/**
 * Base media connection for Discord GoLive — ported from
 * @dank074/discord-video-stream BaseMediaConnection.js.
 *
 * Owns the voice WebSocket (identify/select_protocol/heartbeat/resume),
 * SDP negotiation against Discord's media server, DAVE E2E voice
 * (via @snazzah/davey), and speaking/video attribute signaling.
 */

import { randomUUID } from "node:crypto";
import { EventEmitter } from "node:events";
import Davey from "@snazzah/davey";
import { CodecPayloadType } from "./CodecPayloadType.js";
import type { NativePeerConnection } from "./native.js";
import { isNativeAvailable } from "./native.js";
import { STREAMS_SIMULCAST } from "./utils.js";
import { VoiceOpCodes, VoiceOpCodesBinary } from "./VoiceOpCodes.js";
import { WebRtcConnWrapper } from "./WebRtcWrapper.js";

export interface MediaConnectionStatus {
  hasSession: boolean;
  hasToken: boolean;
  started: boolean;
  resuming: boolean;
}

export interface VideoAttribute {
  fps: number;
  width: number;
  height: number;
}

export interface StreamerLike {
  opts: Record<string, unknown>;
}

export class BaseMediaConnection extends EventEmitter {
  interval: ReturnType<typeof setInterval> | null = null;
  guildId: string | null = null;
  channelId: string;
  botId: string;
  ws: WebSocket | null = null;
  status: MediaConnectionStatus;
  server: string | null = null; // websocket url
  token: string | null = null;
  session_id: string | null = null;
  protected _webRtcWrapper: WebRtcConnWrapper;
  _webRtcParams: {
    address: string;
    port: number;
    audioSsrc: number;
    videoSsrc: number;
    rtxSsrc: number;
    supportedEncryptionModes: string[];
  } | null = null;
  protected _closed = false;
  ready: ((conn: WebRtcConnWrapper) => void) | null;
  protected _streamer: StreamerLike;
  protected _sequenceNumber = -1;
  protected _daveSession: Davey.DAVESession | null = null;
  protected _connectedUsers = new Set<string>();
  protected _daveProtocolVersion = 0;
  protected _davePendingTransitions = new Map<number, number>();
  protected _daveDowngraded = false;

  constructor(
    streamer: StreamerLike,
    guildId: string | null,
    botId: string,
    channelId: string,
    callback: ((conn: WebRtcConnWrapper) => void) | null,
  ) {
    super();
    this._streamer = streamer;
    this.status = {
      hasSession: false,
      hasToken: false,
      started: false,
      resuming: false,
    };
    this.guildId = guildId;
    this.channelId = channelId;
    this.botId = botId;
    this.ready = callback;
    this._webRtcWrapper = new WebRtcConnWrapper(this);
  }

  get type(): "guild" | "call" {
    return this.guildId ? "guild" : "call";
  }

  get webRtcConn(): WebRtcConnWrapper {
    return this._webRtcWrapper;
  }

  get webRtcParams(): BaseMediaConnection["_webRtcParams"] {
    return this._webRtcParams;
  }

  get streamer(): StreamerLike {
    return this._streamer;
  }

  /** daveChannelId — overridden in VoiceConnection (channelId) and StreamConnection (serverId - 1n). */
  get daveChannelId(): string {
    throw new Error("daveChannelId not implemented");
  }

  stop(): void {
    this._closed = true;
    this._webRtcWrapper.close();
    this.ws?.close();
  }

  setSession(session_id: string): void {
    this.session_id = session_id;
    this.status.hasSession = true;
    this.start();
  }

  setTokens(server: string, token: string): void {
    this.token = token;
    this.server = server;
    this.status.hasToken = true;
    this.start();
  }

  start(): void {
    if (this.status.hasSession && this.status.hasToken) {
      if (this.status.started) return;
      this.status.started = true;
      this.ws = new WebSocket(`wss://${this.server}/?v=8`);
      this.ws.binaryType = "arraybuffer";
      this.ws.addEventListener("open", () => {
        if (this.status.resuming) {
          this.status.resuming = false;
          this.resume();
        } else {
          this.identify();
        }
      });
      this.ws.addEventListener("error", (err) => {
        console.error(err);
      });
      this.ws.addEventListener("close", (e) => {
        const wasStarted = this.status.started;
        this.interval && clearInterval(this.interval);
        this.status.started = false;
        const canResume = e.code === 4015 || e.code < 4000;
        if (canResume && wasStarted) {
          this.status.resuming = true;
          this.start();
        } else {
          this._closed = true;
          this._webRtcWrapper?.close();
        }
      });
      this.setupEvents();
    }
  }

  handleReady(d: {
    ip: string;
    port: number;
    ssrc: number;
    streams: { ssrc: number; rtx_ssrc: number }[];
    modes: string[];
  }): void {
    // we hardcoded STREAMS_SIMULCAST, which will always be array of 1
    const stream = d.streams[0];
    console.log(
      `[goLive:${this.constructor.name}] READY ssrc=${d.ssrc} ip=${d.ip} port=${d.port} streams=${JSON.stringify(d.streams)}`,
    );
    this._webRtcParams = {
      address: d.ip,
      port: d.port,
      audioSsrc: d.ssrc,
      videoSsrc: stream.ssrc,
      rtxSsrc: stream.rtx_ssrc,
      supportedEncryptionModes: d.modes,
    };
  }

  async handleProtocolAck(d: {
    sdp?: string;
    dave_protocol_version?: number;
  }): Promise<void> {
    if (!("sdp" in d)) throw new Error("Only WebRTC connections are allowed");
    // DEBUG: dump Discord's real answer SDP — which payload types did it select?
    console.log(
      `[goLive:${this.constructor.name}] DISCORD_ANSWER_SDP ${JSON.stringify(d.sdp ?? "").slice(0, 900)}`,
    );
    this._daveProtocolVersion = d.dave_protocol_version ?? 0;
    this.initDave();
    // Discord's SDP is garbage — generate our own from its pieces
    let ip = "";
    let port = "";
    let iceUsername = "";
    let icePassword = "";
    let fingerprint = "";
    let candidate = "";
    for (const line of (d.sdp ?? "").split("\n")) {
      if (line.startsWith("c=")) ip = line;
      else if (line.startsWith("a=rtcp")) port = line.split(":")[1];
      else if (line.startsWith("a=ice-ufrag")) iceUsername = line;
      else if (line.startsWith("a=ice-pwd")) icePassword = line;
      else if (line.startsWith("a=fingerprint")) fingerprint = line;
      else if (line.startsWith("a=candidate")) candidate = line;
    }
    const audioPayloadType = CodecPayloadType.opus.payload_type;
    const audioSection = `
m=audio ${port} UDP/TLS/RTP/SAVPF ${audioPayloadType}
${ip}
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=setup:passive
a=mid:0
a=maxptime:60
a=inactive
${iceUsername}
${icePassword}
${fingerprint}
${candidate}
a=rtcp-mux
a=rtpmap:${audioPayloadType} opus/48000/2
a=fmtp:${audioPayloadType} minptime=10;useinbandfec=1;usedtx=1
a=rtcp-fb:${audioPayloadType} transport-cc
a=rtcp-fb:${audioPayloadType} nack
a=ice-lite
`.trim();
    const videoPayloads = Object.values(CodecPayloadType).filter(
      (el) => el.type === "video",
    );
    const videoPayloadTypes = videoPayloads.flatMap((el) => [
      el.payload_type,
      el.rtx_payload_type ?? 0,
    ]);
    const videoSection = `
m=video ${port} UDP/TLS/RTP/SAVPF ${videoPayloadTypes.join(" ")}
${ip}
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:13 urn:3gpp:video-orientation
a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=setup:passive
a=mid:1
a=inactive
${iceUsername}
${icePassword}
${fingerprint}
${candidate}
a=rtcp-mux
a=ice-lite
`.trim();
    const videoRtpMap = videoPayloads
      .flatMap((el) => [
        `a=rtpmap:${el.payload_type} ${el.name}/90000`,
        `a=rtpmap:${el.rtx_payload_type} rtx/90000`,
        `a=fmtp:${el.rtx_payload_type} apt=${el.payload_type}`,
        `a=rtcp-fb:${el.payload_type} ccm fir`,
        `a=rtcp-fb:${el.payload_type} nack`,
        `a=rtcp-fb:${el.payload_type} nack pli`,
        `a=rtcp-fb:${el.payload_type} goog-remb`,
        `a=rtcp-fb:${el.payload_type} transport-cc`,
      ])
      .join("\n");
    const builtAnswer = [audioSection, videoSection, videoRtpMap].join("\n");
    this._webRtcWrapper.webRtcConn?.setRemoteDescription(builtAnswer, "answer");
    console.log(
      `[goLive:${this.constructor.name}] SELECT_PROTOCOL_ACK processed — remote answer set (${builtAnswer.length}B) video_mline=${videoPayloadTypes.join(" ")}`,
    );
    this.emit("select_protocol_ack");
  }

  initDave(): void {
    if (this._daveProtocolVersion) {
      if (this._daveSession) {
        this._daveSession.reinit(
          this._daveProtocolVersion,
          this.botId,
          this.daveChannelId,
        );
      } else {
        this._daveSession = new Davey.DAVESession(
          this._daveProtocolVersion,
          this.botId,
          this.daveChannelId,
        );
      }
      this.sendOpcodeBinary(
        VoiceOpCodesBinary.MLS_KEY_PACKAGE,
        this._daveSession.getSerializedKeyPackage(),
      );
    } else if (this._daveSession) {
      this._daveSession.reset();
      this._daveSession.setPassthroughMode(true, 10);
    }
  }

  processInvalidCommit(transitionId: number): void {
    this.sendOpcode(VoiceOpCodes.MLS_INVALID_COMMIT_WELCOME, {
      transition_id: transitionId,
    });
    this.initDave();
  }

  executePendingTransition(transitionId: number): void {
    const newVersion = this._davePendingTransitions.get(transitionId);
    if (newVersion === undefined) {
      console.error("Unrecognized transition ID", { transitionId });
      return;
    }
    const oldVersion = this._daveProtocolVersion;
    this._daveProtocolVersion = newVersion;
    if (oldVersion !== newVersion && newVersion === 0) {
      // Downgraded
      this._daveDowngraded = true;
    } else if (transitionId > 0 && this._daveDowngraded) {
      this._daveDowngraded = false;
      this._daveSession?.setPassthroughMode(true, 10);
    }
    this._davePendingTransitions.delete(transitionId);
  }

  setupEvents(): void {
    this.ws?.addEventListener("message", async (e) => {
      if (e.data instanceof ArrayBuffer) {
        this.handleBinaryMessages(Buffer.from(e.data));
        return;
      }
      const { op, d, seq } = JSON.parse(e.data as string) as {
        op: number;
        // biome-ignore lint/suspicious/noExplicitAny: Discord voice WS payload is dynamically typed
        d: any;
        seq?: number;
      };
      if (seq) this._sequenceNumber = seq;
      if (op === VoiceOpCodes.READY) {
        this.handleReady(d);
        this.setProtocols()
          .then(() => this.ready?.(this._webRtcWrapper))
          .catch((err: unknown) => {
            // PC can be closed while setProtocols is in flight (stream
            // teardown) — don't let that become an unhandledRejection.
            console.log(
              `[goLive:${this.constructor.name}] setProtocols rejected during teardown: ${err instanceof Error ? err.message : String(err)}`,
            );
          });
        this.setVideoAttributes(false);
      } else if (op >= 4000) {
        console.error(`${this.constructor.name} connection error`, d);
      } else if (op === VoiceOpCodes.HELLO) {
        this.setupHeartbeat(d.heartbeat_interval);
      } else if (op === VoiceOpCodes.SELECT_PROTOCOL_ACK) {
        await this.handleProtocolAck(d);
      } else if (op === VoiceOpCodes.SPEAKING) {
        // ignore speaking updates
      } else if (op === VoiceOpCodes.HEARTBEAT_ACK) {
        // ignore heartbeat acknowledgements
      } else if (op === VoiceOpCodes.RESUMED) {
        this.status.started = true;
      } else if (op === VoiceOpCodes.CLIENTS_CONNECT) {
        d.user_ids.forEach((id: string) => {
          this._connectedUsers.add(id);
        });
      } else if (op === VoiceOpCodes.CLIENT_DISCONNECT) {
        this._connectedUsers.delete(d.user_id);
      } else if (op === VoiceOpCodes.DAVE_PREPARE_TRANSITION) {
        this._davePendingTransitions.set(d.transition_id, d.protocol_version);
        if (d.transition_id === 0) {
          this.executePendingTransition(d.transition_id);
        } else {
          if (d.protocol_version === 0) {
            this._daveSession?.setPassthroughMode(true, 120);
          }
          this.sendOpcode(VoiceOpCodes.DAVE_TRANSITION_READY, {
            transition_id: d.transition_id,
          });
        }
      } else if (op === VoiceOpCodes.DAVE_EXECUTE_TRANSITION) {
        this.executePendingTransition(d.transition_id);
      } else if (op === VoiceOpCodes.DAVE_PREPARE_EPOCH) {
        if (d.epoch === 1) {
          this._daveProtocolVersion = d.protocol_version;
          this.initDave();
        }
      }
    });
  }

  handleBinaryMessages(msg: Buffer): void {
    this._sequenceNumber = msg.readUint16BE(0);
    const op = msg.readUint8(2);
    switch (op) {
      case VoiceOpCodesBinary.MLS_EXTERNAL_SENDER: {
        this._daveSession?.setExternalSender(msg.subarray(3));
        break;
      }
      case VoiceOpCodesBinary.MLS_PROPOSALS: {
        const optype = msg.readUint8(3);
        if (!this._daveSession) break;
        const { commit, welcome } = this._daveSession.processProposals(
          optype,
          msg.subarray(4),
          [...this._connectedUsers],
        );
        if (commit) {
          this.sendOpcodeBinary(
            VoiceOpCodesBinary.MLS_COMMIT_WELCOME,
            welcome ? Buffer.concat([commit, welcome]) : commit,
          );
        }
        break;
      }
      case VoiceOpCodesBinary.MLS_ANNOUNCE_COMMIT_TRANSITION: {
        const transitionId = msg.readUInt16BE(3);
        try {
          this._daveSession?.processCommit(msg.subarray(5));
          if (transitionId) {
            this._davePendingTransitions.set(
              transitionId,
              this._daveProtocolVersion,
            );
            this.sendOpcode(VoiceOpCodes.DAVE_TRANSITION_READY, {
              transition_id: transitionId,
            });
          }
        } catch (e) {
          console.debug("MLS commit errored", e);
          this.processInvalidCommit(transitionId);
        }
        break;
      }
      case VoiceOpCodesBinary.MLS_WELCOME: {
        const transitionId = msg.readUInt16BE(3);
        try {
          this._daveSession?.processWelcome(msg.subarray(5));
          if (transitionId) {
            this._davePendingTransitions.set(
              transitionId,
              this._daveProtocolVersion,
            );
            this.sendOpcode(VoiceOpCodes.DAVE_TRANSITION_READY, {
              transition_id: transitionId,
            });
          }
        } catch (e) {
          console.debug("MLS welcome errored", e);
          this.processInvalidCommit(transitionId);
        }
        break;
      }
    }
  }

  get daveReady(): boolean {
    return !!this._daveProtocolVersion && !!this._daveSession?.ready;
  }

  get daveSession(): Davey.DAVESession | null {
    return this._daveSession;
  }

  setupHeartbeat(interval: number): void {
    if (this.interval) {
      clearInterval(this.interval);
    }
    this.interval = setInterval(() => {
      try {
        this.sendOpcode(VoiceOpCodes.HEARTBEAT, {
          t: Date.now(),
          seq_ack: this._sequenceNumber,
        });
      } catch {
        /* ignore */
      }
    }, interval);
  }

  sendOpcode(code: number, data: unknown): void {
    if (this.ws?.readyState !== WebSocket.OPEN) return;
    this.ws.send(JSON.stringify({ op: code, d: data }));
  }

  sendOpcodeBinary(code: number, data: Uint8Array): void {
    if (this.ws?.readyState !== WebSocket.OPEN) return;
    const buf = Buffer.allocUnsafe(data.length + 1);
    buf.writeUInt8(code);
    Buffer.from(data).copy(buf, 1);
    this.ws.send(buf);
  }

  /** serverId — overridden in VoiceConnection (guildId ?? channelId) and StreamConnection (rtc_server_id). */
  get serverId(): string | null {
    throw new Error("serverId not implemented");
  }

  /** identifies with media server with credentials */
  identify(): void {
    if (!this.serverId) throw new Error("Server ID is null or empty");
    if (!this.session_id) throw new Error("Session ID is null or empty");
    if (!this.token) throw new Error("Token is null or empty");
    this.sendOpcode(VoiceOpCodes.IDENTIFY, {
      server_id: this.serverId,
      user_id: this.botId,
      session_id: this.session_id,
      token: this.token,
      video: true,
      streams: STREAMS_SIMULCAST,
      max_dave_protocol_version: Davey.DAVE_PROTOCOL_VERSION ?? 0,
    });
  }

  resume(): void {
    if (!this.serverId) throw new Error("Server ID is null or empty");
    if (!this.session_id) throw new Error("Session ID is null or empty");
    if (!this.token) throw new Error("Token is null or empty");
    this.sendOpcode(VoiceOpCodes.RESUME, {
      server_id: this.serverId,
      session_id: this.session_id,
      token: this.token,
      seq_ack: this._sequenceNumber,
    });
  }

  /** Sets protocols and ip data used for video and audio (vp8 video, opus audio). */
  async setProtocols(): Promise<void> {
    if (!this._webRtcParams) throw new Error("WebRTC parameters not set");
    if (!isNativeAvailable()) {
      throw new Error(
        "libdatachannel-min native binding not built — cannot start GoLive",
      );
    }
    const reconnect = () => {
      const webRtcConn = this._webRtcWrapper.initWebRtc();
      webRtcConn.onStateChange((state) => {
        console.log(`[goLive:${this.constructor.name}] pc state => ${state}`);
        if (state === "closed" && !this._closed) reconnect();
      });
      this._webRtcWrapper.onLocalDescription = (sdp) => {
        const rtc_connection_id = randomUUID();
        console.log(
          `[goLive:${this.constructor.name}] sending SELECT_PROTOCOL (offer ${sdp.length}B, rtc_connection_id=${rtc_connection_id.slice(0, 8)})`,
        );
        this.sendOpcode(VoiceOpCodes.SELECT_PROTOCOL, {
          protocol: "webrtc",
          codecs: Object.values(CodecPayloadType),
          data: sdp,
          sdp,
          rtc_connection_id,
        });
      };
      // createOffer (binding resolves full SDP incl. candidates after gathering)
      void webRtcConn
        .createOffer()
        .then((sdp) => {
          this._webRtcWrapper.onLocalDescription?.(sdp);
        })
        .catch((err: unknown) => {
          // PC closed while offer is gathering (stream teardown / reconnect) —
          // swallow, the reconnect loop will start a fresh offer.
          console.log(
            `[goLive:${this.constructor.name}] createOffer rejected: ${err instanceof Error ? err.message : String(err)}`,
          );
        });
    };
    reconnect();
    return new Promise((resolve) => {
      this.once("select_protocol_ack", () => resolve());
    });
  }

  setVideoAttributes(enabled: boolean, attr?: VideoAttribute): void {
    if (!this._webRtcParams) throw new Error("WebRTC parameters not set");
    const { audioSsrc, videoSsrc, rtxSsrc } = this._webRtcParams;
    if (!enabled) {
      this.sendOpcode(VoiceOpCodes.VIDEO, {
        audio_ssrc: audioSsrc,
        video_ssrc: 0,
        rtx_ssrc: 0,
        streams: [],
      });
    } else {
      if (!attr) throw new Error("Need to specify video attributes");
      this.sendOpcode(VoiceOpCodes.VIDEO, {
        audio_ssrc: audioSsrc,
        video_ssrc: videoSsrc,
        rtx_ssrc: rtxSsrc,
        streams: [
          {
            type: "video",
            rid: "100",
            ssrc: videoSsrc,
            active: true,
            quality: 100,
            rtx_ssrc: rtxSsrc,
            // hardcode the max bitrate because we don't really know anyway
            max_bitrate: 10000 * 1000,
            max_framerate: enabled ? attr.fps : 0,
            max_resolution: {
              type: "fixed",
              width: attr.width,
              height: attr.height,
            },
          },
        ],
      });
    }
  }

  /** Set speaking status */
  setSpeaking(speaking: boolean): void {
    if (!this._webRtcParams) throw new Error("WebRTC connection not ready");
    this.sendOpcode(VoiceOpCodes.SPEAKING, {
      delay: 0,
      speaking: speaking ? 1 : 0,
      ssrc: this._webRtcParams.audioSsrc,
    });
  }
}

export type { NativePeerConnection };