https://juejin.cn/post/7243240618788012090
音波:https://xiangyuecn.github.io/Recorder/assets/%E5%B7%A5%E5%85%B7-%E4%BB%A3%E7%A0%81%E8%BF%90%E8%A1%8C%E5%92%8C%E9%9D%99%E6%80%81%E5%88%86%E5%8F%91Runtime.html?jsname=teach.source_stream.capture_stream
Web RTC + audio API 实现录音,并压缩
<button onclick="record()">开始录音</button>
<button onclick="stopRecord()">结束录音</button>
<!-- <button onclick="resetRecord()">重置录音</button> -->
<audio class="audio-node" id="audio" autoplay></audio>
<script>
class Recoder {
constructor (sampleRate) {
this.leftDataList = []
this.rightDataList = []
this.mediaPlayer = null
this.audioContext = null
this.source = null
this.sampleRate = sampleRate || 44100
}
startRecord () {
return new Promise((resolve, reject) => {
window.navigator.mediaDevices.getUserMedia({
audio: {
sampleRate: 8000, // 采样率
channelCount: 1, // 声道
audioBitsPerSecond : 64,
volume: 1.0, // 音量
autoGainControl: true
}
}).then(mediaStream => {
console.log(mediaStream,'mediaStream')
this.mediaPlayer = mediaStream
this.beginRecord(mediaStream)
resolve()
}).catch(err => {
// 如果用户电脑没有麦克风设备或者用户拒绝了,或者连接出问题了等
// 这里都会抛异常,并且通过err.name可以知道是哪种类型的错误
console.error(err)
reject(err)
})
})
}
beginRecord (mediaStream) {
let audioContext = new (window.AudioContext || window.webkitAudioContext)()
// mediaNode包含 mediaStream,audioContext
let mediaNode = audioContext.createMediaStreamSource(mediaStream)
console.log(mediaNode,'mediaNode')
// 创建一个jsNode
// audioContext.sampleRate = 8000
console.log(audioContext,'audioContext')
let jsNode = this.createJSNode(audioContext)
console.log(jsNode,'jsnode')
// 需要连到扬声器消费掉outputBuffer,process回调才能触发
// 并且由于不给outputBuffer设置内容,所以扬声器不会播放出声音
jsNode.connect(audioContext.destination)
jsNode.onaudioprocess = this.onAudioProcess.bind(this)
// 把mediaNode连接到jsNode
mediaNode.connect(jsNode)
this.audioContext = audioContext
}
onAudioProcess (event) {
console.log('is recording')
// 拿到输入buffer Float32Array
let audioBuffer = event.inputBuffer
let leftChannelData = audioBuffer.getChannelData(0)
// let rightChannelData = audioBuffer.getChannelData(1)
// 需要克隆一下
this.leftDataList.push(leftChannelData.slice(0))
//this.rightDataList.push(rightChannelData.slice(0))
}
createJSNode (audioContext) {
const BUFFER_SIZE = 4096
const INPUT_CHANNEL_COUNT = 1
const OUTPUT_CHANNEL_COUNT = 1
// createJavaScriptNode已被废弃
let creator = audioContext.createScriptProcessor || audioContext.createJavaScriptNode
creator = creator.bind(audioContext)
return creator(BUFFER_SIZE, INPUT_CHANNEL_COUNT, OUTPUT_CHANNEL_COUNT)
}
playRecord (arrayBuffer) {
let blob = new Blob([new Int8Array(arrayBuffer)], {
type: 'audio/mp3' // files[0].type
})
let blobUrl = URL.createObjectURL(blob)
this.source = blob
this.blobUrl = blobUrl
// document.querySelector('.audio-node').src = blobUrl
return blobUrl
}
stopRecord () {
// 停止录音
let leftData = this.mergeArray(this.leftDataList)
//let rightData = this.mergeArray(this.rightDataList)
let allData = this.interSingleData(leftData)
let wavBuffer = this.createWavFile(allData)
let source = this.playRecord(wavBuffer)
this.resetRecord()
return source
}
transformArrayBufferToBase64 (buffer) {
var binary = ''
var bytes = new Uint8Array(buffer)
for (var len = bytes.byteLength, i = 0; i < len; i++) {
binary += String.fromCharCode(bytes[i])
}
return window.btoa(binary)
}
// 停止控件录音
resetRecord () {
this.leftDataList = []
this.rightDataList = []
this.audioContext.close()
this.mediaPlayer.getAudioTracks().forEach(track => {
track.stop()
this.mediaPlayer.removeTrack(track)
})
}
createWavFile (audioData) {
let channelCount = 1
const WAV_HEAD_SIZE = 44
const sampleBits = 16
let sampleRate = this.sampleRate
let buffer = new ArrayBuffer(audioData.length * 2 + WAV_HEAD_SIZE)
// 需要用一个view来操控buffer
let view = new DataView(buffer)
// 写入wav头部信息
// RIFF chunk descriptor/identifier
this.writeUTFBytes(view, 0, 'RIFF')
// RIFF chunk length
view.setUint32(4, 44 + audioData.length * channelCount, true)
// RIFF type
this.writeUTFBytes(view, 8, 'WAVE')
// format chunk identifier
// FMT sub-chunk
this.writeUTFBytes(view, 12, 'fmt ')
// format chunk length
view.setUint32(16, 16, true)
// sample format (raw)
view.setUint16(20, 1, true)
// stereo (2 channels)
view.setUint16(22, channelCount, true)
// sample rate
view.setUint32(24, sampleRate , true)
// byte rate (sample rate * block align)
view.setUint32(28, sampleRate * 2, true)
// block align (channel count * bytes per sample)
view.setUint16(32, 2 * 2, true)
// bits per sample
view.setUint16(34, 16, true)
// data sub-chunk
// data chunk identifier
this.writeUTFBytes(view, 36, 'data')
// data chunk length
view.setUint32(40, audioData.length * 2, true)
console.log(view,'view')
let length = audioData.length
let index = 44
let volume = 1
for (let i = 0; i < length; i++) {
view.setInt16(index, audioData[i] * (0x7FFF * volume), true)
index += 2
}
return buffer
}
writeUTFBytes (view, offset, string) {
var lng = string.length
for (var i = 0; i < lng; i++) {
view.setUint8(offset + i, string.charCodeAt(i))
}
}
interSingleData (left) {
var t = left.length;
let sampleRate = this.audioContext.sampleRate,
outputSampleRate = this.sampleRate
sampleRate += 0.0;
outputSampleRate += 0.0;
var s = 0,
o = sampleRate / outputSampleRate,
u = Math.ceil(t * outputSampleRate / sampleRate),
a = new Float32Array(u);
for (let i = 0; i < u; i++) {
a[i] = left[Math.floor(s)];
s += o;
}
return a;
}
// 交叉合并左右声道的数据
interleaveLeftAndRight (left, right) {
let totalLength = left.length + right.length
let data = new Float32Array(totalLength)
for (let i = 0; i < left.length; i++) {
let k = i * 2
data[k] = left[i]
data[k + 1] = right[i]
}
return data
}
mergeArray (list) {
let length = list.length * list[0].length
let data = new Float32Array(length)
let offset = 0
for (let i = 0; i < list.length; i++) {
data.set(list[i], offset)
offset += list[i].length
}
return data
}
// 播放音乐
playMusic () {
if (!this.value) {
return
}
// 直接使用File对象生成blob url
let blobUrl = URL.createObjectURL(this.files[0])
document.querySelector('.audio-node').src = blobUrl
}
play (arrayBuffer) {
// Safari需要使用webkit前缀
let AudioContext = this.AudioContext || this.webkitAudioContext
let audioContext = new AudioContext()
// 创建一个AudioBufferSourceNode对象,使用AudioContext的工厂函数创建
let audioNode = audioContext.createBufferSource()
// 解码音频,可以使用Promise,但是较老的Safari需要使用回调
audioContext.decodeAudioData(arrayBuffer, function (audioBuffer) {
audioNode.buffer = audioBuffer
audioNode.connect(audioContext.destination)
// 从0s开始播放
audioNode.start(0)
})
}
}
let recoder = new Recoder(8000)
function record() {
recoder.startRecord()
}
function stopRecord(params) {
recoder.stopRecord()
let source = recoder.source
let formData = new FormData()
formData.append('audio', source)
let audio = document.getElementById('audio')
audio.src = recoder.blobUrl
}
</script>
参考 自掘金
1.网上很多都没有停之功能,新增停之
2.这里用的8000采样率和单声道,音频体积为原来的 快 1/12 (我电脑设备的采样录是44K)
3.详细解释请看 H5音频分析