You signed in with another tab or window. Reload to refresh your session.You signed out in another tab or window. Reload to refresh your session.You switched accounts on another tab or window. Reload to refresh your session.Dismiss alert
I'm currently working on a project that uses WebRTC for real-time audio streaming. However, I'm having trouble with the audio quality - it seems like there's some noise suppression or other effect applied that is reducing the quality.
I've tried setting the googAutoGainControl and googNoiseSuppression flags to false in the configurationPeerConnection object, but that doesn't seem to have any effect. I've also tried increasing the audio bitrate, but that didn't help either.
Can anyone provide guidance on how to disable the noise suppression and other effects, and ensure that the audio is being sent in high quality? Any help would be much appreciated.
Thanks!
The text was updated successfully, but these errors were encountered:
For noise reduction, you can use the ffmpeg, with audio filter denoise filter . I've never combined webrtc with ffmpeg, but it should work, you can research about it
Hi there,
I'm currently working on a project that uses WebRTC for real-time audio streaming. However, I'm having trouble with the audio quality - it seems like there's some noise suppression or other effect applied that is reducing the quality.
I've tried setting the googAutoGainControl and googNoiseSuppression flags to false in the configurationPeerConnection object, but that doesn't seem to have any effect. I've also tried increasing the audio bitrate, but that didn't help either.
Can anyone provide guidance on how to disable the noise suppression and other effects, and ensure that the audio is being sent in high quality? Any help would be much appreciated.
Thanks!
The text was updated successfully, but these errors were encountered: