2019独角兽企业重金招聘Python工程师标准>>> 
之前玩过WebRTC，通过HTML5可以轻松实现语音视频聊天，网上的示例很多。最近想了解下HTML5能否录制音频，用什么接口可以做到。这里分享下学习资料。
浏览器
要获取音频和视频，需要用到getUserMedia。桌面平台支持的浏览器包括Chrome, Firefox, Opera和Edge。移动平台包括Chrome, Firefox和Opera，不过只是Android平台，iOS不行。微软似乎也放弃IE了，全力支持Edge。苹果对HTML5的支持始终不够。前段日子还有新闻报道说有开发者针对HTML5起诉苹果。
获取视频音频流
网上示例很多，从Mozilla的文档中找了一段最简单的代码：
var p = navigator.mediaDevices.getUserMedia({ audio: true, video: true });
p.then(function(mediaStream) {
var video = document.querySelector('video');
video.src = window.URL.createObjectURL(mediaStream);
video.onloadedmetadata = function(e) {
// Do something with the video here.
video.play();
};
});
p.catch(function(err) { console.log(err.name); }); // always check for errors at the end.录制Audio
在Mozilla的HTML5文档中看到了MediaRecorder。这个接口简单方便，但是比较新，浏览器的兼容性有限。
桌面
移动
不过也有替代方案，使用AudioNodes。基本步骤如下：
1. 通过getUserMedia获取视频音频流。
2. 通过createMediaStreamSource创建MediaStreamAudioSourceNode。
if (navigator.getUserMedia) {
console.log('getUserMedia supported.');
navigator.getUserMedia (
// constraints: audio and video for this app
{
audio: true,
video: true
},
// Success callback
function(stream) {
video.src = (window.URL && window.URL.createObjectURL(stream)) || stream;
video.onloadedmetadata = function(e) {
video.play();
video.muted = 'true';
};
// Create a MediaStreamAudioSourceNode
// Feed the HTMLMediaElement into it
var source = audioCtx.createMediaStreamSource(stream);
},
// Error callback
function(err) {
console.log('The following gUM error occured: ' + err);
}
);
} else {
console.log('getUserMedia not supported on your browser!');
}3. 链接AudioNodes。创建ScriptProcessorNode。通过onaudioprocess来获取音频数据。
var scriptNode = audioCtx.createScriptProcessor(4096, 1, 1);
scriptNode.onaudioprocess = function(audioProcessingEvent) {
// The input buffer is the song we loaded earlier
var inputBuffer = audioProcessingEvent.inputBuffer;
// Loop through the output channels (in this case there is only one)
for (var channel = 0; channel < outputBuffer.numberOfChannels; channel++) {
var inputData = inputBuffer.getChannelData(channel);
}
}
source.connect(scriptNode);
scriptNode.connect(audioCtx.destination);4. 通过XHR或者WebSockets来发送blob数据。
JavaScript库
流程就这么简单，但是写起来还是比较复杂的。所以需要参考下别人写的代码。在GitHub上可以找到RecordRTC和Recorderjs。前者可以录制视频和音频，如果只想录制音频，可以选择后面这个，比较简单。
关于RecordRTC，有一个站点可以体验视频音频录制：online demo。
现在分析下Recorderjs的源码。
在onaudioprocess中获取音频buffer：
this.context = source.context;
this.node = (this.context.createScriptProcessor || this.context.createJavaScriptNode).call(this.context, this.config.bufferLen, this.config.numChannels, this.config.numChannels);
this.node.onaudioprocess = function (e) {
if (!_this.recording) return;
var buffer = [];
for (var channel = 0; channel < _this.config.numChannels; channel++) {
buffer.push(e.inputBuffer.getChannelData(channel));
}
_this.worker.postMessage({
command: 'record',
buffer: buffer
});
};
source.connect(this.node);
this.node.connect(this.context.destination); //this should not be necessary用数组存储音频buffer：
function record(inputBuffer) {
for (var channel = 0; channel < numChannels; channel++) {
recBuffers[channel].push(inputBuffer[channel]);
}
recLength += inputBuffer[0].length;
}WAV格式编码：
function encodeWAV(samples) {
var buffer = new ArrayBuffer(44 + samples.length * 2);
var view = new DataView(buffer);
/* RIFF identifier */
writeString(view, 0, 'RIFF');
/* RIFF chunk length */
view.setUint32(4, 36 + samples.length * 2, true);
/* RIFF type */
writeString(view, 8, 'WAVE');
/* format chunk identifier */
writeString(view, 12, 'fmt ');
/* format chunk length */
view.setUint32(16, 16, true);
/* sample format (raw) */
view.setUint16(20, 1, true);
/* channel count */
view.setUint16(22, numChannels, true);
/* sample rate */
view.setUint32(24, sampleRate, true);
/* byte rate (sample rate * block align) */
view.setUint32(28, sampleRate * 4, true);
/* block align (channel count * bytes per sample) */
view.setUint16(32, numChannels * 2, true);
/* bits per sample */
view.setUint16(34, 16, true);
/* data chunk identifier */
writeString(view, 36, 'data');
/* data chunk length */
view.setUint32(40, samples.length * 2, true);
floatTo16BitPCM(view, 44, samples);
return view;
}合并所有的buffer，并用WAV格式导出，最后转换成blob：
function exportWAV(type) {
var buffers = [];
for (var channel = 0; channel < numChannels; channel++) {
buffers.push(mergeBuffers(recBuffers[channel], recLength));
}
var interleaved = undefined;
if (numChannels === 2) {
interleaved = interleave(buffers[0], buffers[1]);
} else {
interleaved = buffers[0];
}
var dataview = encodeWAV(interleaved);
var audioBlob = new Blob([dataview], { type: type });
}这样就可以保存或者发送录制的音频文件了。
参考资料
- WebRTC roadmap
- Using the MediaRecorder API
- getUserMedia
- Web Audio API