message: Tweak a bunch of settings
diff: @@ -26,10 +26,15 @@ export class VoiceRecorder {
  mediaTrackConstraints: <MediaTrackConstraints>{
  deviceId: CallMediaHandler.getAudioInput(),
  },
  - encoderSampleRate: 16000, // we could go down to 12khz, but we lose quality
  + encoderSampleRate: 48000, // we could go down to 12khz, but we lose quality. 48khz is a webrtc default
  encoderApplication: 2048, // voice (default is "audio")
  streamPages: true, // so we can have a live EQ for the user
  - encoderFrameSize: 10, // we want updates fairly regularly for the UI
  + encoderFrameSize: 20, // ms, we want updates fairly regularly for the UI
  + numberOfChannels: 1, // stereo isn't important for us
  + //sourceNode: instanceof MediaStreamAudioSourceNode, // TODO: @@ Travis: Use this for EQ stuff.
  + encoderBitRate: 64000, // 64kbps is average for webrtc
  + encoderComplexity: 3, // 0-10, 0 is fast and low complexity
  + resampleQuality: 3, // 0-10, 10 is slow and high quality
  });
  private buffer = new Uint8Array(0);
  private mxc: string;
  