WebRTC: Is it possible to control the microphone and volume levels

Question

I am working on a demo site which includes a slide-out widget that allows a user to place a call.

I am using the SIPml5 tool along with the webrtc2sip back end for h

Answer 1

Well it is possible, but currently only in Chrome and with some assumptions. I am not the auther, you can find inspiration for this code in this open-source library (SimpleWebRtc).

navigator.webkitGetUserMedia(constraints, 
    function(webRTCStream){
        var context = new window.AudioContext();
        var microphone = context.createMediaStreamSource(webRTCStream);
        var gainFilter = context.createGain();
        var destination = context.createMediaStreamDestination();
        var outputStream = destination.stream;
        microphone.connect(gainFilter);
        gainFilter.connect(destination);
        var filteredTrack = outputStream.getAudioTracks()[0];
        webRTCStream.addTrack(filteredTrack);
        var originalTrack = webRTCStream.getAudioTracks()[0];
        webRTCStream.removeTrack(originalTrack);
    },
    function(err) {
        console.log("The following error occured: " + err);
      }
 );

The trick is to modify the stream and then replace the audio track of current stream with audio track of modified stream (taken from MediaStreamDestination stream).

DISCLAIMER:

This doesn't work in FireFox as of version 35, since they merely didn't implement MediaStream.addTrack/removeTrack. I use this check currently

  this.micVolumeIsSupported = function() {
    var MediaStream = window.webkitMediaStream || window.MediaStream;
    return !!MediaStream.prototype.addTrack && !!MediaStream.prototype.removeTrack;
  };
_gainSupported = this.micVolumeIsSupported();

This has a limitation in Chrome due to a bug with stopping stream with mixed up tracks. You might wish to restore these tracks before closing connection or on connection interruption;

this.restoreTracks = function(){
  if(_gainSupported && _tracksSubstituted){
    webRTCStream.addTrack(originalTrack);
    webRTCStream.removeTrack(filteredTrack);
    _tracksSubstituted = false;
  }
};

This works for me