mirror of
https://github.com/danny-avila/LibreChat.git
synced 2026-02-08 18:44:22 +01:00
✨ feat: implement AudioSocketModule and WebRTCHandler for audio streaming; refactor SocketIOService to support module-based event handling
This commit is contained in:
parent
77ca00c87b
commit
2eda62cf67
4 changed files with 292 additions and 156 deletions
40
api/server/services/Files/Audio/AudioSocketModule.js
Normal file
40
api/server/services/Files/Audio/AudioSocketModule.js
Normal file
|
|
@ -0,0 +1,40 @@
|
|||
const { AudioHandler } = require('./WebRTCHandler');
|
||||
const { logger } = require('~/config');
|
||||
|
||||
class AudioSocketModule {
|
||||
constructor(socketIOService) {
|
||||
this.socketIOService = socketIOService;
|
||||
this.audioHandler = new AudioHandler();
|
||||
|
||||
this.moduleId = 'audio-handler';
|
||||
this.registerHandlers();
|
||||
}
|
||||
|
||||
registerHandlers() {
|
||||
this.socketIOService.registerModule(this.moduleId, {
|
||||
connection: (socket) => this.handleConnection(socket),
|
||||
disconnect: (socket) => this.handleDisconnect(socket),
|
||||
});
|
||||
}
|
||||
|
||||
handleConnection(socket) {
|
||||
// Register WebRTC-specific event handlers for this socket
|
||||
this.audioHandler.registerSocketHandlers(socket, this.config);
|
||||
|
||||
logger.debug(`Audio handler registered for client: ${socket.id}`);
|
||||
}
|
||||
|
||||
handleDisconnect(socket) {
|
||||
// Cleanup audio resources for disconnected client
|
||||
this.audioHandler.cleanup(socket.id);
|
||||
logger.debug(`Audio handler cleaned up for client: ${socket.id}`);
|
||||
}
|
||||
|
||||
// Used for app shutdown
|
||||
cleanup() {
|
||||
this.audioHandler.cleanupAll();
|
||||
this.socketIOService.unregisterModule(this.moduleId);
|
||||
}
|
||||
}
|
||||
|
||||
module.exports = { AudioSocketModule };
|
||||
179
api/server/services/Files/Audio/WebRTCHandler.js
Normal file
179
api/server/services/Files/Audio/WebRTCHandler.js
Normal file
|
|
@ -0,0 +1,179 @@
|
|||
const { RTCPeerConnection, RTCIceCandidate, MediaStream } = require('wrtc');
|
||||
const { logger } = require('~/config');
|
||||
|
||||
class WebRTCConnection {
|
||||
constructor(socket, config) {
|
||||
this.socket = socket;
|
||||
this.config = config;
|
||||
this.peerConnection = null;
|
||||
this.audioTransceiver = null;
|
||||
this.pendingCandidates = [];
|
||||
this.state = 'idle';
|
||||
}
|
||||
|
||||
async handleOffer(offer) {
|
||||
try {
|
||||
if (!this.peerConnection) {
|
||||
this.peerConnection = new RTCPeerConnection(this.config.rtcConfig);
|
||||
this.setupPeerConnectionListeners();
|
||||
}
|
||||
|
||||
await this.peerConnection.setRemoteDescription(offer);
|
||||
|
||||
const mediaStream = new MediaStream();
|
||||
|
||||
this.audioTransceiver = this.peerConnection.addTransceiver('audio', {
|
||||
direction: 'sendrecv',
|
||||
streams: [mediaStream],
|
||||
});
|
||||
|
||||
const answer = await this.peerConnection.createAnswer();
|
||||
await this.peerConnection.setLocalDescription(answer);
|
||||
this.socket.emit('webrtc-answer', answer);
|
||||
} catch (error) {
|
||||
logger.error(`Error handling offer: ${error}`);
|
||||
this.socket.emit('webrtc-error', {
|
||||
message: error.message,
|
||||
code: 'OFFER_ERROR',
|
||||
});
|
||||
}
|
||||
}
|
||||
|
||||
setupPeerConnectionListeners() {
|
||||
if (!this.peerConnection) {
|
||||
return;
|
||||
}
|
||||
|
||||
this.peerConnection.ontrack = ({ track }) => {
|
||||
logger.info(`Received ${track.kind} track from client`);
|
||||
|
||||
if (track.kind === 'audio') {
|
||||
this.handleIncomingAudio(track);
|
||||
}
|
||||
|
||||
track.onended = () => {
|
||||
logger.info(`${track.kind} track ended`);
|
||||
};
|
||||
};
|
||||
|
||||
this.peerConnection.onicecandidate = ({ candidate }) => {
|
||||
if (candidate) {
|
||||
this.socket.emit('icecandidate', candidate);
|
||||
}
|
||||
};
|
||||
|
||||
this.peerConnection.onconnectionstatechange = () => {
|
||||
if (!this.peerConnection) {
|
||||
return;
|
||||
}
|
||||
|
||||
const state = this.peerConnection.connectionState;
|
||||
logger.info(`Connection state changed to ${state}`);
|
||||
this.state = state;
|
||||
|
||||
if (state === 'failed' || state === 'closed') {
|
||||
this.cleanup();
|
||||
}
|
||||
};
|
||||
}
|
||||
|
||||
handleIncomingAudio(track) {
|
||||
if (this.peerConnection) {
|
||||
const stream = new MediaStream([track]);
|
||||
this.peerConnection.addTrack(track, stream);
|
||||
}
|
||||
}
|
||||
|
||||
async addIceCandidate(candidate) {
|
||||
try {
|
||||
if (this.peerConnection?.remoteDescription) {
|
||||
if (candidate && candidate.candidate) {
|
||||
await this.peerConnection.addIceCandidate(new RTCIceCandidate(candidate));
|
||||
} else {
|
||||
logger.warn('Invalid ICE candidate');
|
||||
}
|
||||
} else {
|
||||
this.pendingCandidates.push(candidate);
|
||||
}
|
||||
} catch (error) {
|
||||
logger.error(`Error adding ICE candidate: ${error}`);
|
||||
}
|
||||
}
|
||||
|
||||
cleanup() {
|
||||
if (this.peerConnection) {
|
||||
try {
|
||||
this.peerConnection.close();
|
||||
} catch (error) {
|
||||
logger.error(`Error closing peer connection: ${error}`);
|
||||
}
|
||||
this.peerConnection = null;
|
||||
}
|
||||
|
||||
this.audioTransceiver = null;
|
||||
this.pendingCandidates = [];
|
||||
this.state = 'idle';
|
||||
}
|
||||
}
|
||||
|
||||
class AudioHandler {
|
||||
constructor() {
|
||||
this.connections = new Map();
|
||||
this.defaultRTCConfig = {
|
||||
iceServers: [
|
||||
{
|
||||
urls: ['stun:stun.l.google.com:19302', 'stun:stun1.l.google.com:19302'],
|
||||
},
|
||||
],
|
||||
iceCandidatePoolSize: 10,
|
||||
bundlePolicy: 'max-bundle',
|
||||
rtcpMuxPolicy: 'require',
|
||||
};
|
||||
}
|
||||
|
||||
registerSocketHandlers(socket) {
|
||||
const rtcConfig = {
|
||||
rtcConfig: this.defaultRTCConfig,
|
||||
};
|
||||
|
||||
const rtcConnection = new WebRTCConnection(socket, rtcConfig);
|
||||
this.connections.set(socket.id, rtcConnection);
|
||||
|
||||
socket.on('webrtc-offer', (offer) => {
|
||||
logger.debug(`Received WebRTC offer from ${socket.id}`);
|
||||
rtcConnection.handleOffer(offer);
|
||||
});
|
||||
|
||||
socket.on('icecandidate', (candidate) => {
|
||||
rtcConnection.addIceCandidate(candidate);
|
||||
});
|
||||
|
||||
socket.on('vad-status', (status) => {
|
||||
logger.debug(`VAD status from ${socket.id}: ${JSON.stringify(status)}`);
|
||||
});
|
||||
|
||||
socket.on('disconnect', () => {
|
||||
rtcConnection.cleanup();
|
||||
this.connections.delete(socket.id);
|
||||
});
|
||||
|
||||
return rtcConnection;
|
||||
}
|
||||
|
||||
cleanup(socketId) {
|
||||
const connection = this.connections.get(socketId);
|
||||
if (connection) {
|
||||
connection.cleanup();
|
||||
this.connections.delete(socketId);
|
||||
}
|
||||
}
|
||||
|
||||
cleanupAll() {
|
||||
for (const connection of this.connections.values()) {
|
||||
connection.cleanup();
|
||||
}
|
||||
this.connections.clear();
|
||||
}
|
||||
}
|
||||
|
||||
module.exports = { AudioHandler, WebRTCConnection };
|
||||
Loading…
Add table
Add a link
Reference in a new issue