mirror of
https://github.com/fromchat-messenger/web.git
synced 2026-09-23 03:25:07 +03:00
Implement audio calls
This commit is contained in:
@@ -0,0 +1,193 @@
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import { useAppState } from "../state";
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import * as WebRTC from "@/core/calls/webrtc";
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import { CallSignalingHandler } from "@/core/calls/signaling";
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import { setCallSignalingHandler } from "@/core/websocket";
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import { createRef, useEffect } from "react";
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// Global audio ref shared across all instances
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let globalRemoteAudioRef = createRef<HTMLAudioElement>();
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export default function useAudioCall() {
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const { chat, startCall, endCall, setCallStatus, toggleMute, user } = useAppState();
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const remoteAudioRef = globalRemoteAudioRef;
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useEffect(() => {
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if (user.authToken) {
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WebRTC.setAuthToken(user.authToken);
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}
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// Initialize call signaling handler
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const signalingHandler = new CallSignalingHandler(() => ({
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receiveCall: (userId: number, username: string) => {
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// Use the receiveCall function from state
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const state = useAppState.getState();
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state.receiveCall(userId, username);
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},
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endCall
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}));
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setCallSignalingHandler(signalingHandler);
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// Set up call state change handler
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WebRTC.setCallStateChangeHandler((userId: number, state: string) => {
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const call = chat.call;
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if (call.remoteUserId === userId) {
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switch (state) {
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case "connecting":
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setCallStatus("connecting");
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break;
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case "connected":
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setCallStatus("active");
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break;
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case "disconnected":
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case "failed":
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case "closed":
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endCall();
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break;
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}
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}
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});
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// Set up remote stream handler
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WebRTC.setRemoteStreamHandler((_userId: number, stream: MediaStream) => {
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if (!remoteAudioRef.current) {
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return;
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}
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const el = remoteAudioRef.current;
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try {
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el.srcObject = stream;
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el.muted = false;
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el.volume = 1.0;
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el.autoplay = true;
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// Handle audio events
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el.addEventListener("error", () => {
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console.warn("[AUDIO] element error", (el.error?.message) || el.error);
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});
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el.play().catch(() => {
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// Try to play after user interaction if autoplay is blocked
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const playAfterInteraction = () => {
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el.play().catch(() => {});
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document.removeEventListener('click', playAfterInteraction);
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document.removeEventListener('touchstart', playAfterInteraction);
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};
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document.addEventListener('click', playAfterInteraction);
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document.addEventListener('touchstart', playAfterInteraction);
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});
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} catch (e) {
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console.warn("failed to attach remote stream:", e);
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}
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});
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return () => {
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WebRTC.cleanup();
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setCallSignalingHandler(null);
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};
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}, [user.authToken, chat.call.remoteUserId, setCallStatus, endCall, startCall]);
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async function requestAudioPermissions(): Promise<boolean> {
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try {
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const stream = await navigator.mediaDevices.getUserMedia({
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audio: true,
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video: false
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});
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// Stop the stream immediately as we just needed permission
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stream.getTracks().forEach(track => track.stop());
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return true;
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} catch (error) {
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console.error("Failed to get audio permissions:", error);
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return false;
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}
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};
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async function initiateCall(userId: number, username: string) {
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const hasPermission = await requestAudioPermissions();
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if (!hasPermission) {
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console.log("Audio permission denied");
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return;
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}
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// Start the call in state
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startCall(userId, username);
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setCallStatus("calling");
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// Initiate WebRTC call
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const success = await WebRTC.initiateCall(userId, username);
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if (!success) {
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endCall();
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}
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};
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async function acceptCall() {
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if (!chat.call.remoteUserId) {
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return;
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}
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setCallStatus("connecting");
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const success = await WebRTC.acceptCall(chat.call.remoteUserId);
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if (!success) {
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endCall();
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}
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};
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async function rejectCall() {
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if (!chat.call.remoteUserId) {
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return;
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}
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await WebRTC.rejectCall(chat.call.remoteUserId);
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endCall();
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};
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async function handleEndCall() {
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if (chat.call.remoteUserId) {
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await WebRTC.endCall(chat.call.remoteUserId);
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}
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endCall();
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};
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function handleToggleMute() {
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if (chat.call.remoteUserId) {
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const isMuted = WebRTC.toggleMute(chat.call.remoteUserId);
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// Update mute state in store
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if (isMuted !== chat.call.isMuted) {
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toggleMute();
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}
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}
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};
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async function handleIncomingCall(userId: number, username: string) {
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await WebRTC.handleIncomingCall(userId, username);
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};
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async function handleCallOffer(userId: number, offer: RTCSessionDescriptionInit) {
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await WebRTC.handleCallOffer(userId, offer);
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};
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async function handleCallAnswer(userId: number, answer: RTCSessionDescriptionInit) {
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await WebRTC.handleCallAnswer(userId, answer);
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};
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async function handleIceCandidate(userId: number, candidate: RTCIceCandidateInit) {
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await WebRTC.handleIceCandidate(userId, candidate);
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};
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return {
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call: chat.call,
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initiateCall,
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acceptCall,
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rejectCall,
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endCall: handleEndCall,
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toggleMute: handleToggleMute,
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handleIncomingCall,
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handleCallOffer,
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handleCallAnswer,
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handleIceCandidate,
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remoteAudioRef
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};
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}
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