flat: 优化语音
This commit is contained in:
BIN
hook/.DS_Store
vendored
BIN
hook/.DS_Store
vendored
Binary file not shown.
@@ -31,8 +31,8 @@ export function useAudioRecorder(wsUrl) {
|
||||
// 配置常量
|
||||
const SAMPLE_RATE = 16000;
|
||||
const SILENCE_THRESHOLD = 0.02; // 静音阈值 (0-1)
|
||||
const SILENCE_DURATION = 400; // 静音持续时间(ms)后切片
|
||||
const MIN_SOUND_DURATION = 300; // 最小有效声音持续时间(ms)
|
||||
const SILENCE_DURATION = 100; // 静音持续时间(ms)后切片
|
||||
const MIN_SOUND_DURATION = 200; // 最小有效声音持续时间(ms)
|
||||
|
||||
// 音频处理变量
|
||||
const lastSoundTime = ref(0);
|
||||
@@ -125,6 +125,7 @@ export function useAudioRecorder(wsUrl) {
|
||||
socket.value = new WebSocket(wsUrl);
|
||||
|
||||
socket.value.onopen = () => {
|
||||
console.log('open')
|
||||
isSocketConnected.value = true;
|
||||
resolve();
|
||||
};
|
||||
@@ -191,7 +192,6 @@ export function useAudioRecorder(wsUrl) {
|
||||
if (audioChunks.value.length === 0 || !socket.value || socket.value.readyState !== WebSocket.OPEN) {
|
||||
return;
|
||||
}
|
||||
|
||||
try {
|
||||
// 合并所有块
|
||||
const totalBytes = audioChunks.value.reduce((total, chunk) => total + chunk.byteLength, 0);
|
||||
|
249
hook/useTTSPlayer.js
Normal file
249
hook/useTTSPlayer.js
Normal file
@@ -0,0 +1,249 @@
|
||||
import {
|
||||
ref,
|
||||
onUnmounted,
|
||||
onBeforeUnmount,
|
||||
onMounted
|
||||
} from 'vue'
|
||||
import {
|
||||
onHide,
|
||||
onUnload
|
||||
} from '@dcloudio/uni-app'
|
||||
import WavDecoder from '@/lib/wav-decoder@1.3.0.js'
|
||||
|
||||
export function useTTSPlayer(wsUrl) {
|
||||
const isSpeaking = ref(false)
|
||||
const isPaused = ref(false)
|
||||
const isComplete = ref(false)
|
||||
|
||||
const audioContext = new(window.AudioContext || window.webkitAudioContext)()
|
||||
let playTime = audioContext.currentTime
|
||||
let sourceNodes = []
|
||||
let socket = null
|
||||
let sampleRate = 16000
|
||||
let numChannels = 1
|
||||
let isHeaderDecoded = false
|
||||
let pendingText = null
|
||||
|
||||
let currentPlayId = 0
|
||||
let activePlayId = 0
|
||||
|
||||
const speak = (text) => {
|
||||
currentPlayId++
|
||||
const myPlayId = currentPlayId
|
||||
reset()
|
||||
pendingText = text
|
||||
activePlayId = myPlayId
|
||||
}
|
||||
|
||||
const pause = () => {
|
||||
if (audioContext.state === 'running') {
|
||||
audioContext.suspend()
|
||||
isPaused.value = true
|
||||
isSpeaking.value = false
|
||||
}
|
||||
}
|
||||
|
||||
const resume = () => {
|
||||
if (audioContext.state === 'suspended') {
|
||||
audioContext.resume()
|
||||
isPaused.value = false
|
||||
isSpeaking.value = true
|
||||
}
|
||||
}
|
||||
|
||||
const cancelAudio = () => {
|
||||
stop()
|
||||
}
|
||||
|
||||
const stop = () => {
|
||||
isSpeaking.value = false
|
||||
isPaused.value = false
|
||||
isComplete.value = false
|
||||
playTime = audioContext.currentTime
|
||||
|
||||
sourceNodes.forEach(node => {
|
||||
try {
|
||||
node.stop()
|
||||
node.disconnect()
|
||||
} catch (e) {}
|
||||
})
|
||||
sourceNodes = []
|
||||
|
||||
if (socket) {
|
||||
socket.close()
|
||||
socket = null
|
||||
}
|
||||
|
||||
isHeaderDecoded = false
|
||||
pendingText = null
|
||||
}
|
||||
|
||||
const reset = () => {
|
||||
stop()
|
||||
isSpeaking.value = false
|
||||
isPaused.value = false
|
||||
isComplete.value = false
|
||||
playTime = audioContext.currentTime
|
||||
initWebSocket()
|
||||
}
|
||||
|
||||
const initWebSocket = () => {
|
||||
const thisPlayId = currentPlayId
|
||||
socket = new WebSocket(wsUrl)
|
||||
socket.binaryType = 'arraybuffer'
|
||||
|
||||
socket.onopen = () => {
|
||||
if (pendingText && thisPlayId === activePlayId) {
|
||||
const seepdText = extractSpeechText(pendingText)
|
||||
socket.send(seepdText)
|
||||
pendingText = null
|
||||
}
|
||||
}
|
||||
|
||||
socket.onmessage = async (e) => {
|
||||
if (thisPlayId !== activePlayId) return // 忽略旧播放的消息
|
||||
|
||||
if (typeof e.data === 'string') {
|
||||
try {
|
||||
const msg = JSON.parse(e.data)
|
||||
if (msg.status === 'complete') {
|
||||
isComplete.value = true
|
||||
setTimeout(() => {
|
||||
if (thisPlayId === activePlayId) {
|
||||
isSpeaking.value = false
|
||||
}
|
||||
}, (playTime - audioContext.currentTime) * 1000)
|
||||
}
|
||||
} catch (e) {
|
||||
console.log('[TTSPlayer] 文本消息:', e.data)
|
||||
}
|
||||
} else if (e.data instanceof ArrayBuffer) {
|
||||
if (!isHeaderDecoded) {
|
||||
try {
|
||||
const decoded = await WavDecoder.decode(e.data)
|
||||
sampleRate = decoded.sampleRate
|
||||
numChannels = decoded.channelData.length
|
||||
decoded.channelData.forEach((channel, i) => {
|
||||
const audioBuffer = audioContext.createBuffer(1, channel.length,
|
||||
sampleRate)
|
||||
audioBuffer.copyToChannel(channel, 0)
|
||||
playBuffer(audioBuffer)
|
||||
})
|
||||
isHeaderDecoded = true
|
||||
} catch (err) {
|
||||
console.error('WAV 解码失败:', err)
|
||||
}
|
||||
} else {
|
||||
const pcm = new Int16Array(e.data)
|
||||
const audioBuffer = pcmToAudioBuffer(pcm, sampleRate, numChannels)
|
||||
playBuffer(audioBuffer)
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
const pcmToAudioBuffer = (pcm, sampleRate, numChannels) => {
|
||||
const length = pcm.length / numChannels
|
||||
const audioBuffer = audioContext.createBuffer(numChannels, length, sampleRate)
|
||||
for (let ch = 0; ch < numChannels; ch++) {
|
||||
const channelData = audioBuffer.getChannelData(ch)
|
||||
for (let i = 0; i < length; i++) {
|
||||
const sample = pcm[i * numChannels + ch]
|
||||
channelData[i] = sample / 32768
|
||||
}
|
||||
}
|
||||
return audioBuffer
|
||||
}
|
||||
|
||||
const playBuffer = (audioBuffer) => {
|
||||
if (!isSpeaking.value) {
|
||||
playTime = audioContext.currentTime
|
||||
}
|
||||
const source = audioContext.createBufferSource()
|
||||
source.buffer = audioBuffer
|
||||
source.connect(audioContext.destination)
|
||||
source.start(playTime)
|
||||
sourceNodes.push(source)
|
||||
playTime += audioBuffer.duration
|
||||
isSpeaking.value = true
|
||||
}
|
||||
|
||||
onUnmounted(() => {
|
||||
stop()
|
||||
})
|
||||
|
||||
// 页面刷新/关闭时
|
||||
onMounted(() => {
|
||||
if (typeof window !== 'undefined') {
|
||||
window.addEventListener('beforeunload', cancelAudio)
|
||||
}
|
||||
})
|
||||
|
||||
onBeforeUnmount(() => {
|
||||
cancelAudio()
|
||||
if (typeof window !== 'undefined') {
|
||||
window.removeEventListener('beforeunload', cancelAudio)
|
||||
}
|
||||
})
|
||||
|
||||
onHide(cancelAudio)
|
||||
onUnload(cancelAudio)
|
||||
|
||||
initWebSocket()
|
||||
|
||||
return {
|
||||
speak,
|
||||
pause,
|
||||
resume,
|
||||
cancelAudio,
|
||||
isSpeaking,
|
||||
isPaused,
|
||||
isComplete
|
||||
}
|
||||
}
|
||||
|
||||
function extractSpeechText(markdown) {
|
||||
const jobRegex = /``` job-json\s*({[\s\S]*?})\s*```/g;
|
||||
const jobs = [];
|
||||
let match;
|
||||
let lastJobEndIndex = 0;
|
||||
let firstJobStartIndex = -1;
|
||||
|
||||
// 提取岗位 json 数据及前后位置
|
||||
while ((match = jobRegex.exec(markdown)) !== null) {
|
||||
const jobStr = match[1];
|
||||
try {
|
||||
const job = JSON.parse(jobStr);
|
||||
jobs.push(job);
|
||||
if (firstJobStartIndex === -1) {
|
||||
firstJobStartIndex = match.index;
|
||||
}
|
||||
lastJobEndIndex = jobRegex.lastIndex;
|
||||
} catch (e) {
|
||||
console.warn('JSON 解析失败', e);
|
||||
}
|
||||
}
|
||||
|
||||
// 提取引导语(第一个 job-json 之前的文字)
|
||||
const guideText = firstJobStartIndex > 0 ?
|
||||
markdown.slice(0, firstJobStartIndex).trim() :
|
||||
'';
|
||||
|
||||
// 提取结束语(最后一个 job-json 之后的文字)
|
||||
const endingText = lastJobEndIndex < markdown.length ?
|
||||
markdown.slice(lastJobEndIndex).trim() :
|
||||
'';
|
||||
|
||||
// 岗位信息格式化为语音文本
|
||||
const jobTexts = jobs.map((job, index) => {
|
||||
return `第 ${index + 1} 个岗位,岗位名称是:${job.jobTitle},公司是:${job.companyName},薪资:${job.salary},地点:${job.location},学历要求:${job.education},经验要求:${job.experience}。`;
|
||||
});
|
||||
|
||||
// 拼接总语音内容
|
||||
const finalTextParts = [];
|
||||
if (guideText) finalTextParts.push(guideText);
|
||||
finalTextParts.push(...jobTexts);
|
||||
if (endingText) finalTextParts.push(endingText);
|
||||
|
||||
return finalTextParts.join('\n');
|
||||
}
|
Reference in New Issue
Block a user