devices tab after loading the remap-module and be able to switch what qtox records from on the recording tab.
The point is I checked it and it was actually "6". But the worst thing is that now I don't have any sound at all... Well, it works on a gdm screen, but it doesn't — in the system. And my system doesn't shows anything, pavucontrol doesn't works.
"Life is what happens to you while you're busy making other plans" © John Winston Ono Lennon.
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As far as I can see, this happened because the system switched to "HDA Nvidia" ("Nvidia GPU 99 HDMI/DP") for some reason. I've opened alsamixer and it is shown as my default soundcard now.
"Life is what happens to you while you're busy making other plans" © John Winston Ono Lennon.
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None of the things I suggested should be able to have such an effect. Outputs of
systemctl --user status pulseaudio
sudo fuser -v /dev/snd/*
again?
Online
None of the things I suggested should be able to have such an effect. Outputs of
systemctl --user status pulseaudio sudo fuser -v /dev/snd/*
again?
First, I've tried the
pacmd list-sinks
and it said that there's no pulseaudio daemon running at all.
Then I've tried the
systemctl --user status pulseaudio
and it gave me this:
× pulseaudio.service - Sound Service
Loaded: loaded (/usr/lib/systemd/user/pulseaudio.service; disabled; vendor>
Active: failed (Result: exit-code) since Wed 2021-05-19 10:03:58 EEST; 6mi>
TriggeredBy: × pulseaudio.socket
Process: 1082 ExecStart=/usr/bin/pulseaudio --daemonize=no --log-target=jou>
Main PID: 1082 (code=exited, status=1/FAILURE)
CPU: 25ms
тра 19 10:03:58 arch systemd[973]: pulseaudio.service: Scheduled restart job, r>
тра 19 10:03:58 arch systemd[973]: Stopped Sound Service.
тра 19 10:03:58 arch systemd[973]: pulseaudio.service: Start request repeated t>
тра 19 10:03:58 arch systemd[973]: pulseaudio.service: Failed with result 'exit>
тра 19 10:03:58 arch systemd[973]: Failed to start Sound Service.
The output of
sudo fuser -v /dev/snd/*
is empty at all...
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I have also tried to run "Audacity" (just to see if my soundcard is still alive) and it actually works: I hear the normal clicking in both speakers for a couple of seconds and then I am actually able to start the recording with PreSonus AudioBox (single channel, as usual). I assume that the system changed the default audiosource for some reason, and I went to my gnome settings dialogue to try to switch it back, but it ain't showing any sound sources there. It's all empty.
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your pulseaudio isn't starting or it's crashing, how exactly did you try to comment the adjustment to /etc/pulse/default.pa (and where did you add them, they should be at the end) ? What's the output of
pulseaudio -vvv
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Trying to fix that issue still...
I have listed my cards with
cat /proc/asound/cards
and got
0 [NVidia ]: HDA-Intel - HDA NVidia
HDA NVidia at 0xf7080000 irq 84
1 [VSL ]: USB-Audio - AudioBox 22 VSL
PreSonus AudioBox 22 VSL at usb-0000:0a:00.3-2, high speed
So I decided to edit my ~/.asoundrc (because the sound works well at gdm screen, but isn't working inside my system after the user is logged in), and now it looks like this:
pcm.!default {
type hw
card 1
}
ctl.!default {
type hw
card 1
}
#pcm.pulse { type pulse }
#ctl.pulse { type pulse }
#pcm.!default { type pulse }
#ctl.!default { type pulse }
Still no result, and I am not sure if I am on the right way. So I will give up trying for now and waiting for Community to help. I am having an important online meeting tomorrow, so I am in a kind of a rush, but in case if it all won't be fixed till tomorrow I am about to create another user to log in for my conference. Do you think it might work?
(BTW, thank you so much, V1del, for trying to help. I appreciate that a lot!)
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your pulseaudio isn't starting or it's crashing, how exactly did you try to comment the adjustment to /etc/pulse/default.pa (and where did you add them, they should be at the end) ? What's the output of
pulseaudio -vvv
$ pulseaudio -vvv
I: [pulseaudio] main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Дію заборонено
I: [pulseaudio] main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Дію заборонено
D: [pulseaudio] core-rtclock.c: Timer slack is set to 50 us.
D: [pulseaudio] core-util.c: RealtimeKit worked.
I: [pulseaudio] core-util.c: Successfully gained nice level -11.
I: [pulseaudio] main.c: This is PulseAudio 14.2
D: [pulseaudio] main.c: Compilation CFLAGS: Not yet supported on meson
D: [pulseaudio] main.c: Running on host: Linux x86_64 5.12.4-arch1-2 #1 SMP PREEMPT Sat, 15 May 2021 20:58:02 +0000
D: [pulseaudio] main.c: Found 12 CPUs.
I: [pulseaudio] main.c: Page size is 4096 bytes
D: [pulseaudio] main.c: Compiled with Valgrind support: no
D: [pulseaudio] main.c: Running in valgrind mode: no
D: [pulseaudio] main.c: Running in VM: no
D: [pulseaudio] main.c: Running from build tree: no
D: [pulseaudio] main.c: Optimized build: yes
D: [pulseaudio] main.c: All asserts enabled.
I: [pulseaudio] main.c: Machine ID is eb3257aeb8014161bc68faa77e9ca63e.
I: [pulseaudio] main.c: Using runtime directory /run/user/1000/pulse.
I: [pulseaudio] main.c: Using state directory /home/bohdan/.config/pulse.
I: [pulseaudio] main.c: Using modules directory /usr/lib/pulse-14.2/modules.
I: [pulseaudio] main.c: Running in system mode: no
I: [pulseaudio] main.c: System supports high resolution timers
D: [pulseaudio] log.c: Invalid UTF-8 string following below:
D: [pulseaudio] memblock.c: Using shared memfd memory pool with 1024 slots of size 64,0 Кі each, total size is 64,0 Мі, maximum usable slot size is 65472
I: [pulseaudio] cpu-x86.c: CPU flags: CMOV MMX SSE SSE2 SSE3 SSSE3 SSE4_1 SSE4_2 MMXEXT
I: [pulseaudio] svolume_mmx.c: Initialising MMX optimized volume functions.
I: [pulseaudio] remap_mmx.c: Initialising MMX optimized remappers.
I: [pulseaudio] svolume_sse.c: Initialising SSE2 optimized volume functions.
I: [pulseaudio] remap_sse.c: Initialising SSE2 optimized remappers.
I: [pulseaudio] sconv_sse.c: Initialising SSE2 optimized conversions.
I: [pulseaudio] svolume_orc.c: Initialising ORC optimized volume functions.
E: [pulseaudio] main.c: Unknown command: \#!/usr/bin/pulseaudio -nF
E: [pulseaudio] main.c: Не вдалося ініціалізувати фонову службу через помилки під час виконання команд запуску. Джерело команд: /etc/pulse/default.pa
I: [pulseaudio] main.c: Daemon terminated.
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And here's a tail of my /etc/pulse/default.pa:
$ cat /etc/pulse/default.pa | tail
### make sense, and module-filter-apply does the heavy-lifting of
### loading modules and rerouting streams.
load-module module-filter-heuristics
load-module module-filter-apply
### Make some devices default
#set-default-sink output
#set-default-source input
.nofail
load-module module-remap-source source_name=mono_Mix master=alsa_input.usb-PreSonus_AudioBox_22_VSL_2214-00.analog-stereo master_channel_map=front-left channel_map=mono
I've tried both commenting and uncommenting the last two lines with the same result.
Last edited by neostanniy (2021-05-19 07:43:00)
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Please post that in english by prepending LC_ALL=C to the command
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Please post that in english by prepending LC_ALL=C to the command
Sure, I apologize for that. By bad, sir.
$ LC_ALL=C pulseaudio -vvv
I: [pulseaudio] main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
I: [pulseaudio] main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
D: [pulseaudio] core-rtclock.c: Timer slack is set to 50 us.
D: [pulseaudio] core-util.c: RealtimeKit worked.
I: [pulseaudio] core-util.c: Successfully gained nice level -11.
I: [pulseaudio] main.c: This is PulseAudio 14.2
D: [pulseaudio] main.c: Compilation CFLAGS: Not yet supported on meson
D: [pulseaudio] main.c: Running on host: Linux x86_64 5.12.4-arch1-2 #1 SMP PREEMPT Sat, 15 May 2021 20:58:02 +0000
D: [pulseaudio] main.c: Found 12 CPUs.
I: [pulseaudio] main.c: Page size is 4096 bytes
D: [pulseaudio] main.c: Compiled with Valgrind support: no
D: [pulseaudio] main.c: Running in valgrind mode: no
D: [pulseaudio] main.c: Running in VM: no
D: [pulseaudio] main.c: Running from build tree: no
D: [pulseaudio] main.c: Optimized build: yes
D: [pulseaudio] main.c: All asserts enabled.
I: [pulseaudio] main.c: Machine ID is eb3257aeb8014161bc68faa77e9ca63e.
I: [pulseaudio] main.c: Using runtime directory /run/user/1000/pulse.
I: [pulseaudio] main.c: Using state directory /home/bohdan/.config/pulse.
I: [pulseaudio] main.c: Using modules directory /usr/lib/pulse-14.2/modules.
I: [pulseaudio] main.c: Running in system mode: no
I: [pulseaudio] main.c: System supports high resolution timers
D: [pulseaudio] memblock.c: Using shared memfd memory pool with 1024 slots of size 64.0 KiB each, total size is 64.0 MiB, maximum usable slot size is 65472
I: [pulseaudio] cpu-x86.c: CPU flags: CMOV MMX SSE SSE2 SSE3 SSSE3 SSE4_1 SSE4_2 MMXEXT
I: [pulseaudio] svolume_mmx.c: Initialising MMX optimized volume functions.
I: [pulseaudio] remap_mmx.c: Initialising MMX optimized remappers.
I: [pulseaudio] svolume_sse.c: Initialising SSE2 optimized volume functions.
I: [pulseaudio] remap_sse.c: Initialising SSE2 optimized remappers.
I: [pulseaudio] sconv_sse.c: Initialising SSE2 optimized conversions.
I: [pulseaudio] svolume_orc.c: Initialising ORC optimized volume functions.
E: [pulseaudio] main.c: Unknown command: \#!/usr/bin/pulseaudio -nF
E: [pulseaudio] main.c: Failed to initialize daemon due to errors while executing startup commands. Source of commands: /etc/pulse/default.pa
I: [pulseaudio] main.c: Daemon terminated.
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E: [pulseaudio] main.c: Unknown command: \#!/usr/bin/pulseaudio -nF
It looks like there is a stray backslash at the start of /etc/pulse/default.pa. You should remove it, the first line should look like:
#!/usr/bin/pulseaudio -nF
| alias CUTF='LANG=en_XX.UTF-8@POSIX ' |
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E: [pulseaudio] main.c: Unknown command: \#!/usr/bin/pulseaudio -nF
It looks like there is a stray backslash at the start of /etc/pulse/default.pa. You should remove it, the first line should look like:
#!/usr/bin/pulseaudio -nF
I feel like a complete idiot now. El idiotto... Will do and reboot.
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Yes, that was the reason! We've got the sound back working now.
I am terribly sorry for being such a noob...
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However, the problem with single channel (which was the initial in the post) remains.
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Now that pulseaudio is running again, you should be able to load the remap-source module as mentioned by V1del in post #4
Last edited by progandy (2021-05-19 09:37:08)
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Thanks all and excuse my noobiness (is there such a word? :-) ) with that "/". That what happens when you haven't read the "How to Exit Vim" book.
Now I am bout to re-follow the instructions mentioned in #4 comment. However, I am not sure if I hadn't messed up my ~/.asoundrc file. Right now it looks like this:
$ cat ~/.asoundrc
#pcm.!default {
# type hw
# card 1
#}
#
#ctl.!default {
# type hw
# card 1
#}
pcm.pulse { type pulse }
ctl.pulse { type pulse }
pcm.!default { type pulse }
ctl.!default { type pulse }
Should I do smth with it before taking any further actions or should I keep it as it is?
Thank you very much!
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that can stay as is, but again to just test whether this works you can without restarting pulse use pacmd to load the module-remap-source with different parameters (just the name has to be different or you need to unload previous attempts via the number that pacmd will return on successful module load) once you have a sink you should be able to switch qtox to that in pavucontrol.
Last edited by V1del (2021-05-19 09:36:30)
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The asoundrc is fine. If you want to you could also delete it and install pulseaudio-alsa, that should do basically the same for all users.
Last edited by progandy (2021-05-19 09:37:48)
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