我想将我的音频文件作为麦克风输入播放(不发送我的实时语音,而是我的音频文件)给 WebRTC 连接的用户。谁能告诉我怎么做?
我在 JS 代码中做了一些以下尝试,例如:
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";
}
在这种情况下,请帮助我。这将是非常可观的。
最佳答案
这是一个演示,可以帮助您使用 chrome 流式传输 mp3 或 wav:
这是它的写法:
和演示的源代码:
在第 3 方 WebRTC 应用程序中使用
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
}
更新时间:下午 5:55 - 2014 年 8 月 28 日,星期四
以下是从服务器获取 mp3 的方法:
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);
});
});
关于input - WebRTC 作为麦克风播放音频输入,我们在Stack Overflow上找到一个类似的问题: https://stackoverflow.com/questions/25532200/