【发布时间】:2014-10-21 08:03:48
【问题描述】:
我想将我的音频文件作为麦克风输入(不发送我的现场声音,但我的音频文件)播放给连接到 WebRTC 的用户。谁能告诉我怎么做?
我在 JS 代码中做了以下尝试,比如:
1. base64 音频
<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. 音频缓冲区
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";
}
请在这种情况下帮助我。这将是非常可观的。
【问题讨论】:
标签: input audio-streaming webrtc microphone