Is there a way to choose codecs in WebRTC PeerConnection?

问题

Hey I would like to know if there is any way to choose codecs when creating the offer/answer in WebRTC. There are currently not many video codecs to choose from, but there are audio codecs like Opus, PCMU, PCMA etc.

回答1:

In general, yes. Here is example of how to prefer Opus codec during establishing conneciton. You should call 'preferOpus' from a callback function for createAnswer or createOffer.

var preferOpus = function(sdp) {
  var sdpLines = sdp.split('\r\n');

  for (var i = 0; i < sdpLines.length; i++) {
    if (sdpLines[i].search('m=audio') !== -1) {
      var mLineIndex = i;
      break;
    }
  }

  if (mLineIndex === null) return sdp;

  for (i = 0; i < sdpLines.length; i++) {
    if (sdpLines[i].search('opus/48000') !== -1) {
      var opusPayload = extractSdp(sdpLines[i], /:(\d+) opus\/48000/i);
      if (opusPayload) 
        sdpLines[mLineIndex] = setDefaultCodec(sdpLines[mLineIndex], opusPayload);
      break;
    }
  }

  sdpLines = removeCN(sdpLines, mLineIndex);

  sdp = sdpLines.join('\r\n');
  return sdp;
};

var extractSdp = function(sdpLine, pattern) {
  var result = sdpLine.match(pattern);
  return (result && result.length == 2)? result[1]: null;
};

var setDefaultCodec = function(mLine, payload) {
  var elements = mLine.split(' ');
  var newLine = new Array();
  var index = 0;
  for (var i = 0; i < elements.length; i++) {
    if (index === 3) newLine[index++] = payload;
    if (elements[i] !== payload) newLine[index++] = elements[i];
  }
  return newLine.join(' ');
};

var removeCN = function(sdpLines, mLineIndex) {
  var mLineElements = sdpLines[mLineIndex].split(' ');
  for (var i = sdpLines.length-1; i >= 0; i--) {
    var payload = extractSdp(sdpLines[i], /a=rtpmap:(\d+) CN\/\d+/i);
    if (payload) {
      var cnPos = mLineElements.indexOf(payload);
      if (cnPos !== -1) mLineElements.splice(cnPos, 1);
      sdpLines.splice(i, 1);
    }
  }
  sdpLines[mLineIndex] = mLineElements.join(' ');
  return sdpLines;
};

回答2:

Choosing Opus will only get you halfway there. Event with the codec it might default to mono and around 42 kb/s as it's primarily designed for voice.

If you are not using voice input, and want consistent music, you can disable the audio processing features using constraints:

navigator.mediaDevices.getUserMedia({
  audio: {
    autoGainControl: false,
    channelCount: 2,
    echoCancellation: false,
    latency: 0,
    noiseSuppression: false,
    sampleRate: 48000,
    sampleSize: 16,
    volume: 1.0
  }
});

Then set the SDP to be stereo and increase the maxaveragebitrate:

let answer = await peer.conn.createAnswer(offerOptions);
answer.sdp = answer.sdp.replace('useinbandfec=1', 'useinbandfec=1; stereo=1; maxaveragebitrate=510000');
await peer.conn.setLocalDescription(answer);

This should output a string which looks like this:

a=fmtp:111 minptime=10;useinbandfec=1; stereo=1; maxaveragebitrate=510000

This gives a potential maximum bitrate of 520kb/s for stereo, which is 260kps per channel. Actual bitrate depends on the speed of your network and strength of your signal.

You can read more about the other available attributes at: https://tools.ietf.org/html/rfc7587

来源：https://stackoverflow.com/questions/21402990/is-there-a-way-to-choose-codecs-in-webrtc-peerconnection

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