WebRTC Play Audio Input as Microphone - input

Question

I want to play my audio file as microphone input (without sending my live voice but my audio file) to the WebRTC connected user. Can anybody tell me how could it be done?
I have done some following tries in the JS code, like:
1. base64 Audio
<script>
var base64string = "T2dnUwACAAAAAAA..";
var snd = new Audio("data:audio/wav;base64," + base64string);
snd.play();
var Sound = (function () {
var df = document.createDocumentFragment();
return function Sound(src) {
var snd = new Audio(src);
df.appendChild(snd);
snd.addEventListener('ended', function () {df.removeChild(snd);});
snd.play();
return snd;
}
}());
var snd = Sound("data:audio/wav;base64," + base64string);
</script>
2. AudioBuffer
window.AudioContext = window.AudioContext || window.webkitAudioContext;
var audioContext = new AudioContext();
var isPlaying = false;
var sourceNode = null;
var theBuffer = null;
window.onload = function() {
var request = new XMLHttpRequest();
request.open("GET", "sounds/DEMO_positive_resp.wav", true);
request.responseType = "arraybuffer";
request.onload = function() {
audioContext.decodeAudioData( request.response, function(buffer) {
theBuffer = buffer;
} );
}
request.send();
}
function togglePlayback() {
var now = audioContext.currentTime;
if (isPlaying) {
//stop playing and return
sourceNode.stop( now );
sourceNode = null;
analyser = null;
isPlaying = false;
if (!window.cancelAnimationFrame)
window.cancelAnimationFrame = window.webkitCancelAnimationFrame;
//window.cancelAnimationFrame( rafID );
return "start";
}
sourceNode = audioContext.createBufferSource();
sourceNode.buffer = theBuffer;
sourceNode.loop = true;
analyser = audioContext.createAnalyser();
analyser.fftSize = 2048;
sourceNode.connect( analyser );
analyser.connect( audioContext.destination );
sourceNode.start( now );
isPlaying = true;
isLiveInput = true;
return "stop";
}
Please help me out in this case. It would be highly appreciable.

Answer 1

Here is a demo that may help you stream mp3 or wav using chrome:
https://www.webrtc-experiment.com/RTCMultiConnection/stream-mp3-live.html
Here is, how it is written:
http://www.rtcmulticonnection.org/docs/getting-started/#stream-mp3-live
And source code of the demo:
https://github.com/muaz-khan/RTCMultiConnection/blob/master/demos/stream-mp3-live.html
https://github.com/muaz-khan/WebRTC-Experiment/issues/222
Use in 3rd party WebRTC applications
window.AudioContext = window.AudioContext || window.webkitAudioContext;
var context = new AudioContext();
var gainNode = context.createGain();
gainNode.connect(context.destination);
// don't play for self
gainNode.gain.value = 0;
document.querySelector('input[type=file]').onchange = function() {
this.disabled = true;
var reader = new FileReader();
reader.onload = (function(e) {
// Import callback function that provides PCM audio data decoded as an audio buffer
context.decodeAudioData(e.target.result, function(buffer) {
// Create the sound source
var soundSource = context.createBufferSource();
soundSource.buffer = buffer;
soundSource.start(0, 0 / 1000);
soundSource.connect(gainNode);
var destination = context.createMediaStreamDestination();
soundSource.connect(destination);
createPeerConnection(destination.stream);
});
});
reader.readAsArrayBuffer(this.files[0]);
};
function createPeerConnection(mp3Stream) {
// you need to place 3rd party WebRTC code here
}
Updated at: 5:55 PM - Thursday, August 28, 2014
Here is how to get mp3 from server:
function HTTP_GET(url, callback) {
var xhr = new XMLHttpRequest();
xhr.open('GET', url, true);
xhr.responseType = 'arraybuffer';
xhr.send();
xhr.onload = function(e) {
if (xhr.status != 200) {
alert("Unexpected status code " + xhr.status + " for " + url);
return false;
}
callback(xhr.response); // return array-buffer
};
}
// invoke above "HTTP_GET" method
// to load mp3 as array-buffer
HTTP_GET('http://domain.com/file.mp3', function(array_buffer) {
// Import callback function that provides PCM audio data decoded as an audio buffer
context.decodeAudioData(array_buffer, function(buffer) {
// Create the sound source
var soundSource = context.createBufferSource();
soundSource.buffer = buffer;
soundSource.start(0, 0 / 1000);
soundSource.connect(gainNode);
var destination = context.createMediaStreamDestination();
soundSource.connect(destination);
createPeerConnection(destination.stream);
});
});