mirror of
https://github.com/danny-avila/LibreChat.git
synced 2025-12-25 04:40:15 +01:00
✨ feat: Implement WebRTC messaging and audio handling in the WebRTC service
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parent
cf4b73b5e3
commit
9a33292f88
8 changed files with 674 additions and 137 deletions
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@ -1,12 +1,15 @@
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const { WebSocketServer } = require('ws');
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const fs = require('fs');
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const path = require('path');
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const { RTCPeerConnection } = require('wrtc');
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module.exports.WebSocketService = class {
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constructor(server) {
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this.wss = new WebSocketServer({ server, path: '/ws' });
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this.log('Server initialized');
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this.clientAudioBuffers = new Map();
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this.activeClients = new Map();
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this.iceServers = [
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{ urls: 'stun:stun.l.google.com:19302' },
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{ urls: 'stun:stun1.l.google.com:19302' },
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];
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this.setupHandlers();
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}
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@ -17,7 +20,13 @@ module.exports.WebSocketService = class {
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setupHandlers() {
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this.wss.on('connection', (ws) => {
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const clientId = Date.now().toString();
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this.clientAudioBuffers.set(clientId, []);
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this.activeClients.set(clientId, {
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ws,
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state: 'idle',
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audioBuffer: [],
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currentTranscription: '',
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isProcessing: false,
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});
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this.log(`Client connected: ${clientId}`);
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@ -29,42 +38,175 @@ module.exports.WebSocketService = class {
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return;
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}
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if (message.type === 'audio-chunk') {
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if (!this.clientAudioBuffers.has(clientId)) {
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this.clientAudioBuffers.set(clientId, []);
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}
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this.clientAudioBuffers.get(clientId).push(message.data);
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}
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switch (message.type) {
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case 'call-start':
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this.handleCallStart(clientId);
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break;
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if (message.type === 'request-response') {
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const filePath = path.join(__dirname, './assets/response.mp3');
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const audioFile = fs.readFileSync(filePath);
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ws.send(JSON.stringify({ type: 'audio-response', data: audioFile.toString('base64') }));
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}
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case 'audio-chunk':
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await this.handleAudioChunk(clientId, message.data);
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break;
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if (message.type === 'call-ended') {
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const allChunks = this.clientAudioBuffers.get(clientId);
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this.writeAudioFile(clientId, allChunks);
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this.clientAudioBuffers.delete(clientId);
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case 'processing-start':
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await this.processAudioStream(clientId);
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break;
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case 'audio-received':
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this.confirmAudioReceived(clientId);
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break;
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case 'call-ended':
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this.handleCallEnd(clientId);
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break;
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}
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});
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ws.on('close', () => {
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this.handleCallEnd(clientId);
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this.activeClients.delete(clientId);
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this.log(`Client disconnected: ${clientId}`);
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this.clientAudioBuffers.delete(clientId);
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});
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ws.on('error', (error) => {
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this.log(`Error for client ${clientId}: ${error.message}`);
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this.handleCallEnd(clientId);
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});
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});
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}
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writeAudioFile(clientId, base64Chunks) {
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if (!base64Chunks || base64Chunks.length === 0) {
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async handleCallStart(clientId) {
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const client = this.activeClients.get(clientId);
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if (!client) {
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return;
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}
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const filePath = path.join(__dirname, `recorded_${clientId}.webm`);
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const buffer = Buffer.concat(
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base64Chunks.map((chunk) => Buffer.from(chunk.split(',')[1], 'base64')),
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try {
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client.state = 'active';
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client.audioBuffer = [];
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client.currentTranscription = '';
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client.isProcessing = false;
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const peerConnection = new RTCPeerConnection({
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iceServers: this.iceServers,
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sdpSemantics: 'unified-plan',
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});
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client.peerConnection = peerConnection;
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client.dataChannel = peerConnection.createDataChannel('audio', {
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ordered: true,
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maxRetransmits: 3,
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});
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client.dataChannel.onopen = () => this.log(`Data channel opened for ${clientId}`);
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client.dataChannel.onmessage = async (event) => {
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await this.handleAudioChunk(clientId, event.data);
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};
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peerConnection.onicecandidate = (event) => {
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if (event.candidate) {
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client.ws.send(
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JSON.stringify({
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type: 'ice-candidate',
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candidate: event.candidate,
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}),
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);
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}
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};
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peerConnection.onnegotiationneeded = async () => {
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try {
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const offer = await peerConnection.createOffer();
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await peerConnection.setLocalDescription(offer);
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client.ws.send(
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JSON.stringify({
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type: 'webrtc-offer',
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sdp: peerConnection.localDescription,
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}),
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);
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} catch (error) {
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this.log(`Negotiation failed for ${clientId}: ${error}`);
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}
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};
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this.log(`Call started for client ${clientId}`);
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} catch (error) {
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this.log(`Error starting call for ${clientId}: ${error.message}`);
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this.handleCallEnd(clientId);
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}
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}
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async handleAudioChunk(clientId, data) {
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const client = this.activeClients.get(clientId);
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if (!client || client.state !== 'active') {
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return;
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}
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client.audioBuffer.push(data);
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client.ws.send(JSON.stringify({ type: 'audio-received' }));
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}
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async processAudioStream(clientId) {
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const client = this.activeClients.get(clientId);
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if (!client || client.state !== 'active' || client.isProcessing) {
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return;
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}
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client.isProcessing = true;
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try {
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// Process transcription
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client.ws.send(
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JSON.stringify({
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type: 'transcription',
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data: 'Processing audio...',
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}),
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);
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// Stream LLM response
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client.ws.send(
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JSON.stringify({
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type: 'llm-response',
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data: 'Processing response...',
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}),
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);
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// Stream TTS chunks
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client.ws.send(
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JSON.stringify({
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type: 'tts-chunk',
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data: 'audio_data_here',
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}),
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);
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} catch (error) {
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this.log(`Processing error for client ${clientId}: ${error.message}`);
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} finally {
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client.isProcessing = false;
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client.audioBuffer = [];
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}
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}
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confirmAudioReceived(clientId) {
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const client = this.activeClients.get(clientId);
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if (!client) {
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return;
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}
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client.ws.send(
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JSON.stringify({
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type: 'audio-received',
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data: null,
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}),
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);
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fs.writeFileSync(filePath, buffer);
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this.log(`Saved audio to ${filePath}`);
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}
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handleCallEnd(clientId) {
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const client = this.activeClients.get(clientId);
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if (!client) {
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return;
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}
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client.state = 'idle';
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client.audioBuffer = [];
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client.currentTranscription = '';
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}
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};
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