init
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258
hook/useRealtimeRecorder.js
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258
hook/useRealtimeRecorder.js
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import {
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ref,
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onUnmounted
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} from 'vue'
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import {
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$api,
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} from '../common/globalFunction';
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import config from '@/config'
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export function useAudioRecorder() {
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const isRecording = ref(false)
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const isStopping = ref(false)
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const isSocketConnected = ref(false)
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const recordingDuration = ref(0)
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const audioDataForDisplay = ref(new Array(16).fill(0))
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const volumeLevel = ref(0)
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const recognizedText = ref('')
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const lastFinalText = ref('')
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let audioStream = null
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let audioContext = null
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let audioInput = null
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let scriptProcessor = null
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let websocket = null
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let durationTimer = null
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const generateUUID = () => {
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return ([1e7] + -1e3 + -4e3 + -8e3 + -1e11)
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.replace(/[018]/g, c =>
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(c ^ crypto.getRandomValues(new Uint8Array(1))[0] & 15 >> c / 4).toString(16)
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).replace(/-/g, '')
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}
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const fetchWsUrl = async () => {
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const res = await $api.createRequest('/app/speech/getToken')
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if (res.code !== 200) throw new Error('无法获取语音识别 wsUrl')
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const wsUrl = res.msg
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return wsUrl
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}
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function extractWsParams(wsUrl) {
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const url = new URL(wsUrl)
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const appkey = url.searchParams.get('appkey')
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const token = url.searchParams.get('token')
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return {
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appkey,
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token
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}
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}
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const connectWebSocket = async () => {
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const wsUrl = await fetchWsUrl()
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const {
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appkey,
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token
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} = extractWsParams(wsUrl)
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return new Promise((resolve, reject) => {
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websocket = new WebSocket(wsUrl)
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websocket.binaryType = 'arraybuffer'
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websocket.onopen = () => {
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isSocketConnected.value = true
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// 发送 StartTranscription 消息（参考 demo.html）
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const startTranscriptionMessage = {
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header: {
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appkey: appkey, // 不影响使用，可留空或由 wsUrl 带入
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namespace: 'SpeechTranscriber',
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name: 'StartTranscription',
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task_id: generateUUID(),
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message_id: generateUUID()
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},
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payload: {
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format: 'pcm',
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sample_rate: 16000,
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enable_intermediate_result: true,
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enable_punctuation_prediction: true,
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enable_inverse_text_normalization: true
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}
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}
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websocket.send(JSON.stringify(startTranscriptionMessage))
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resolve()
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}
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websocket.onerror = (e) => {
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isSocketConnected.value = false
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reject(e)
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}
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websocket.onclose = () => {
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isSocketConnected.value = false
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}
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websocket.onmessage = (e) => {
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const msg = JSON.parse(e.data)
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const name = msg?.header?.name
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const payload = msg?.payload
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switch (name) {
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case 'TranscriptionResultChanged': {
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// 中间识别文本（可选：使用 stash_result.unfixedText 更精确）
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const text = payload?.unfixed_result || payload?.result || ''
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lastFinalText.value = text
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break
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}
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case 'SentenceBegin': {
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// 可选：开始新的一句，重置状态
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// console.log('开始新的句子识别')
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break
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}
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case 'SentenceEnd': {
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const text = payload?.result || ''
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const confidence = payload?.confidence || 0
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if (text && confidence > 0.5) {
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recognizedText.value += text
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lastFinalText.value = ''
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// console.log('识别完成:', {
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// text,
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// confidence
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// })
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}
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break
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}
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case 'TranscriptionStarted': {
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// console.log('识别任务已开始')
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break
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}
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case 'TranscriptionCompleted': {
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lastFinalText.value = ''
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// console.log('识别全部完成')
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break
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}
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case 'TaskFailed': {
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console.error('识别失败:', msg?.header?.status_text)
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break
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}
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default:
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console.log('未知消息类型:', name, msg)
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break
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}
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}
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})
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}
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const startRecording = async () => {
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if (isRecording.value) return
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try {
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recognizedText.value = ''
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lastFinalText.value = ''
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await connectWebSocket()
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audioStream = await navigator.mediaDevices.getUserMedia({
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audio: true
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})
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audioContext = new(window.AudioContext || window.webkitAudioContext)({
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sampleRate: 16000
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})
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audioInput = audioContext.createMediaStreamSource(audioStream)
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scriptProcessor = audioContext.createScriptProcessor(2048, 1, 1)
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scriptProcessor.onaudioprocess = (event) => {
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const input = event.inputBuffer.getChannelData(0)
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const pcm = new Int16Array(input.length)
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let sum = 0
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for (let i = 0; i < input.length; ++i) {
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const s = Math.max(-1, Math.min(1, input[i]))
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pcm[i] = s * 0x7FFF
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sum += s * s
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}
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volumeLevel.value = Math.sqrt(sum / input.length)
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audioDataForDisplay.value = Array(16).fill(volumeLevel.value)
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if (websocket?.readyState === WebSocket.OPEN) {
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websocket.send(pcm.buffer)
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}
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}
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audioInput.connect(scriptProcessor)
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scriptProcessor.connect(audioContext.destination)
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isRecording.value = true
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recordingDuration.value = 0
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durationTimer = setInterval(() => recordingDuration.value++, 1000)
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} catch (err) {
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console.error('启动失败:', err)
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cleanup()
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}
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}
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const stopRecording = () => {
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if (!isRecording.value || isStopping.value) return
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isStopping.value = true
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if (websocket?.readyState === WebSocket.OPEN) {
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websocket.send(JSON.stringify({
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header: {
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namespace: 'SpeechTranscriber',
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name: 'StopTranscription',
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message_id: generateUUID()
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}
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}))
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websocket.close()
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}
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cleanup()
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isStopping.value = false
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}
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const cancelRecording = () => {
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if (!isRecording.value || isStopping.value) return
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isStopping.value = true
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websocket?.close()
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cleanup()
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isStopping.value = false
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}
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const cleanup = () => {
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clearInterval(durationTimer)
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scriptProcessor?.disconnect()
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audioInput?.disconnect()
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audioStream?.getTracks().forEach(track => track.stop())
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audioContext?.close()
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audioStream = null
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audioContext = null
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audioInput = null
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scriptProcessor = null
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websocket = null
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isRecording.value = false
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isSocketConnected.value = false
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}
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onUnmounted(() => {
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if (isRecording.value) stopRecording()
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})
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return {
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isRecording,
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isStopping,
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isSocketConnected,
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recordingDuration,
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audioDataForDisplay,
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volumeLevel,
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recognizedText,
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lastFinalText,
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startRecording,
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stopRecording,
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cancelRecording
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}
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}
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