grab audio webRTC audio stream and play with the buffer in javascript

gistfile1.js

var el = document.getElementById('sound');
var vid = document.getElementById('vid'); 
var audio = new webkitAudioContext();
var js = audio.createJavaScriptNode(1024);
js.connect(audio.destination);


js.onaudioprocess = function(obj){
    var buf = obj.outputBuffer.getChannelData(1) // obj.inputBuffer.getChannelData(1)
    var buf2 = obj.inputBuffer.getChannelData(1);
	var x;
	for(x = 0; x < buf.length; x++){
		buf[x] = buf2[x]
	}
//	console.log(buf2);
}

js.connect(audio.destination)

window.URL = window.URL || window.webkitURL;
navigator.getUserMedia  = navigator.getUserMedia || navigator.webkitGetUserMedia ||
                          navigator.mozGetUserMedia || navigator.msGetUserMedia;

function convertToMono( input ) {
    var splitter = audio.createChannelSplitter(2);
    var merger = audio.createChannelMerger(2);

    input.connect( splitter );
    splitter.connect( merger, 0, 0 );
    splitter.connect( merger, 0, 1 );
    return merger;
}

function gotStream(stream) {

    // Create an AudioNode from the stream.

    var input = audio.createMediaStreamSource(stream);
    var url = webkitURL.createObjectURL(stream);
    vid.src = url;
    audioInput = convertToMono( input );

    audioInput.connect(js);
}

if (!navigator.webkitGetUserMedia)
        return(alert("Error: getUserMedia not supported!"));

    navigator.webkitGetUserMedia({audio:true, video: true}, gotStream, function(e) {
            alert('Error getting audio');
            console.log(e);
        });