refactor(stores): reorganize flat store files into modular directory structure

Migrate from flat file organization to modular directory structure under src/stores/:

- auth/: authStore, registerStore, verificationStore, sessionStore
- call/: callStore
- settings/: chatSettingsStore, themeStore
- ui/: homeTabBarVisibilityStore, homeTabPressStore
- utils/: routePayloadCache
- group/sources.ts, group/profileResolver.ts
- message/sources.ts
- post/sources.ts

Update all import paths across components and screens to use new module paths. Maintain backward compatibility through deprecated re-export files for gradual migration.

src/stores/call/callStore.ts

@@ -0,0 +1,859 @@
+import { create } from 'zustand';
+import { MediaStream } from 'react-native-webrtc';
+import {
+  wsService,
+  WSCallIncomingMessage,
+  WSCallSDPMessage,
+  WSCallICEMessage,
+  WSErrorMessage,
+} from '@/services/core';
+import { webrtcManager, ICEServer } from '@/services/webrtc';
+import { getCurrentUserId } from '../auth/sessionStore';
+import { useUserStore } from '../userStore';
+import { userManager } from '../user';
+
+export type CallStatus = 
+  | 'idle'           // 空闲状态
+  | 'calling'        // 正在呼出（已发送邀请，等待对方响应）
+  | 'ringing'        // 来电响铃中
+  | 'connecting'     // 正在建立连接（WebRTC 协商中）
+  | 'connected'      // 已接通
+  | 'reconnecting'   // 网络断开，正在重连
+  | 'ended'          // 已结束
+  | 'failed';        // 连接失败
+
+export type CallType = 'voice' | 'video';
+
+export interface CallSession {
+  id: string;
+  conversationId: string;
+  peerId: string;
+  peerName?: string;
+  peerAvatar?: string | null;
+  status: CallStatus;
+  callType: CallType;
+  startedAt?: number;
+  duration: number;
+  isMuted: boolean;
+  isSpeakerOn: boolean;
+  isVideoEnabled: boolean;
+  isPeerVideoEnabled: boolean;
+  isInitiator: boolean;
+}
+
+export interface IncomingCallInfo {
+  callId: string;
+  conversationId: string;
+  callerId: string;
+  callerName?: string;
+  callerAvatar?: string | null;
+  callType: string;
+  iceServers: ICEServer[];
+  receivedAt: number;
+  lifetime?: number;
+}
+
+interface CallState {
+  currentCall: CallSession | null;
+  incomingCall: IncomingCallInfo | null;
+  callDuration: number;
+  peerStream: MediaStream | null;
+  localStream: MediaStream | null;
+  isMinimized: boolean;
+
+  initCall: () => () => void;
+  startCall: (
+    conversationId: string,
+    calleeId: string,
+    calleeInfo?: { nickname?: string; username?: string; avatar?: string | null },
+    callType?: CallType
+  ) => Promise<void>;
+  acceptCall: () => Promise<void>;
+  rejectCall: () => void;
+  endCall: (reason?: string) => Promise<void>;
+  toggleMute: () => void;
+  toggleSpeaker: () => void;
+  toggleMinimize: () => void;
+  toggleVideo: () => Promise<void>;
+  setVideoEnabled: (enabled: boolean) => Promise<void>;
+  reset: () => void;
+}
+
+// Module-level variables for timers
+let durationTimer: ReturnType<typeof setInterval> | null = null;
+let callTimeoutTimer: ReturnType<typeof setTimeout> | null = null;
+let initCallUnsub: (() => void) | null = null;
+let unsubInvited: (() => void) | null = null;
+let pendingOffer: { callId: string; sdp: string } | null = null;
+let rtcUnsubscribe: (() => void) | null = null;  // WebRTC event subscription
+
+// Constants
+const CALL_LIFETIME_MS = 55000;
+const CALL_TIMEOUT_MS = 115000;
+const IGNORE_CALL_ID_TTL = 30000;
+
+// Track processed call IDs to prevent duplicates
+const processedCallIds = new Map<string, number>();
+
+function cleanupProcessedCallIds() {
+  const now = Date.now();
+  for (const [callId, timestamp] of processedCallIds) {
+    if (now - timestamp > IGNORE_CALL_ID_TTL) {
+      processedCallIds.delete(callId);
+    }
+  }
+}
+
+/**
+ * Unified WebRTC event handler
+ * This is called from both initiator and receiver paths
+ */
+function setupWebRTCEvents(callId: string, myUserId: string): void {
+  // Clean up any existing subscription
+  if (rtcUnsubscribe) {
+    rtcUnsubscribe();
+    rtcUnsubscribe = null;
+  }
+
+  rtcUnsubscribe = webrtcManager.onEvent((event) => {
+    switch (event.type) {
+      case 'icecandidate':
+        if (event.candidate) {
+          wsService.sendCallICE(callId, JSON.stringify(event.candidate.toJSON()));
+        }
+        break;
+
+      case 'remotestream':
+        handleRemoteStream(event.stream);
+        break;
+
+      case 'negotiationneeded':
+        handleNegotiationNeeded(callId, event.offer);
+        break;
+
+      case 'connectionstatechange':
+        handleConnectionStateChange(event.state);
+        break;
+
+      case 'error':
+        console.error('[CallStore] WebRTC error:', event.error);
+        break;
+    }
+  });
+}
+
+/**
+ * Handle remote stream with video track detection
+ */
+function handleRemoteStream(stream: MediaStream): void {
+  callStore.setState({ peerStream: stream });
+
+  // Detect video tracks in remote stream
+  const videoTracks = stream.getVideoTracks();
+  const hasPeerVideo = videoTracks.length > 0 && videoTracks.some((t) => t.enabled);
+
+  console.log('[CallStore] Remote stream received, hasVideo:', hasPeerVideo, 'videoTracks:', videoTracks.length);
+
+  callStore.setState((s) => ({
+    currentCall: s.currentCall
+      ? { ...s.currentCall, isPeerVideoEnabled: hasPeerVideo }
+      : null,
+  }));
+}
+
+/**
+ * Handle negotiation needed event - send offer to peer
+ */
+function handleNegotiationNeeded(callId: string, offer: RTCSessionDescriptionInit): void {
+  console.log('[CallStore] Negotiation needed, sending offer');
+  wsService.sendCallSDP(callId, 'offer', offer.sdp || '');
+}
+
+  /**
+   * Handle connection state change with enhanced state machine
+   */
+  function handleConnectionStateChange(state: string): void {
+    console.log('[CallStore] Connection state changed:', state);
+
+    const { currentCall } = callStore.getState();
+    if (!currentCall) return;
+
+    switch (state) {
+      case 'connected':
+        // 连接成功，开始计时
+        const now = Date.now();
+        callStore.setState((s) => ({
+          currentCall: s.currentCall
+            ? { ...s.currentCall, status: 'connected', startedAt: now }
+            : null,
+        }));
+
+        if (durationTimer) clearInterval(durationTimer);
+        durationTimer = setInterval(() => {
+          callStore.setState((s) => ({ callDuration: s.callDuration + 1 }));
+        }, 1000);
+        break;
+
+      case 'disconnected':
+        // 临时断开，进入重连状态
+        if (currentCall.status === 'connected') {
+          callStore.setState((s) => ({
+            currentCall: s.currentCall
+              ? { ...s.currentCall, status: 'reconnecting' }
+              : null,
+          }));
+        }
+        break;
+
+      case 'failed':
+        // 连接失败
+        if (currentCall.status === 'connected' || currentCall.status === 'reconnecting') {
+          callStore.getState().endCall('connection_failed');
+        } else if (currentCall.status === 'connecting') {
+          // 初始连接失败
+          callStore.getState().endCall('connection_failed');
+        }
+        break;
+
+      case 'closed':
+        // 连接关闭
+        if (currentCall.status !== 'ended' && currentCall.status !== 'failed') {
+          callStore.getState().endCall('connection_closed');
+        }
+        break;
+    }
+  }
+
+/**
+ * Handle incoming SDP offer with Glare handling
+ */
+async function handleIncomingOffer(msg: WSCallSDPMessage, myUserId: string): Promise<void> {
+  let pc = webrtcManager.getPeerConnection();
+  if (!pc) {
+    // Cache offer for later processing
+    pendingOffer = { callId: msg.call_id, sdp: msg.payload.sdp };
+    console.log('[CallStore] Caching pending offer, PeerConnection not ready yet');
+    return;
+  }
+
+  let signalingState = pc.signalingState;
+  console.log('[CallStore] Received offer, signalingState:', signalingState);
+
+  // Glare handling: if we're not in stable state, we have a conflict
+  if (signalingState !== 'stable') {
+    const remoteUserId = msg.from_id;
+
+    // Compare user IDs to determine who wins
+    // Higher user ID wins the negotiation
+    if (myUserId > remoteUserId) {
+      console.log('[CallStore] Glare: I win (my ID > remote ID), ignoring incoming offer');
+      return;
+    }
+
+    console.log('[CallStore] Glare: Remote wins (remote ID > my ID), rolling back');
+
+    // Rollback to stable state
+    try {
+      await webrtcManager.rollback();
+    } catch (err) {
+      console.error('[CallStore] Rollback failed:', err);
+      // If rollback fails because we're already stable, that's fine - just proceed
+      if (pc.signalingState !== 'stable') {
+        return;
+      }
+    }
+
+    // Re-check state after rollback
+    signalingState = pc.signalingState;
+    if (signalingState !== 'stable') {
+      console.warn('[CallStore] Not stable after rollback, state:', signalingState);
+      return;
+    }
+  }
+
+  pendingOffer = null;
+
+  // Set remote description and create answer
+  try {
+    await webrtcManager.setRemoteDescription({
+      type: msg.payload.sdp_type as 'offer' | 'answer',
+      sdp: msg.payload.sdp,
+    });
+
+    const answer = await webrtcManager.createAnswer();
+    wsService.sendCallSDP(msg.call_id, 'answer', answer.sdp || '');
+  } catch (err) {
+    console.error('[CallStore] Failed to handle incoming offer:', err);
+  }
+}
+
+/**
+ * Handle incoming SDP answer
+ */
+async function handleIncomingAnswer(msg: WSCallSDPMessage): Promise<void> {
+  const pc = webrtcManager.getPeerConnection();
+  if (!pc) return;
+
+  const signalingState = pc.signalingState;
+  console.log('[CallStore] Received answer, signalingState:', signalingState);
+
+  if (signalingState !== 'have-local-offer') {
+    console.warn('[CallStore] Ignoring answer, signaling state is', signalingState);
+    return;
+  }
+
+  try {
+    await webrtcManager.setRemoteDescription({
+      type: msg.payload.sdp_type as 'offer' | 'answer',
+      sdp: msg.payload.sdp,
+    });
+  } catch (err) {
+    console.error('[CallStore] Failed to set remote description from answer:', err);
+  }
+}
+
+/**
+ * Clean up all resources
+ */
+function cleanupResources(): void {
+  if (callTimeoutTimer) {
+    clearTimeout(callTimeoutTimer);
+    callTimeoutTimer = null;
+  }
+  if (durationTimer) {
+    clearInterval(durationTimer);
+    durationTimer = null;
+  }
+  if (rtcUnsubscribe) {
+    rtcUnsubscribe();
+    rtcUnsubscribe = null;
+  }
+  pendingOffer = null;
+}
+
+export const callStore = create<CallState>((set, get) => ({
+  currentCall: null,
+  incomingCall: null,
+  callDuration: 0,
+  peerStream: null,
+  localStream: null,
+  isMinimized: false,
+
+  initCall: () => {
+    // Prevent duplicate handler registration
+    if (initCallUnsub) {
+      initCallUnsub();
+      initCallUnsub = null;
+    }
+
+    const unsubs: Array<() => void> = [];
+
+    unsubs.push(
+      wsService.on('call_incoming', async (msg) => {
+        cleanupProcessedCallIds();
+
+        if (processedCallIds.has(msg.call_id)) {
+          console.log('[CallStore] Ignoring already processed call:', msg.call_id);
+          return;
+        }
+
+        const { currentCall, incomingCall } = get();
+        if (incomingCall) {
+          wsService.sendCallBusy(msg.call_id);
+          processedCallIds.set(msg.call_id, Date.now());
+          return;
+        }
+        if (currentCall && currentCall.status !== 'idle') {
+          wsService.sendCallBusy(msg.call_id);
+          processedCallIds.set(msg.call_id, Date.now());
+          return;
+        }
+
+        // Check lifetime expiry
+        const callAge = Date.now() - msg.created_at;
+        const lifetime = msg.lifetime || 60000;
+        if (callAge > lifetime - 5000) {
+          console.log('[CallStore] Ignoring stale incoming call, age:', callAge);
+          wsService.sendCallReject(msg.call_id);
+          processedCallIds.set(msg.call_id, Date.now());
+          return;
+        }
+
+        // Fetch caller info
+        let caller: { nickname?: string; username?: string; avatar?: string | null } | null =
+          useUserStore.getState().userCache[msg.caller_id];
+        if (!caller) {
+          try {
+            const fetchedCaller = await userManager.getUserById(msg.caller_id);
+            if (fetchedCaller) {
+              caller = {
+                nickname: fetchedCaller.nickname,
+                username: fetchedCaller.username,
+                avatar: fetchedCaller.avatar,
+              };
+            }
+          } catch (err) {
+            console.log('[CallStore] Failed to fetch caller info:', err);
+          }
+        }
+        const callerName = caller?.nickname || caller?.username || msg.caller_id;
+
+        set({
+          incomingCall: {
+            callId: msg.call_id,
+            conversationId: msg.conversation_id,
+            callerId: msg.caller_id,
+            callerName,
+            callerAvatar: caller?.avatar,
+            callType: msg.call_type,
+            iceServers: msg.ice_servers || [],
+            receivedAt: Date.now(),
+            lifetime: msg.lifetime,
+          },
+        });
+
+        processedCallIds.set(msg.call_id, Date.now());
+
+        // Set timeout based on lifetime
+        if (callTimeoutTimer) clearTimeout(callTimeoutTimer);
+        const remainingTime = Math.max(lifetime - callAge - 1000, 5000);
+        callTimeoutTimer = setTimeout(() => {
+          const { incomingCall: ic } = get();
+          if (ic?.callId === msg.call_id) {
+            console.log('[CallStore] Incoming call timeout');
+            wsService.sendCallReject(msg.call_id);
+            processedCallIds.set(msg.call_id, Date.now());
+            set({ incomingCall: null });
+          }
+        }, remainingTime);
+      })
+    );
+
+    unsubs.push(
+      wsService.on('call_accepted', async (msg) => {
+        const { currentCall } = get();
+        if (!currentCall || currentCall.id !== msg.call_id) return;
+
+        if (!currentCall.isInitiator) {
+          console.log('[CallStore] Ignoring call_accepted, we are not the initiator');
+          return;
+        }
+
+        if (callTimeoutTimer) {
+          clearTimeout(callTimeoutTimer);
+          callTimeoutTimer = null;
+        }
+
+        const myUserId = getCurrentUserId() || '';
+
+        try {
+          const isVideoCall = currentCall.callType === 'video';
+          await webrtcManager.initialize(msg.ice_servers || []);
+          const newStream = await webrtcManager.createLocalStream(!isVideoCall);
+
+          set({ localStream: newStream });
+
+          // Setup unified WebRTC event handler
+          setupWebRTCEvents(currentCall.id, myUserId);
+
+          // Start call with transceiver-based approach
+          const offer = await webrtcManager.startCall(true, currentCall.callType);
+          if (offer) {
+            wsService.sendCallSDP(currentCall.id, 'offer', offer.sdp || '');
+          }
+        } catch (err) {
+          console.error('[CallStore] call_accepted error:', err);
+          get().endCall('connection_failed');
+        }
+      })
+    );
+
+    unsubs.push(
+      wsService.on('call_rejected', (msg) => {
+        const { currentCall } = get();
+        if (currentCall?.id !== msg.call_id) return;
+        console.log('[CallStore] Call rejected');
+        get().endCall('rejected');
+      })
+    );
+
+    unsubs.push(
+      wsService.on('call_busy', (msg) => {
+        const { currentCall } = get();
+        if (currentCall?.id !== msg.call_id) return;
+        console.log('[CallStore] Call busy');
+        get().endCall('busy');
+      })
+    );
+
+    unsubs.push(
+      wsService.on('call_ended', (msg) => {
+        const { currentCall, incomingCall } = get();
+
+        processedCallIds.set(msg.call_id, Date.now());
+
+        if (currentCall?.id === msg.call_id) {
+          console.log('[CallStore] Call ended, duration:', msg.duration);
+          get().endCall('ended');
+          return;
+        }
+
+        if (incomingCall?.callId === msg.call_id) {
+          console.log('[CallStore] Incoming call cancelled by caller');
+          if (callTimeoutTimer) {
+            clearTimeout(callTimeoutTimer);
+            callTimeoutTimer = null;
+          }
+          set({ incomingCall: null });
+        }
+      })
+    );
+
+    unsubs.push(
+      wsService.on('call_answered_elsewhere', (msg) => {
+        const { incomingCall } = get();
+        if (incomingCall?.callId === msg.call_id) {
+          console.log('[CallStore] Call answered on another device');
+          processedCallIds.set(msg.call_id, Date.now());
+          if (callTimeoutTimer) {
+            clearTimeout(callTimeoutTimer);
+            callTimeoutTimer = null;
+          }
+          set({ incomingCall: null });
+        }
+      })
+    );
+
+    unsubs.push(
+      wsService.on('error', (msg: WSErrorMessage) => {
+        const { currentCall, incomingCall } = get();
+        console.log('[CallStore] Server error:', msg.code, msg.message);
+
+        if (msg.code === 'callee_offline') {
+          if (currentCall && currentCall.status === 'ringing') {
+            console.log('[CallStore] Callee is offline');
+            get().endCall('callee_offline');
+          }
+        }
+
+        if (msg.code === 'call_already_answered') {
+          if (incomingCall) {
+            console.log('[CallStore] Call already answered on another device');
+            processedCallIds.set(incomingCall.callId, Date.now());
+            if (callTimeoutTimer) {
+              clearTimeout(callTimeoutTimer);
+              callTimeoutTimer = null;
+            }
+            set({ incomingCall: null });
+          }
+        }
+      })
+    );
+
+    unsubs.push(
+      wsService.on('call_sdp', async (msg: WSCallSDPMessage) => {
+        const { currentCall } = get();
+        if (!currentCall || currentCall.id !== msg.call_id) return;
+
+        const myUserId = getCurrentUserId() || '';
+        if (msg.from_id === myUserId) return;
+
+        if (msg.payload.sdp_type === 'offer') {
+          await handleIncomingOffer(msg, myUserId);
+        } else if (msg.payload.sdp_type === 'answer') {
+          await handleIncomingAnswer(msg);
+        }
+      })
+    );
+
+    unsubs.push(
+      wsService.on('call_ice', (msg: WSCallICEMessage) => {
+        const { currentCall } = get();
+        if (!currentCall || currentCall.id !== msg.call_id) return;
+
+        const myUserId = getCurrentUserId() || '';
+        if (msg.from_id === myUserId) return;
+
+        try {
+          const candidate = typeof msg.payload.candidate === 'string'
+            ? JSON.parse(msg.payload.candidate)
+            : msg.payload.candidate;
+          webrtcManager.addIceCandidate(candidate);
+        } catch (err) {
+          console.error('[CallStore] call_ice error:', err);
+        }
+      })
+    );
+
+    unsubs.push(
+      wsService.on('call_peer_muted', (msg) => {
+        console.log('[CallStore] Peer muted:', msg.user_id, msg.muted);
+      })
+    );
+
+    const cleanup = () => {
+      cleanupResources();
+      if (unsubInvited) {
+        unsubInvited();
+        unsubInvited = null;
+      }
+      unsubs.forEach((unsub) => unsub());
+      initCallUnsub = null;
+    };
+    initCallUnsub = cleanup;
+    return cleanup;
+  },
+
+  startCall: async (
+    conversationId: string,
+    calleeId: string,
+    calleeInfo?: { nickname?: string; username?: string; avatar?: string | null },
+    callType: CallType = 'voice'
+  ) => {
+    const { currentCall } = get();
+    if (currentCall && currentCall.status !== 'idle') {
+      console.warn('[CallStore] Already in a call');
+      return;
+    }
+
+    const myUserId = getCurrentUserId();
+    if (!myUserId) {
+      console.error('[CallStore] Not logged in');
+      return;
+    }
+
+    const cachedCallee = useUserStore.getState().userCache[calleeId];
+    const callee = calleeInfo || cachedCallee;
+    const calleeName = callee?.nickname || callee?.username || calleeId;
+
+    set({
+      currentCall: {
+        id: '',
+        conversationId,
+        peerId: calleeId,
+        peerName: calleeName,
+        peerAvatar: callee?.avatar,
+        callType,
+        status: 'calling',  // 改为 'calling' 表示正在呼出
+        duration: 0,
+        isMuted: false,
+        isSpeakerOn: false,
+        isVideoEnabled: callType === 'video',
+        isPeerVideoEnabled: false,
+        isInitiator: true,
+      },
+    });
+
+    wsService.sendCallInvite(conversationId, calleeId, callType);
+
+    if (unsubInvited) {
+      unsubInvited();
+    }
+    unsubInvited = wsService.on('call_invited', (msg) => {
+      set((s) => ({
+        currentCall: s.currentCall
+          ? { ...s.currentCall, id: msg.call_id }
+          : null,
+      }));
+    });
+
+    if (callTimeoutTimer) clearTimeout(callTimeoutTimer);
+    callTimeoutTimer = setTimeout(() => {
+      const { currentCall: cc } = get();
+      // 只有在 'calling' 状态（呼出中）才超时
+      if (cc && cc.status === 'calling') {
+        console.warn('[CallStore] Call timeout');
+        get().endCall('timeout');
+      }
+    }, CALL_TIMEOUT_MS);
+  },
+
+  acceptCall: async () => {
+    const { incomingCall } = get();
+    if (!incomingCall) return;
+
+    console.log('[CallStore] acceptCall, incomingCall.callType:', incomingCall.callType);
+
+    if (callTimeoutTimer) {
+      clearTimeout(callTimeoutTimer);
+      callTimeoutTimer = null;
+    }
+
+    const isVideoCall = incomingCall.callType === 'video';
+    console.log('[CallStore] acceptCall, isVideoCall:', isVideoCall);
+    const myUserId = getCurrentUserId() || '';
+
+    set({
+      currentCall: {
+        id: incomingCall.callId,
+        conversationId: incomingCall.conversationId,
+        peerId: incomingCall.callerId,
+        peerName: incomingCall.callerName,
+        peerAvatar: incomingCall.callerAvatar,
+        callType: incomingCall.callType as CallType,
+        status: 'connecting',
+        duration: 0,
+        isMuted: false,
+        isSpeakerOn: false,
+        isVideoEnabled: isVideoCall,
+        isPeerVideoEnabled: isVideoCall,
+        isInitiator: false,
+      },
+      incomingCall: null,
+    });
+
+    wsService.sendCallAnswer(incomingCall.callId);
+
+    try {
+      await webrtcManager.initialize(incomingCall.iceServers);
+      const newStream = await webrtcManager.createLocalStream(!isVideoCall);
+
+      set({ localStream: newStream });
+
+      // Setup unified WebRTC event handler
+      setupWebRTCEvents(incomingCall.callId, myUserId);
+
+      // Start call (non-initiator, will wait for offer)
+      await webrtcManager.startCall(false, incomingCall.callType as CallType);
+
+      // Note: For non-initiator, we don't create an offer here.
+      // We wait for the initiator's offer via handleIncomingOffer.
+    } catch (err) {
+      console.error('[CallStore] Failed to accept call:', err);
+      get().endCall('connection_failed');
+    }
+  },
+
+  rejectCall: () => {
+    const { incomingCall } = get();
+    if (!incomingCall) return;
+
+    if (callTimeoutTimer) {
+      clearTimeout(callTimeoutTimer);
+      callTimeoutTimer = null;
+    }
+
+    wsService.sendCallReject(incomingCall.callId);
+    set({ incomingCall: null });
+  },
+
+  endCall: async (reason = 'ended') => {
+    const { currentCall } = get();
+    if (!currentCall) return;
+
+    cleanupResources();
+
+    if (unsubInvited) {
+      unsubInvited();
+      unsubInvited = null;
+    }
+
+    const callId = currentCall.id;
+
+    set({
+      currentCall: null,
+      callDuration: 0,
+      peerStream: null,
+      localStream: null,
+    });
+
+    webrtcManager.dispose();
+
+    if (callId && reason !== 'ended') {
+      wsService.sendCallEnd(callId, reason);
+    }
+  },
+
+  toggleMute: () => {
+    const { currentCall } = get();
+    if (!currentCall) return;
+
+    const newMuted = !currentCall.isMuted;
+    webrtcManager.setMuted(newMuted);
+    if (currentCall.id) {
+      wsService.sendCallMute(currentCall.id, newMuted);
+    }
+    set((s) => ({
+      currentCall: s.currentCall ? { ...s.currentCall, isMuted: newMuted } : null,
+    }));
+  },
+
+  toggleSpeaker: () => {
+    const { currentCall } = get();
+    if (!currentCall) return;
+    set((s) => ({
+      currentCall: s.currentCall
+        ? { ...s.currentCall, isSpeakerOn: !s.currentCall.isSpeakerOn }
+        : null,
+    }));
+  },
+
+  toggleMinimize: () => {
+    set((s) => ({ isMinimized: !s.isMinimized }));
+  },
+
+  toggleVideo: async () => {
+    const { currentCall } = get();
+    if (!currentCall) return;
+
+    const newVideoEnabled = !currentCall.isVideoEnabled;
+    await get().setVideoEnabled(newVideoEnabled);
+  },
+
+  setVideoEnabled: async (enabled: boolean) => {
+    const { currentCall } = get();
+    if (!currentCall) return;
+
+    try {
+      if (enabled) {
+        const newStream = await webrtcManager.enableVideo();
+        set({ localStream: newStream });
+      } else {
+        const newStream = await webrtcManager.disableVideo();
+        set({ localStream: newStream });
+      }
+
+      set((s) => ({
+        currentCall: s.currentCall
+          ? { ...s.currentCall, isVideoEnabled: enabled, callType: enabled ? 'video' : 'voice' }
+          : null,
+      }));
+    } catch (err) {
+      console.error('[CallStore] Failed to toggle video:', err);
+    }
+  },
+
+  // 重置所有状态，用于登出时清理
+  reset: () => {
+    // 清理所有定时器和资源
+    cleanupResources();
+
+    // 清理 initCall 的订阅
+    if (initCallUnsub) {
+      initCallUnsub();
+      initCallUnsub = null;
+    }
+
+    // 清理 invited 订阅
+    if (unsubInvited) {
+      unsubInvited();
+      unsubInvited = null;
+    }
+
+    // 清理 WebRTC 资源
+    webrtcManager.dispose();
+
+    // 清理已处理的通话ID缓存
+    processedCallIds.clear();
+
+    // 重置状态
+    set({
+      currentCall: null,
+      incomingCall: null,
+      callDuration: 0,
+      peerStream: null,
+      localStream: null,
+      isMinimized: false,
+    });
+  },
+}));