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 {
  // Create a new MediaStream instance to force zustand subscribers to update
  // because react-native-webrtc may add tracks to the same stream object reference
  const newStream = new MediaStream();
  stream.getTracks().forEach((track) => {
    newStream.addTrack(track);
  });

  callStore.setState({ peerStream: newStream });

  // Detect video tracks in remote stream
  const videoTracks = newStream.getVideoTracks();
  const hasPeerVideo = videoTracks.length > 0 && videoTracks.some((t) => t.enabled);

  console.log('[CallStore] Remote stream received, hasVideo:', hasPeerVideo, 'videoTracks:', videoTracks.length, 'audioTracks:', newStream.getAudioTracks().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,
    });
  },
}));