I am using Android’s webview for a webrtc video application. I am observing a behaviour where device audio is getting reduced when the remote video starts playing.
Is there any way to avoid ducking in this scenario ? Also are there any instances where the audio could be less even if
- device outbound bitrate is high
- network is good at both sides
- both devices are functioning well
How to resolve this issue from android side ? If i try to use volume for stream, i am not able to get the current stream, but can set to stream_voice_call based on the audioManager api to check.
Can any settings to be made on webview side as well ?