How do I fix microphone issues with Pulseaudio and ALC1220?
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I've been trying to get the microphone to work on a new PC, but Pulseaudio produces distorted audio, that sounds as if it's sped-up.
I've observed this on Audacity and Discord. Audacity in particular will only record around a quarter of a second of audio every 6 seconds or so, and stopping the recording before that time elapses will discard all audio information. On longer recordings, any information that didn't fall inside those ~6 seconds recording intervals will also be discarded.
If I try to get audio through ALSA on Audacity, it seems to work, albeit with some low background static noise. If I use arecord, it produces a mostly clean recording with regular popping/crackling noises.
After that, switching back to pulse on Audacity will produce normal speed audio but with a crackling noise at the beginning of the recording that gradually diminishes into background noise with constant popping.
Using the tsched=0 option in Pulseaudio's configuration file or manually setting the sampling rate in the daemon configuration don't fix the issue.
Is there any way to fix this or is this particular sound card just buggy on Linux?
Motherboard: ASUS ROG STRIX X470-F GAMING
Output of arecord --list-devices
:
**** List of CAPTURE Hardware Devices ****
card 1: Generic [HD-Audio Generic], device 0: ALC1220 Analog [ALC1220 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: Generic [HD-Audio Generic], device 2: ALC1220 Alt Analog [ALC1220 Alt Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
Output of uname -r
:
4.19.3-300.fc29.x86_64
pulseaudio realtek microphone
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I've been trying to get the microphone to work on a new PC, but Pulseaudio produces distorted audio, that sounds as if it's sped-up.
I've observed this on Audacity and Discord. Audacity in particular will only record around a quarter of a second of audio every 6 seconds or so, and stopping the recording before that time elapses will discard all audio information. On longer recordings, any information that didn't fall inside those ~6 seconds recording intervals will also be discarded.
If I try to get audio through ALSA on Audacity, it seems to work, albeit with some low background static noise. If I use arecord, it produces a mostly clean recording with regular popping/crackling noises.
After that, switching back to pulse on Audacity will produce normal speed audio but with a crackling noise at the beginning of the recording that gradually diminishes into background noise with constant popping.
Using the tsched=0 option in Pulseaudio's configuration file or manually setting the sampling rate in the daemon configuration don't fix the issue.
Is there any way to fix this or is this particular sound card just buggy on Linux?
Motherboard: ASUS ROG STRIX X470-F GAMING
Output of arecord --list-devices
:
**** List of CAPTURE Hardware Devices ****
card 1: Generic [HD-Audio Generic], device 0: ALC1220 Analog [ALC1220 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: Generic [HD-Audio Generic], device 2: ALC1220 Alt Analog [ALC1220 Alt Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
Output of uname -r
:
4.19.3-300.fc29.x86_64
pulseaudio realtek microphone
There's a bugtracker issue for that, but no solution. My guess would be that the codec needs a quirk, probably some additional initialisation. There's no public data sheet for the ALC1220, so making one will be difficult, and needs someone with kernel hacking and codec experience. Comparing the codec state with the state under Windows with a working driver may help.
– dirkt
Nov 27 at 7:59
add a comment |
up vote
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down vote
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up vote
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down vote
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I've been trying to get the microphone to work on a new PC, but Pulseaudio produces distorted audio, that sounds as if it's sped-up.
I've observed this on Audacity and Discord. Audacity in particular will only record around a quarter of a second of audio every 6 seconds or so, and stopping the recording before that time elapses will discard all audio information. On longer recordings, any information that didn't fall inside those ~6 seconds recording intervals will also be discarded.
If I try to get audio through ALSA on Audacity, it seems to work, albeit with some low background static noise. If I use arecord, it produces a mostly clean recording with regular popping/crackling noises.
After that, switching back to pulse on Audacity will produce normal speed audio but with a crackling noise at the beginning of the recording that gradually diminishes into background noise with constant popping.
Using the tsched=0 option in Pulseaudio's configuration file or manually setting the sampling rate in the daemon configuration don't fix the issue.
Is there any way to fix this or is this particular sound card just buggy on Linux?
Motherboard: ASUS ROG STRIX X470-F GAMING
Output of arecord --list-devices
:
**** List of CAPTURE Hardware Devices ****
card 1: Generic [HD-Audio Generic], device 0: ALC1220 Analog [ALC1220 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: Generic [HD-Audio Generic], device 2: ALC1220 Alt Analog [ALC1220 Alt Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
Output of uname -r
:
4.19.3-300.fc29.x86_64
pulseaudio realtek microphone
I've been trying to get the microphone to work on a new PC, but Pulseaudio produces distorted audio, that sounds as if it's sped-up.
I've observed this on Audacity and Discord. Audacity in particular will only record around a quarter of a second of audio every 6 seconds or so, and stopping the recording before that time elapses will discard all audio information. On longer recordings, any information that didn't fall inside those ~6 seconds recording intervals will also be discarded.
If I try to get audio through ALSA on Audacity, it seems to work, albeit with some low background static noise. If I use arecord, it produces a mostly clean recording with regular popping/crackling noises.
After that, switching back to pulse on Audacity will produce normal speed audio but with a crackling noise at the beginning of the recording that gradually diminishes into background noise with constant popping.
Using the tsched=0 option in Pulseaudio's configuration file or manually setting the sampling rate in the daemon configuration don't fix the issue.
Is there any way to fix this or is this particular sound card just buggy on Linux?
Motherboard: ASUS ROG STRIX X470-F GAMING
Output of arecord --list-devices
:
**** List of CAPTURE Hardware Devices ****
card 1: Generic [HD-Audio Generic], device 0: ALC1220 Analog [ALC1220 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: Generic [HD-Audio Generic], device 2: ALC1220 Alt Analog [ALC1220 Alt Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
Output of uname -r
:
4.19.3-300.fc29.x86_64
pulseaudio realtek microphone
pulseaudio realtek microphone
edited Nov 26 at 22:21
asked Nov 26 at 19:53
Aeder
134
134
There's a bugtracker issue for that, but no solution. My guess would be that the codec needs a quirk, probably some additional initialisation. There's no public data sheet for the ALC1220, so making one will be difficult, and needs someone with kernel hacking and codec experience. Comparing the codec state with the state under Windows with a working driver may help.
– dirkt
Nov 27 at 7:59
add a comment |
There's a bugtracker issue for that, but no solution. My guess would be that the codec needs a quirk, probably some additional initialisation. There's no public data sheet for the ALC1220, so making one will be difficult, and needs someone with kernel hacking and codec experience. Comparing the codec state with the state under Windows with a working driver may help.
– dirkt
Nov 27 at 7:59
There's a bugtracker issue for that, but no solution. My guess would be that the codec needs a quirk, probably some additional initialisation. There's no public data sheet for the ALC1220, so making one will be difficult, and needs someone with kernel hacking and codec experience. Comparing the codec state with the state under Windows with a working driver may help.
– dirkt
Nov 27 at 7:59
There's a bugtracker issue for that, but no solution. My guess would be that the codec needs a quirk, probably some additional initialisation. There's no public data sheet for the ALC1220, so making one will be difficult, and needs someone with kernel hacking and codec experience. Comparing the codec state with the state under Windows with a working driver may help.
– dirkt
Nov 27 at 7:59
add a comment |
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There's a bugtracker issue for that, but no solution. My guess would be that the codec needs a quirk, probably some additional initialisation. There's no public data sheet for the ALC1220, so making one will be difficult, and needs someone with kernel hacking and codec experience. Comparing the codec state with the state under Windows with a working driver may help.
– dirkt
Nov 27 at 7:59