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!
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!