HTML5录音实践总结(Preact)_html5_脚本之家

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const mediaStream = await window.navigator.mediaDevices.getUserMedia({ audio: { // sampleRate: 44100, // 采样率不生效需要手动重采样 channelCount: 1, // 声道 // echoCancellation: true, // noiseSuppression: true, // 降噪实测效果不错 const audioContext = new window.AudioContext() const inputSampleRate = audioContext.sampleRate const mediaNode = audioContext.createMediaStreamSource(mediaStream) if (!audioContext.createScriptProcessor) { audioContext.createScriptProcessor = audioContext.createJavaScriptNode // 创建一个jsNode const jsNode = audioContext.createScriptProcessor(4096, 1, 1) jsNode.connect(audioContext.destination) jsNode.onaudioprocess = (e) => { // e.inputBuffer.getChannelData(0) (left) // 双通道通过e.inputBuffer.getChannelData(1)获取 (right) mediaNode.connect(jsNode)

简要流程如下:

start=>start: 开始 getUserMedia=>operation: 获取MediaStream audioContext=>operation: 创建AudioContext scriptNode=>operation: 创建scriptNode并关联AudioContext onaudioprocess=>operation: 设置onaudioprocess并处理数据 end=>end: 结束 start->getUserMedia->audioContext->scriptNode->onaudioprocess->end

停止录制只需要把 audioContext 挂在的 node 卸载即可,然后把存储的每一帧数据合并即可产出 PCM 数据

jsNode.disconnect() mediaNode.disconnect() jsNode.onaudioprocess = null

PCM 数据处理

通过 WebRTC 获取的 PCM 数据格式是 Float32 的, 如果是双通道录音的话, 还需要增加合并通道

const leftDataList = []; const rightDataList = []; function onAudioProcess(event) { // 一帧的音频PCM数据 let audioBuffer = event.inputBuffer; leftDataList.push(audioBuffer.getChannelData(0).slice(0)); rightDataList.push(audioBuffer.getChannelData(1).slice(0)); // 交叉合并左右声道的数据 function 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;

Float32 转 Int16

const float32 = new Float32Array(1) const int16 = Int16Array.from( float32.map(x => (x > 0 ? x * 0x7fff : x * 0x8000)),

arrayBuffer 转 Base64

注意: 在浏览器上有个 btoa() 函数也是可以转换为 Base64 但是输入参数必须为字符串, 如果传递 buffer 参数会先被 toString() 然后再 Base64 , 使用 ffplay 播放反序列化的 Base64 , 会比较刺耳

使用 base64-arraybuffer 即可完成

import { encode } from 'base64-arraybuffer' const float32 = new Float32Array(1) const int16 = Int16Array.from( float32.map(x => (x > 0 ? x * 0x7fff : x * 0x8000)), console.log(encode(int16.buffer))

验证 Base64 是否正确, 可以在 node 下把产出的 Base64 转换为 Int16 的 PCM 文件, 然后使用 FFPlay 播放, 看看音频是否正常播放

PCM 文件播放

# 单通道采样率:16000 Int16 ffplay -f s16le -ar 16k -ac 1 test.pcm # 双通道采样率:48000 Float32 ffplay -f f32le -ar 48000 -ac 2 test.pcm

重采样/调整采样率

虽然 getUserMedia 参数可设置采样率, 但是在最新Chrome也不生效, 所以需要手动做个重采样

const mediaStream = await window.navigator.mediaDevices.getUserMedia({ audio: { // sampleRate: 44100, // 采样率设置不生效 channelCount: 1, // 声道 // echoCancellation: true, // 减低回音 // noiseSuppression: true, // 降噪, 实测效果不错

使用 wave-resampler 即可完成

import { resample } from 'wave-resampler' const inputSampleRate = 44100 const outputSampleRate = 16000 const resampledBuffers = resample( // 需要onAudioProcess每一帧的buffer合并后的数组 mergeArray(audioBuffers), inputSampleRate, outputSampleRate,

PCM 转 MP3

import { Mp3Encoder } from 'lamejs' let mp3buf const mp3Data = [] const sampleBlockSize = 576 * 10 // 工作缓存区, 576的倍数 const mp3Encoder = new Mp3Encoder(1, outputSampleRate, kbps) const samples = float32ToInt16( audioBuffers, inputSampleRate, outputSampleRate, let remaining = samples.length for (let i = 0; remaining >= 0; i += sampleBlockSize) { const left = samples.subarray(i, i + sampleBlockSize) mp3buf = mp3Encoder.encodeBuffer(left) mp3Data.push(new Int8Array(mp3buf)) remaining -= sampleBlockSize mp3Data.push(new Int8Array(mp3Encoder.flush())) console.log(mp3Data) // 工具函数 function float32ToInt16(audioBuffers, inputSampleRate, outputSampleRate) { const float32 = resample( // 需要onAudioProcess每一帧的buffer合并后的数组 mergeArray(audioBuffers), inputSampleRate, outputSampleRate, const int16 = Int16Array.from( float32.map(x => (x > 0 ? x * 0x7fff : x * 0x8000)), return int16

使用 lamejs 即可, 但是体积较大(160+KB), 如果没有存储需求可使用 WAV 格式

> ls -alh -rwxrwxrwx 1 root root 95K 4月 22 12:45 12s.mp3* -rwxrwxrwx 1 root root 1.1M 4月 22 12:44 12s.wav* -rwxrwxrwx 1 root root 235K 4月 22 12:41 30s.mp3* -rwxrwxrwx 1 root root 2.6M 4月 22 12:40 30s.wav* -rwxrwxrwx 1 root root 63K 4月 22 12:49 8s.mp3* -rwxrwxrwx 1 root root 689K 4月 22 12:48 8s.wav*

PCM 转 WAV

function mergeArray(list) { const length = list.length * list[0].length const 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 function writeUTFBytes(view, offset, string) { var lng = string.length for (let i = 0; i < lng; i++) { view.setUint8(offset + i, string.charCodeAt(i)) function createWavBuffer(audioData, sampleRate = 44100, channels = 1) { const WAV_HEAD_SIZE = 44 const buffer = new ArrayBuffer(audioData.length * 2 + WAV_HEAD_SIZE) // 需要用一个view来操控buffer const view = new DataView(buffer) // 写入wav头部信息 // RIFF chunk descriptor/identifier writeUTFBytes(view, 0, 'RIFF') // RIFF chunk length view.setUint32(4, 44 + audioData.length * 2, true) // RIFF type writeUTFBytes(view, 8, 'WAVE') // format chunk identifier // FMT sub-chunk 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, channels, 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, channels * 2, true) // bits per sample view.setUint16(34, 16, true) // data sub-chunk // data chunk identifier writeUTFBytes(view, 36, 'data') // data chunk length view.setUint32(40, audioData.length * 2, true) // 写入PCM数据 let index = 44 const volume = 1 const { length } = audioData for (let i = 0; i < length; i++) { view.setInt16(index, audioData[i] * (0x7fff * volume), true) index += 2 return buffer // 需要onAudioProcess每一帧的buffer合并后的数组 createWavBuffer(mergeArray(audioBuffers))

WAV 基本上是 PCM 加上一些音频信息

简单的短时能量计算

function shortTimeEnergy(audioData) { let sum = 0 const energy = [] const { length } = audioData for (let i = 0; i < length; i++) { sum += audioData[i] ** 2 if ((i + 1) % 256 === 0) { energy.push(sum) sum = 0 } else if (i === length - 1) { energy.push(sum) return energy

由于计算结果有会因设备的录音增益差异较大, 计算出数据也较大, 所以使用比值简单区分人声和噪音

查看 DEMO

const NoiseVoiceWatershedWave = 2.3 const energy = shortTimeEnergy(e.inputBuffer.getChannelData(0).slice(0)) const avg = energy.reduce((a, b) => a + b) / energy.length const nextState = Math.max(...energy) / avg > NoiseVoiceWatershedWave ? 'voice' : 'noise'

Web Worker 优化性能

音频数据数据量较大, 所以可以使用 Web Worker 进行优化, 不卡 UI 线程

在 Webpack 项目里 Web Worker 比较简单, 安装 worker-loader 即可

preact.config.js

export default (config, env, helpers) => { config.module.rules.push({ test: /\.worker\.js$/, use: { loader: 'worker-loader', options: { inline: true } },

recorder.worker.js

self.addEventListener('message', event => { console.log(event.data) // 转MP3/转Base64/转WAV等等 const output = '' self.postMessage(output)

使用 Worker

async function toMP3(audioBuffers, inputSampleRate, outputSampleRate = 16000) { const { default: Worker } = await import('./recorder.worker') const worker = new Worker() // 简单使用, 项目可以在recorder实例化的时候创建worker实例, 有并法需求可多个实例 return new Promise(resolve => { worker.postMessage({ audioBuffers: audioBuffers, inputSampleRate: inputSampleRate, outputSampleRate: outputSampleRate, type: 'mp3', worker.onmessage = event => resolve(event.data)

音频的存储

浏览器持久化储存的地方有 LocalStorage 和 IndexedDB , 其中 LocalStorage 较为常用, 但是只能储存字符串, 而 IndexedDB 可直接储存 Blob , 所以优先选择 IndexedDB ,使用 LocalStorage 则需要转 Base64 体积将会更大

所以为了避免占用用户太多空间, 所以选择MP3格式进行存储

IndexedDB 简单封装如下, 熟悉后台的同学可以找个 ORM 库方便数据读写

const indexedDB = window.indexedDB || window.webkitIndexedDB || window.mozIndexedDB || window.OIndexedDB || window.msIndexedDB const IDBTransaction = window.IDBTransaction || window.webkitIDBTransaction || window.OIDBTransaction || window.msIDBTransaction const readWriteMode = typeof IDBTransaction.READ_WRITE === 'undefined' ? 'readwrite' : IDBTransaction.READ_WRITE const dbVersion = 1 const storeDefault = 'mp3' let dbLink function initDB(store) { return new Promise((resolve, reject) => { if (dbLink) resolve(dbLink) // Create/open database const request = indexedDB.open('audio', dbVersion) request.onsuccess = event => { const db = request.result db.onerror = event => { reject(event) if (db.version === dbVersion) resolve(db) request.onerror = event => { reject(event) // For future use. Currently only in latest Firefox versions request.onupgradeneeded = event => { dbLink = event.target.result const { transaction } = event.target if (!dbLink.objectStoreNames.contains(store)) { dbLink.createObjectStore(store) transaction.oncomplete = event => { // Now store is available to be populated resolve(dbLink) export const writeIDB = async (name, blob, store = storeDefault) => { const db = await initDB(store) const transaction = db.transaction([store], readWriteMode) const objStore = transaction.objectStore(store) return new Promise((resolve, reject) => { const request = objStore.put(blob, name) request.onsuccess = event => resolve(event) request.onerror = event => reject(event) transaction.commit && transaction.commit() export const readIDB = async (name, store = storeDefault) => { const db = await initDB(store) const transaction = db.transaction([store], readWriteMode) const objStore = transaction.objectStore(store) return new Promise((resolve, reject) => { const request = objStore.get(name) request.onsuccess = event => resolve(event.target.result) request.onerror = event => reject(event) transaction.commit && transaction.commit() export const clearIDB = async (store = storeDefault) => { const db = await initDB(store) const transaction = db.transaction([store], readWriteMode) const objStore = transaction.objectStore(store) return new Promise((resolve, reject) => { const request = objStore.clear() request.onsuccess = event => resolve(event) request.onerror = event => reject(event) transaction.commit && transaction.commit()

WebView 开启 WebRTC

见 WebView WebRTC not working

webView.setWebChromeClient(new WebChromeClient(){ @TargetApi(Build.VERSION_CODES.LOLLIPOP) @Override public void onPermissionRequest(final PermissionRequest request) { request.grant(request.getResources());

到此这篇关于HTML5录音实践总结(Preact)的文章就介绍到这了,更多相关html5录音内容请搜索脚本之家以前的文章或继续浏览下面的相关文章，希望大家以后多多支持脚本之家！

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