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Hi, I installed pulseaudio because of my problems with plain alsa in chrome. Now sound output is working fine, but I have another problem. My microphone is recording all output (for example if I play some youtube video into my headphones, this output is also recorded by input headphones microphone). Is there a way to disable this using pusleaudio? When I was using alsa, there was output mixing setting to disable it, but I can't see anything similar in pulse (im using ncpamixer). Thanks a lot
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Pulseaudio allocates a monitoring source for each output which you can mute if you don't want that, however this wouldn't inherently bleed into any actual microphone node.
What's your
amixer -c0 #Or whatever your real card's index is according to arecord -l
pacmd list-sources
pacmd list-source-outputs
while noticing this issue
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amixer -c0:
Simple mixer control 'Master',0
Capabilities: pvolume pvolume-joined pswitch pswitch-joined
Playback channels: Mono
Limits: Playback 0 - 87
Mono: Playback 79 [91%] [-6.00dB] [on]
Simple mixer control 'Headphone',0
Capabilities: pvolume pswitch
Playback channels: Front Left - Front Right
Limits: Playback 0 - 87
Mono:
Front Left: Playback 87 [100%] [0.00dB] [on]
Front Right: Playback 87 [100%] [0.00dB] [on]
Simple mixer control 'Speaker',0
Capabilities: pvolume pswitch
Playback channels: Front Left - Front Right
Limits: Playback 0 - 87
Mono:
Front Left: Playback 0 [0%] [-65.25dB] [off]
Front Right: Playback 0 [0%] [-65.25dB] [off]
Simple mixer control 'PCM',0
Capabilities: pvolume
Playback channels: Front Left - Front Right
Limits: Playback 0 - 255
Mono:
Front Left: Playback 255 [100%] [0.00dB]
Front Right: Playback 255 [100%] [0.00dB]
Simple mixer control 'Beep',0
Capabilities: pvolume pswitch
Playback channels: Front Left - Front Right
Limits: Playback 0 - 31
Mono:
Front Left: Playback 0 [0%] [-34.50dB] [on]
Front Right: Playback 0 [0%] [-34.50dB] [on]
Simple mixer control 'Capture',0
Capabilities: cvolume cswitch
Capture channels: Front Left - Front Right
Limits: Capture 0 - 63
Front Left: Capture 63 [100%] [30.00dB] [on]
Front Right: Capture 63 [100%] [30.00dB] [on]
Simple mixer control 'Auto-Mute Mode',0
Capabilities: enum
Items: 'Disabled' 'Enabled'
Item0: 'Disabled'
Simple mixer control 'Digital',0
Capabilities: cvolume
Capture channels: Front Left - Front Right
Limits: Capture 0 - 120
Front Left: Capture 53 [44%] [-3.50dB]
Front Right: Capture 53 [44%] [-3.50dB]
Simple mixer control 'Headset Mic',0
Capabilities: pvolume pswitch cswitch cswitch-joined cswitch-exclusive
Capture exclusive group: 0
Playback channels: Front Left - Front Right
Capture channels: Mono
Limits: Playback 0 - 31
Mono: Capture [on]
Front Left: Playback 31 [100%] [12.00dB] [on]
Front Right: Playback 31 [100%] [12.00dB] [on]
Simple mixer control 'Headset Mic Boost',0
Capabilities: volume
Playback channels: Front Left - Front Right
Capture channels: Front Left - Front Right
Limits: 0 - 3
Front Left: 2 [67%] [20.00dB]
Front Right: 2 [67%] [20.00dB]
Simple mixer control 'Internal Mic',0
Capabilities: cswitch cswitch-joined cswitch-exclusive
Capture exclusive group: 0
Capture channels: Mono
Mono: Capture [off]
Simple mixer control 'Internal Mic Boost',0
Capabilities: volume
Playback channels: Front Left - Front Right
Capture channels: Front Left - Front Right
Limits: 0 - 3
Front Left: 0 [0%] [0.00dB]
Front Right: 0 [0%] [0.00dB]
Simple mixer control 'Loopback Mixing',0
Capabilities: enum
Items: 'Disabled' 'Enabled'
Item0: 'Disabled'
pacmd list-sources:
3 source(s) available.
index: 0
name: <alsa_output.pci-0000_01_00.1.hdmi-stereo.monitor>
driver: <module-alsa-card.c>
flags: DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
state: SUSPENDED
suspend cause: IDLE
priority: 1030
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 .. 1999.82 ms
monitor_of: 0
card: 0 <alsa_card.pci-0000_01_00.1>
module: 6
properties:
device.description = "Monitor of GP106 High Definition Audio Controller Digital Stereo (HDMI)"
device.class = "monitor"
alsa.card = "1"
alsa.card_name = "HDA NVidia"
alsa.long_card_name = "HDA NVidia at 0xa4000000 irq 17"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:01:00.1"
sysfs.path = "/devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1"
device.bus = "pci"
device.vendor.id = "10de"
device.vendor.name = "NVIDIA Corporation"
device.product.id = "10f1"
device.product.name = "GP106 High Definition Audio Controller"
device.string = "1"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
index: 1
name: <alsa_output.pci-0000_00_1f.3.analog-stereo.monitor>
driver: <module-alsa-card.c>
flags: DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
state: SUSPENDED
suspend cause: IDLE
priority: 1030
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: yes
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: 1
card: 1 <alsa_card.pci-0000_00_1f.3>
module: 7
properties:
device.description = "Monitor of Built-in Audio Analog Stereo"
device.class = "monitor"
alsa.card = "0"
alsa.card_name = "HDA Intel PCH"
alsa.long_card_name = "HDA Intel PCH at 0xa4414000 irq 148"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:00:1f.3"
sysfs.path = "/devices/pci0000:00/0000:00:1f.3/sound/card0"
device.bus = "pci"
device.vendor.id = "8086"
device.vendor.name = "Intel Corporation"
device.product.id = "a348"
device.product.name = "Cannon Lake PCH cAVS"
device.form_factor = "internal"
device.string = "0"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
* index: 2
name: <alsa_input.pci-0000_00_1f.3.analog-stereo>
driver: <module-alsa-card.c>
flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
state: SUSPENDED
suspend cause: IDLE
priority: 9039
volume: front-left: 44652 / 68% / -10.00 dB, front-right: 44652 / 68% / -10.00 dB
balance 0.00
base volume: 6554 / 10% / -60.00 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
card: 1 <alsa_card.pci-0000_00_1f.3>
module: 7
properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "ALC255 Analog"
alsa.id = "ALC255 Analog"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "0"
alsa.card_name = "HDA Intel PCH"
alsa.long_card_name = "HDA Intel PCH at 0xa4414000 irq 148"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:00:1f.3"
sysfs.path = "/devices/pci0000:00/0000:00:1f.3/sound/card0"
device.bus = "pci"
device.vendor.id = "8086"
device.vendor.name = "Intel Corporation"
device.product.id = "a348"
device.product.name = "Cannon Lake PCH cAVS"
device.form_factor = "internal"
device.string = "front:0"
device.buffering.buffer_size = "352800"
device.buffering.fragment_size = "176400"
device.access_mode = "mmap+timer"
device.profile.name = "analog-stereo"
device.profile.description = "Analog Stereo"
device.description = "Built-in Audio Analog Stereo"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
ports:
analog-input-internal-mic: Internal Microphone (priority 8900, latency offset 0 usec, available: unknown)
properties:
device.icon_name = "audio-input-microphone"
analog-input-headset-mic: Headset Microphone (priority 8800, latency offset 0 usec, available: unknown)
properties:
device.icon_name = "audio-input-microphone"
active port: <analog-input-headset-mic>
pacmd list-source-outputs:
1 source output(s) available.
index: 11
driver: <protocol-native.c>
flags: START_CORKED
state: RUNNING
source: 2 <alsa_input.pci-0000_00_1f.3.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: 10.00 ms
sample spec: s16le 2ch 44100Hz
channel map: front-left,front-right
Stereo
resample method: (null)
owner module: 9
client: 2 <Google Chrome input>
properties:
application.icon_name = "google-chrome"
media.name = "RecordStream"
application.name = "Google Chrome input"
native-protocol.peer = "UNIX socket client"
native-protocol.version = "34"
application.process.id = "1385"
application.process.user = "rstasta"
application.process.host = "archlinux"
application.process.binary = "chrome"
application.language = "en_US.UTF-8"
window.x11.display = ":0"
application.process.machine_id = "734f4c878f9b4670afe24735be4111e5"
application.process.session_id = "1"
module-stream-restore.id = "source-output-by-application-name:Google Chrome input"
Last edited by Raadush (2021-07-26 12:05:23)
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Looks normalish, is this actually directly receiving the audio feed or is this just a matter of lowering the sensitivity?
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Directly. I have output in headphones so microphone is not hearing anything.
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If that sound card has hardware loopback, then maybe pulseaudio does not have controls for that. You might have to use alsa to disable it.
Simple mixer control 'Loopback Mixing',0 seems to be disabled
Simple mixer control 'Capture',0 seems to be active, maybe try to mute that with amixer?
| alias CUTF='LANG=en_XX.UTF-8@POSIX ' |
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I was using alsa, but recentrly, google chrome stopped working properly with alsa, so I had to install pulseaudio
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pulseaudio uses the alsa drivers as well, so you can still use amixer to mute channels that are not visible in pulseaudio.
| alias CUTF='LANG=en_XX.UTF-8@POSIX ' |
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In amixer I have loopback disabled as you can see in logs, I have muted all other channels that are not needed, but problem is still there. For example when I use skype, I can't mute Skype, because I want to hear something, but when I unmute, all skype voices are recorded back, so my contacts are getting echo since I record what they are saying and am sending it back through my mic
Last edited by Raadush (2021-08-02 11:31:52)
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Well the question here is whether the Capture control is refering to a loopback or your actual capture, this is sometimes hard to tell just from the outputs, what's your current amixer -c0 output?
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