"use strict";

var _interopRequireDefault = require("@babel/runtime/helpers/interopRequireDefault");
Object.defineProperty(exports, "__esModule", {
  value: true
});
exports.MatrixCall = exports.CallType = exports.CallState = exports.CallParty = exports.CallEvent = exports.CallErrorCode = exports.CallError = exports.CallDirection = void 0;
exports.createNewMatrixCall = createNewMatrixCall;
exports.genCallID = genCallID;
exports.setTracksEnabled = setTracksEnabled;
exports.supportsMatrixCall = supportsMatrixCall;
var _defineProperty2 = _interopRequireDefault(require("@babel/runtime/helpers/defineProperty"));
var _uuid = require("uuid");
var _sdpTransform = require("sdp-transform");
var _logger = require("../logger");
var utils = _interopRequireWildcard(require("../utils"));
var _event = require("../@types/event");
var _randomstring = require("../randomstring");
var _callEventTypes = require("./callEventTypes");
var _callFeed = require("./callFeed");
var _typedEventEmitter = require("../models/typed-event-emitter");
var _deviceinfo = require("../crypto/deviceinfo");
var _groupCall = require("./groupCall");
var _httpApi = require("../http-api");
function _getRequireWildcardCache(nodeInterop) { if (typeof WeakMap !== "function") return null; var cacheBabelInterop = new WeakMap(); var cacheNodeInterop = new WeakMap(); return (_getRequireWildcardCache = function (nodeInterop) { return nodeInterop ? cacheNodeInterop : cacheBabelInterop; })(nodeInterop); }
function _interopRequireWildcard(obj, nodeInterop) { if (!nodeInterop && obj && obj.__esModule) { return obj; } if (obj === null || typeof obj !== "object" && typeof obj !== "function") { return { default: obj }; } var cache = _getRequireWildcardCache(nodeInterop); if (cache && cache.has(obj)) { return cache.get(obj); } var newObj = {}; var hasPropertyDescriptor = Object.defineProperty && Object.getOwnPropertyDescriptor; for (var key in obj) { if (key !== "default" && Object.prototype.hasOwnProperty.call(obj, key)) { var desc = hasPropertyDescriptor ? Object.getOwnPropertyDescriptor(obj, key) : null; if (desc && (desc.get || desc.set)) { Object.defineProperty(newObj, key, desc); } else { newObj[key] = obj[key]; } } } newObj.default = obj; if (cache) { cache.set(obj, newObj); } return newObj; }
function ownKeys(object, enumerableOnly) { var keys = Object.keys(object); if (Object.getOwnPropertySymbols) { var symbols = Object.getOwnPropertySymbols(object); enumerableOnly && (symbols = symbols.filter(function (sym) { return Object.getOwnPropertyDescriptor(object, sym).enumerable; })), keys.push.apply(keys, symbols); } return keys; }
function _objectSpread(target) { for (var i = 1; i < arguments.length; i++) { var source = null != arguments[i] ? arguments[i] : {}; i % 2 ? ownKeys(Object(source), !0).forEach(function (key) { (0, _defineProperty2.default)(target, key, source[key]); }) : Object.getOwnPropertyDescriptors ? Object.defineProperties(target, Object.getOwnPropertyDescriptors(source)) : ownKeys(Object(source)).forEach(function (key) { Object.defineProperty(target, key, Object.getOwnPropertyDescriptor(source, key)); }); } return target; }
var MediaType;
(function (MediaType) {
  MediaType["AUDIO"] = "audio";
  MediaType["VIDEO"] = "video";
})(MediaType || (MediaType = {}));
var CodecName; // add more as needed
// Used internally to specify modifications to codec parameters in SDP
(function (CodecName) {
  CodecName["OPUS"] = "opus";
})(CodecName || (CodecName = {}));
let CallState;
exports.CallState = CallState;
(function (CallState) {
  CallState["Fledgling"] = "fledgling";
  CallState["InviteSent"] = "invite_sent";
  CallState["WaitLocalMedia"] = "wait_local_media";
  CallState["CreateOffer"] = "create_offer";
  CallState["CreateAnswer"] = "create_answer";
  CallState["Connecting"] = "connecting";
  CallState["Connected"] = "connected";
  CallState["Ringing"] = "ringing";
  CallState["Ended"] = "ended";
})(CallState || (exports.CallState = CallState = {}));
let CallType;
exports.CallType = CallType;
(function (CallType) {
  CallType["Voice"] = "voice";
  CallType["Video"] = "video";
})(CallType || (exports.CallType = CallType = {}));
let CallDirection;
exports.CallDirection = CallDirection;
(function (CallDirection) {
  CallDirection["Inbound"] = "inbound";
  CallDirection["Outbound"] = "outbound";
})(CallDirection || (exports.CallDirection = CallDirection = {}));
let CallParty;
exports.CallParty = CallParty;
(function (CallParty) {
  CallParty["Local"] = "local";
  CallParty["Remote"] = "remote";
})(CallParty || (exports.CallParty = CallParty = {}));
let CallEvent;
exports.CallEvent = CallEvent;
(function (CallEvent) {
  CallEvent["Hangup"] = "hangup";
  CallEvent["State"] = "state";
  CallEvent["Error"] = "error";
  CallEvent["Replaced"] = "replaced";
  CallEvent["LocalHoldUnhold"] = "local_hold_unhold";
  CallEvent["RemoteHoldUnhold"] = "remote_hold_unhold";
  CallEvent["HoldUnhold"] = "hold_unhold";
  CallEvent["FeedsChanged"] = "feeds_changed";
  CallEvent["AssertedIdentityChanged"] = "asserted_identity_changed";
  CallEvent["LengthChanged"] = "length_changed";
  CallEvent["DataChannel"] = "datachannel";
  CallEvent["SendVoipEvent"] = "send_voip_event";
})(CallEvent || (exports.CallEvent = CallEvent = {}));
let CallErrorCode;
/**
 * The version field that we set in m.call.* events
 */
exports.CallErrorCode = CallErrorCode;
(function (CallErrorCode) {
  CallErrorCode["UserHangup"] = "user_hangup";
  CallErrorCode["LocalOfferFailed"] = "local_offer_failed";
  CallErrorCode["NoUserMedia"] = "no_user_media";
  CallErrorCode["UnknownDevices"] = "unknown_devices";
  CallErrorCode["SendInvite"] = "send_invite";
  CallErrorCode["CreateAnswer"] = "create_answer";
  CallErrorCode["CreateOffer"] = "create_offer";
  CallErrorCode["SendAnswer"] = "send_answer";
  CallErrorCode["SetRemoteDescription"] = "set_remote_description";
  CallErrorCode["SetLocalDescription"] = "set_local_description";
  CallErrorCode["AnsweredElsewhere"] = "answered_elsewhere";
  CallErrorCode["IceFailed"] = "ice_failed";
  CallErrorCode["InviteTimeout"] = "invite_timeout";
  CallErrorCode["Replaced"] = "replaced";
  CallErrorCode["SignallingFailed"] = "signalling_timeout";
  CallErrorCode["UserBusy"] = "user_busy";
  CallErrorCode["Transferred"] = "transferred";
  CallErrorCode["NewSession"] = "new_session";
})(CallErrorCode || (exports.CallErrorCode = CallErrorCode = {}));
const VOIP_PROTO_VERSION = "1";

/** The fallback ICE server to use for STUN or TURN protocols. */
const FALLBACK_ICE_SERVER = "stun:turn.matrix.org";

/** The length of time a call can be ringing for. */
const CALL_TIMEOUT_MS = 60 * 1000; // ms
/** The time after which we increment callLength */
const CALL_LENGTH_INTERVAL = 1000; // ms
/** The time after which we end the call, if ICE got disconnected */
const ICE_DISCONNECTED_TIMEOUT = 30 * 1000; // ms

class CallError extends Error {
  constructor(code, msg, err) {
    // Still don't think there's any way to have proper nested errors
    super(msg + ": " + err);
    (0, _defineProperty2.default)(this, "code", void 0);
    this.code = code;
  }
}
exports.CallError = CallError;
function genCallID() {
  return Date.now().toString() + (0, _randomstring.randomString)(16);
}
function getCodecParamMods(isPtt) {
  const mods = [{
    mediaType: "audio",
    codec: "opus",
    enableDtx: true,
    maxAverageBitrate: isPtt ? 12000 : undefined
  }];
  return mods;
}
// generates keys for the map of transceivers
// kind is unfortunately a string rather than MediaType as this is the type of
// track.kind
function getTransceiverKey(purpose, kind) {
  return purpose + ":" + kind;
}
class MatrixCall extends _typedEventEmitter.TypedEventEmitter {
  // whether this call should have push-to-talk semantics
  // This should be set by the consumer on incoming & outgoing calls.

  // A queue for candidates waiting to go out.
  // We try to amalgamate candidates into a single candidate message where
  // possible

  // our transceivers for each purpose and type of media

  // The party ID of the other side: undefined if we haven't chosen a partner
  // yet, null if we have but they didn't send a party ID.

  // The logic of when & if a call is on hold is nontrivial and explained in is*OnHold
  // This flag represents whether we want the other party to be on hold

  // the stats for the call at the point it ended. We can't get these after we
  // tear the call down, so we just grab a snapshot before we stop the call.
  // The typescript definitions have this type as 'any' :(

  // Perfect negotiation state: https://www.w3.org/TR/webrtc/#perfect-negotiation-example

  // If candidates arrive before we've picked an opponent (which, in particular,
  // will happen if the opponent sends candidates eagerly before the user answers
  // the call) we buffer them up here so we can then add the ones from the party we pick

  // Used to keep the timer for the delay before actually stopping our
  // video track after muting (see setLocalVideoMuted)

  // Used to allow connection without Video and Audio. To establish a webrtc connection without media a Data channel is
  // needed At the moment this property is true if we allow MatrixClient with isVoipWithNoMediaAllowed = true

  /**
   * Construct a new Matrix Call.
   * @param opts - Config options.
   */
  constructor(opts) {
    var _opts$forceTURN;
    super();
    (0, _defineProperty2.default)(this, "roomId", void 0);
    (0, _defineProperty2.default)(this, "callId", void 0);
    (0, _defineProperty2.default)(this, "invitee", void 0);
    (0, _defineProperty2.default)(this, "hangupParty", void 0);
    (0, _defineProperty2.default)(this, "hangupReason", void 0);
    (0, _defineProperty2.default)(this, "direction", void 0);
    (0, _defineProperty2.default)(this, "ourPartyId", void 0);
    (0, _defineProperty2.default)(this, "peerConn", void 0);
    (0, _defineProperty2.default)(this, "toDeviceSeq", 0);
    (0, _defineProperty2.default)(this, "isPtt", false);
    (0, _defineProperty2.default)(this, "_state", CallState.Fledgling);
    (0, _defineProperty2.default)(this, "client", void 0);
    (0, _defineProperty2.default)(this, "forceTURN", void 0);
    (0, _defineProperty2.default)(this, "turnServers", void 0);
    (0, _defineProperty2.default)(this, "candidateSendQueue", []);
    (0, _defineProperty2.default)(this, "candidateSendTries", 0);
    (0, _defineProperty2.default)(this, "candidatesEnded", false);
    (0, _defineProperty2.default)(this, "feeds", []);
    (0, _defineProperty2.default)(this, "transceivers", new Map());
    (0, _defineProperty2.default)(this, "inviteOrAnswerSent", false);
    (0, _defineProperty2.default)(this, "waitForLocalAVStream", false);
    (0, _defineProperty2.default)(this, "successor", void 0);
    (0, _defineProperty2.default)(this, "opponentMember", void 0);
    (0, _defineProperty2.default)(this, "opponentVersion", void 0);
    (0, _defineProperty2.default)(this, "opponentPartyId", void 0);
    (0, _defineProperty2.default)(this, "opponentCaps", void 0);
    (0, _defineProperty2.default)(this, "iceDisconnectedTimeout", void 0);
    (0, _defineProperty2.default)(this, "inviteTimeout", void 0);
    (0, _defineProperty2.default)(this, "removeTrackListeners", new Map());
    (0, _defineProperty2.default)(this, "remoteOnHold", false);
    (0, _defineProperty2.default)(this, "callStatsAtEnd", void 0);
    (0, _defineProperty2.default)(this, "makingOffer", false);
    (0, _defineProperty2.default)(this, "ignoreOffer", false);
    (0, _defineProperty2.default)(this, "responsePromiseChain", void 0);
    (0, _defineProperty2.default)(this, "remoteCandidateBuffer", new Map());
    (0, _defineProperty2.default)(this, "remoteAssertedIdentity", void 0);
    (0, _defineProperty2.default)(this, "remoteSDPStreamMetadata", void 0);
    (0, _defineProperty2.default)(this, "callLengthInterval", void 0);
    (0, _defineProperty2.default)(this, "callStartTime", void 0);
    (0, _defineProperty2.default)(this, "opponentDeviceId", void 0);
    (0, _defineProperty2.default)(this, "opponentDeviceInfo", void 0);
    (0, _defineProperty2.default)(this, "opponentSessionId", void 0);
    (0, _defineProperty2.default)(this, "groupCallId", void 0);
    (0, _defineProperty2.default)(this, "stopVideoTrackTimer", void 0);
    (0, _defineProperty2.default)(this, "isOnlyDataChannelAllowed", void 0);
    (0, _defineProperty2.default)(this, "stats", void 0);
    (0, _defineProperty2.default)(this, "gotLocalIceCandidate", event => {
      if (event.candidate) {
        if (this.candidatesEnded) {
          _logger.logger.warn(`Call ${this.callId} gotLocalIceCandidate() got candidate after candidates have ended - ignoring!`);
          return;
        }
        _logger.logger.debug(`Call ${this.callId} got local ICE ${event.candidate.sdpMid} ${event.candidate.candidate}`);
        if (this.callHasEnded()) return;

        // As with the offer, note we need to make a copy of this object, not
        // pass the original: that broke in Chrome ~m43.
        if (event.candidate.candidate === "") {
          this.queueCandidate(null);
        } else {
          this.queueCandidate(event.candidate);
        }
      }
    });
    (0, _defineProperty2.default)(this, "onIceGatheringStateChange", event => {
      var _this$peerConn;
      _logger.logger.debug(`Call ${this.callId} onIceGatheringStateChange() ice gathering state changed to ${this.peerConn.iceGatheringState}`);
      if (((_this$peerConn = this.peerConn) === null || _this$peerConn === void 0 ? void 0 : _this$peerConn.iceGatheringState) === "complete") {
        this.queueCandidate(null);
      }
    });
    (0, _defineProperty2.default)(this, "getLocalOfferFailed", err => {
      _logger.logger.error(`Call ${this.callId} getLocalOfferFailed() running`, err);
      this.emit(CallEvent.Error, new CallError(CallErrorCode.LocalOfferFailed, "Failed to get local offer!", err), this);
      this.terminate(CallParty.Local, CallErrorCode.LocalOfferFailed, false);
    });
    (0, _defineProperty2.default)(this, "getUserMediaFailed", err => {
      if (this.successor) {
        this.successor.getUserMediaFailed(err);
        return;
      }
      _logger.logger.warn(`Call ${this.callId} getUserMediaFailed() failed to get user media - ending call`, err);
      this.emit(CallEvent.Error, new CallError(CallErrorCode.NoUserMedia, "Couldn't start capturing media! Is your microphone set up and " + "does this app have permission?", err), this);
      this.terminate(CallParty.Local, CallErrorCode.NoUserMedia, false);
    });
    (0, _defineProperty2.default)(this, "onIceConnectionStateChanged", () => {
      var _this$peerConn2, _this$peerConn$iceCon, _this$peerConn3, _this$peerConn4, _this$peerConn6;
      if (this.callHasEnded()) {
        return; // because ICE can still complete as we're ending the call
      }

      _logger.logger.debug(`Call ${this.callId} onIceConnectionStateChanged() running (state=${(_this$peerConn2 = this.peerConn) === null || _this$peerConn2 === void 0 ? void 0 : _this$peerConn2.iceConnectionState})`);

      // ideally we'd consider the call to be connected when we get media but
      // chrome doesn't implement any of the 'onstarted' events yet
      if (["connected", "completed"].includes((_this$peerConn$iceCon = (_this$peerConn3 = this.peerConn) === null || _this$peerConn3 === void 0 ? void 0 : _this$peerConn3.iceConnectionState) !== null && _this$peerConn$iceCon !== void 0 ? _this$peerConn$iceCon : "")) {
        clearTimeout(this.iceDisconnectedTimeout);
        this.iceDisconnectedTimeout = undefined;
        this.state = CallState.Connected;
        if (!this.callLengthInterval && !this.callStartTime) {
          this.callStartTime = Date.now();
          this.callLengthInterval = setInterval(() => {
            this.emit(CallEvent.LengthChanged, Math.round((Date.now() - this.callStartTime) / 1000), this);
          }, CALL_LENGTH_INTERVAL);
        }
      } else if (((_this$peerConn4 = this.peerConn) === null || _this$peerConn4 === void 0 ? void 0 : _this$peerConn4.iceConnectionState) == "failed") {
        var _this$peerConn5;
        // Firefox for Android does not yet have support for restartIce()
        // (the types say it's always defined though, so we have to cast
        // to prevent typescript from warning).
        if ((_this$peerConn5 = this.peerConn) !== null && _this$peerConn5 !== void 0 && _this$peerConn5.restartIce) {
          this.candidatesEnded = false;
          this.peerConn.restartIce();
        } else {
          _logger.logger.info(`Call ${this.callId} onIceConnectionStateChanged() hanging up call (ICE failed and no ICE restart method)`);
          this.hangup(CallErrorCode.IceFailed, false);
        }
      } else if (((_this$peerConn6 = this.peerConn) === null || _this$peerConn6 === void 0 ? void 0 : _this$peerConn6.iceConnectionState) == "disconnected") {
        this.iceDisconnectedTimeout = setTimeout(() => {
          _logger.logger.info(`Call ${this.callId} onIceConnectionStateChanged() hanging up call (ICE disconnected for too long)`);
          this.hangup(CallErrorCode.IceFailed, false);
        }, ICE_DISCONNECTED_TIMEOUT);
        this.state = CallState.Connecting;
      }

      // In PTT mode, override feed status to muted when we lose connection to
      // the peer, since we don't want to block the line if they're not saying anything.
      // Experimenting in Chrome, this happens after 5 or 6 seconds, which is probably
      // fast enough.
      if (this.isPtt && ["failed", "disconnected"].includes(this.peerConn.iceConnectionState)) {
        for (const feed of this.getRemoteFeeds()) {
          feed.setAudioVideoMuted(true, true);
        }
      }
    });
    (0, _defineProperty2.default)(this, "onSignallingStateChanged", () => {
      var _this$peerConn7;
      _logger.logger.debug(`Call ${this.callId} onSignallingStateChanged() running (state=${(_this$peerConn7 = this.peerConn) === null || _this$peerConn7 === void 0 ? void 0 : _this$peerConn7.signalingState})`);
    });
    (0, _defineProperty2.default)(this, "onTrack", ev => {
      if (ev.streams.length === 0) {
        _logger.logger.warn(`Call ${this.callId} onTrack() called with streamless track streamless (kind=${ev.track.kind})`);
        return;
      }
      const stream = ev.streams[0];
      this.pushRemoteFeed(stream);
      if (!this.removeTrackListeners.has(stream)) {
        const onRemoveTrack = () => {
          if (stream.getTracks().length === 0) {
            _logger.logger.info(`Call ${this.callId} onTrack() removing track (streamId=${stream.id})`);
            this.deleteFeedByStream(stream);
            stream.removeEventListener("removetrack", onRemoveTrack);
            this.removeTrackListeners.delete(stream);
          }
        };
        stream.addEventListener("removetrack", onRemoveTrack);
        this.removeTrackListeners.set(stream, onRemoveTrack);
      }
    });
    (0, _defineProperty2.default)(this, "onDataChannel", ev => {
      this.emit(CallEvent.DataChannel, ev.channel, this);
    });
    (0, _defineProperty2.default)(this, "onNegotiationNeeded", async () => {
      _logger.logger.info(`Call ${this.callId} onNegotiationNeeded() negotiation is needed!`);
      if (this.state !== CallState.CreateOffer && this.opponentVersion === 0) {
        _logger.logger.info(`Call ${this.callId} onNegotiationNeeded() opponent does not support renegotiation: ignoring negotiationneeded event`);
        return;
      }
      this.queueGotLocalOffer();
    });
    (0, _defineProperty2.default)(this, "onHangupReceived", msg => {
      _logger.logger.debug(`Call ${this.callId} onHangupReceived() running`);

      // party ID must match (our chosen partner hanging up the call) or be undefined (we haven't chosen
      // a partner yet but we're treating the hangup as a reject as per VoIP v0)
      if (this.partyIdMatches(msg) || this.state === CallState.Ringing) {
        // default reason is user_hangup
        this.terminate(CallParty.Remote, msg.reason || CallErrorCode.UserHangup, true);
      } else {
        _logger.logger.info(`Call ${this.callId} onHangupReceived() ignoring message from party ID ${msg.party_id}: our partner is ${this.opponentPartyId}`);
      }
    });
    (0, _defineProperty2.default)(this, "onRejectReceived", msg => {
      _logger.logger.debug(`Call ${this.callId} onRejectReceived() running`);

      // No need to check party_id for reject because if we'd received either
      // an answer or reject, we wouldn't be in state InviteSent

      const shouldTerminate =
      // reject events also end the call if it's ringing: it's another of
      // our devices rejecting the call.
      [CallState.InviteSent, CallState.Ringing].includes(this.state) ||
      // also if we're in the init state and it's an inbound call, since
      // this means we just haven't entered the ringing state yet
      this.state === CallState.Fledgling && this.direction === CallDirection.Inbound;
      if (shouldTerminate) {
        this.terminate(CallParty.Remote, msg.reason || CallErrorCode.UserHangup, true);
      } else {
        _logger.logger.debug(`Call ${this.callId} onRejectReceived() called in wrong state (state=${this.state})`);
      }
    });
    (0, _defineProperty2.default)(this, "onAnsweredElsewhere", msg => {
      _logger.logger.debug(`Call ${this.callId} onAnsweredElsewhere() running`);
      this.terminate(CallParty.Remote, CallErrorCode.AnsweredElsewhere, true);
    });
    this.roomId = opts.roomId;
    this.invitee = opts.invitee;
    this.client = opts.client;
    if (!this.client.deviceId) throw new Error("Client must have a device ID to start calls");
    this.forceTURN = (_opts$forceTURN = opts.forceTURN) !== null && _opts$forceTURN !== void 0 ? _opts$forceTURN : false;
    this.ourPartyId = this.client.deviceId;
    this.opponentDeviceId = opts.opponentDeviceId;
    this.opponentSessionId = opts.opponentSessionId;
    this.groupCallId = opts.groupCallId;
    // Array of Objects with urls, username, credential keys
    this.turnServers = opts.turnServers || [];
    if (this.turnServers.length === 0 && this.client.isFallbackICEServerAllowed()) {
      this.turnServers.push({
        urls: [FALLBACK_ICE_SERVER]
      });
    }
    for (const server of this.turnServers) {
      utils.checkObjectHasKeys(server, ["urls"]);
    }
    this.callId = genCallID();
    // If the Client provides calls without audio and video we need a datachannel for a webrtc connection
    this.isOnlyDataChannelAllowed = this.client.isVoipWithNoMediaAllowed;
  }

  /**
   * Place a voice call to this room.
   * @throws If you have not specified a listener for 'error' events.
   */
  async placeVoiceCall() {
    await this.placeCall(true, false);
  }

  /**
   * Place a video call to this room.
   * @throws If you have not specified a listener for 'error' events.
   */
  async placeVideoCall() {
    await this.placeCall(true, true);
  }

  /**
   * Create a datachannel using this call's peer connection.
   * @param label - A human readable label for this datachannel
   * @param options - An object providing configuration options for the data channel.
   */
  createDataChannel(label, options) {
    const dataChannel = this.peerConn.createDataChannel(label, options);
    this.emit(CallEvent.DataChannel, dataChannel, this);
    return dataChannel;
  }
  getOpponentMember() {
    return this.opponentMember;
  }
  getOpponentDeviceId() {
    return this.opponentDeviceId;
  }
  getOpponentSessionId() {
    return this.opponentSessionId;
  }
  opponentCanBeTransferred() {
    return Boolean(this.opponentCaps && this.opponentCaps["m.call.transferee"]);
  }
  opponentSupportsDTMF() {
    return Boolean(this.opponentCaps && this.opponentCaps["m.call.dtmf"]);
  }
  getRemoteAssertedIdentity() {
    return this.remoteAssertedIdentity;
  }
  get state() {
    return this._state;
  }
  set state(state) {
    const oldState = this._state;
    this._state = state;
    this.emit(CallEvent.State, state, oldState, this);
  }
  get type() {
    // we may want to look for a video receiver here rather than a track to match the
    // sender behaviour, although in practice they should be the same thing
    return this.hasUserMediaVideoSender || this.hasRemoteUserMediaVideoTrack ? CallType.Video : CallType.Voice;
  }
  get hasLocalUserMediaVideoTrack() {
    var _this$localUsermediaS;
    return !!((_this$localUsermediaS = this.localUsermediaStream) !== null && _this$localUsermediaS !== void 0 && _this$localUsermediaS.getVideoTracks().length);
  }
  get hasRemoteUserMediaVideoTrack() {
    return this.getRemoteFeeds().some(feed => {
      var _feed$stream;
      return feed.purpose === _callEventTypes.SDPStreamMetadataPurpose.Usermedia && ((_feed$stream = feed.stream) === null || _feed$stream === void 0 ? void 0 : _feed$stream.getVideoTracks().length);
    });
  }
  get hasLocalUserMediaAudioTrack() {
    var _this$localUsermediaS2;
    return !!((_this$localUsermediaS2 = this.localUsermediaStream) !== null && _this$localUsermediaS2 !== void 0 && _this$localUsermediaS2.getAudioTracks().length);
  }
  get hasRemoteUserMediaAudioTrack() {
    return this.getRemoteFeeds().some(feed => {
      var _feed$stream2;
      return feed.purpose === _callEventTypes.SDPStreamMetadataPurpose.Usermedia && !!((_feed$stream2 = feed.stream) !== null && _feed$stream2 !== void 0 && _feed$stream2.getAudioTracks().length);
    });
  }
  get hasUserMediaAudioSender() {
    var _this$transceivers$ge;
    return Boolean((_this$transceivers$ge = this.transceivers.get(getTransceiverKey(_callEventTypes.SDPStreamMetadataPurpose.Usermedia, "audio"))) === null || _this$transceivers$ge === void 0 ? void 0 : _this$transceivers$ge.sender);
  }
  get hasUserMediaVideoSender() {
    var _this$transceivers$ge2;
    return Boolean((_this$transceivers$ge2 = this.transceivers.get(getTransceiverKey(_callEventTypes.SDPStreamMetadataPurpose.Usermedia, "video"))) === null || _this$transceivers$ge2 === void 0 ? void 0 : _this$transceivers$ge2.sender);
  }
  get localUsermediaFeed() {
    return this.getLocalFeeds().find(feed => feed.purpose === _callEventTypes.SDPStreamMetadataPurpose.Usermedia);
  }
  get localScreensharingFeed() {
    return this.getLocalFeeds().find(feed => feed.purpose === _callEventTypes.SDPStreamMetadataPurpose.Screenshare);
  }
  get localUsermediaStream() {
    var _this$localUsermediaF;
    return (_this$localUsermediaF = this.localUsermediaFeed) === null || _this$localUsermediaF === void 0 ? void 0 : _this$localUsermediaF.stream;
  }
  get localScreensharingStream() {
    var _this$localScreenshar;
    return (_this$localScreenshar = this.localScreensharingFeed) === null || _this$localScreenshar === void 0 ? void 0 : _this$localScreenshar.stream;
  }
  get remoteUsermediaFeed() {
    return this.getRemoteFeeds().find(feed => feed.purpose === _callEventTypes.SDPStreamMetadataPurpose.Usermedia);
  }
  get remoteScreensharingFeed() {
    return this.getRemoteFeeds().find(feed => feed.purpose === _callEventTypes.SDPStreamMetadataPurpose.Screenshare);
  }
  get remoteUsermediaStream() {
    var _this$remoteUsermedia;
    return (_this$remoteUsermedia = this.remoteUsermediaFeed) === null || _this$remoteUsermedia === void 0 ? void 0 : _this$remoteUsermedia.stream;
  }
  get remoteScreensharingStream() {
    var _this$remoteScreensha;
    return (_this$remoteScreensha = this.remoteScreensharingFeed) === null || _this$remoteScreensha === void 0 ? void 0 : _this$remoteScreensha.stream;
  }
  getFeedByStreamId(streamId) {
    return this.getFeeds().find(feed => feed.stream.id === streamId);
  }

  /**
   * Returns an array of all CallFeeds
   * @returns CallFeeds
   */
  getFeeds() {
    return this.feeds;
  }

  /**
   * Returns an array of all local CallFeeds
   * @returns local CallFeeds
   */
  getLocalFeeds() {
    return this.feeds.filter(feed => feed.isLocal());
  }

  /**
   * Returns an array of all remote CallFeeds
   * @returns remote CallFeeds
   */
  getRemoteFeeds() {
    return this.feeds.filter(feed => !feed.isLocal());
  }
  async initOpponentCrypto() {
    var _this$getOpponentMemb, _deviceInfoMap$get;
    if (!this.opponentDeviceId) return;
    if (!this.client.getUseE2eForGroupCall()) return;
    // It's possible to want E2EE and yet not have the means to manage E2EE
    // ourselves (for example if the client is a RoomWidgetClient)
    if (!this.client.isCryptoEnabled()) {
      // All we know is the device ID
      this.opponentDeviceInfo = new _deviceinfo.DeviceInfo(this.opponentDeviceId);
      return;
    }
    // if we've got to this point, we do want to init crypto, so throw if we can't
    if (!this.client.crypto) throw new Error("Crypto is not initialised.");
    const userId = this.invitee || ((_this$getOpponentMemb = this.getOpponentMember()) === null || _this$getOpponentMemb === void 0 ? void 0 : _this$getOpponentMemb.userId);
    if (!userId) throw new Error("Couldn't find opponent user ID to init crypto");
    const deviceInfoMap = await this.client.crypto.deviceList.downloadKeys([userId], false);
    this.opponentDeviceInfo = (_deviceInfoMap$get = deviceInfoMap.get(userId)) === null || _deviceInfoMap$get === void 0 ? void 0 : _deviceInfoMap$get.get(this.opponentDeviceId);
    if (this.opponentDeviceInfo === undefined) {
      throw new _groupCall.GroupCallUnknownDeviceError(userId);
    }
  }

  /**
   * Generates and returns localSDPStreamMetadata
   * @returns localSDPStreamMetadata
   */
  getLocalSDPStreamMetadata(updateStreamIds = false) {
    const metadata = {};
    for (const localFeed of this.getLocalFeeds()) {
      if (updateStreamIds) {
        localFeed.sdpMetadataStreamId = localFeed.stream.id;
      }
      metadata[localFeed.sdpMetadataStreamId] = {
        purpose: localFeed.purpose,
        audio_muted: localFeed.isAudioMuted(),
        video_muted: localFeed.isVideoMuted()
      };
    }
    return metadata;
  }

  /**
   * Returns true if there are no incoming feeds,
   * otherwise returns false
   * @returns no incoming feeds
   */
  noIncomingFeeds() {
    return !this.feeds.some(feed => !feed.isLocal());
  }
  pushRemoteFeed(stream) {
    // Fallback to old behavior if the other side doesn't support SDPStreamMetadata
    if (!this.opponentSupportsSDPStreamMetadata()) {
      this.pushRemoteFeedWithoutMetadata(stream);
      return;
    }
    const userId = this.getOpponentMember().userId;
    const purpose = this.remoteSDPStreamMetadata[stream.id].purpose;
    const audioMuted = this.remoteSDPStreamMetadata[stream.id].audio_muted;
    const videoMuted = this.remoteSDPStreamMetadata[stream.id].video_muted;
    if (!purpose) {
      _logger.logger.warn(`Call ${this.callId} pushRemoteFeed() ignoring stream because we didn't get any metadata about it (streamId=${stream.id})`);
      return;
    }
    if (this.getFeedByStreamId(stream.id)) {
      _logger.logger.warn(`Call ${this.callId} pushRemoteFeed() ignoring stream because we already have a feed for it (streamId=${stream.id})`);
      return;
    }
    this.feeds.push(new _callFeed.CallFeed({
      client: this.client,
      call: this,
      roomId: this.roomId,
      userId,
      deviceId: this.getOpponentDeviceId(),
      stream,
      purpose,
      audioMuted,
      videoMuted
    }));
    this.emit(CallEvent.FeedsChanged, this.feeds, this);
    _logger.logger.info(`Call ${this.callId} pushRemoteFeed() pushed stream (streamId=${stream.id}, active=${stream.active}, purpose=${purpose})`);
  }

  /**
   * This method is used ONLY if the other client doesn't support sending SDPStreamMetadata
   */
  pushRemoteFeedWithoutMetadata(stream) {
    var _this$feeds$find;
    const userId = this.getOpponentMember().userId;
    // We can guess the purpose here since the other client can only send one stream
    const purpose = _callEventTypes.SDPStreamMetadataPurpose.Usermedia;
    const oldRemoteStream = (_this$feeds$find = this.feeds.find(feed => !feed.isLocal())) === null || _this$feeds$find === void 0 ? void 0 : _this$feeds$find.stream;

    // Note that we check by ID and always set the remote stream: Chrome appears
    // to make new stream objects when transceiver directionality is changed and the 'active'
    // status of streams change - Dave
    // If we already have a stream, check this stream has the same id
    if (oldRemoteStream && stream.id !== oldRemoteStream.id) {
      _logger.logger.warn(`Call ${this.callId} pushRemoteFeedWithoutMetadata() ignoring new stream because we already have stream (streamId=${stream.id})`);
      return;
    }
    if (this.getFeedByStreamId(stream.id)) {
      _logger.logger.warn(`Call ${this.callId} pushRemoteFeedWithoutMetadata() ignoring stream because we already have a feed for it (streamId=${stream.id})`);
      return;
    }
    this.feeds.push(new _callFeed.CallFeed({
      client: this.client,
      call: this,
      roomId: this.roomId,
      audioMuted: false,
      videoMuted: false,
      userId,
      deviceId: this.getOpponentDeviceId(),
      stream,
      purpose
    }));
    this.emit(CallEvent.FeedsChanged, this.feeds, this);
    _logger.logger.info(`Call ${this.callId} pushRemoteFeedWithoutMetadata() pushed stream (streamId=${stream.id}, active=${stream.active})`);
  }
  pushNewLocalFeed(stream, purpose, addToPeerConnection = true) {
    const userId = this.client.getUserId();

    // Tracks don't always start off enabled, eg. chrome will give a disabled
    // audio track if you ask for user media audio and already had one that
    // you'd set to disabled (presumably because it clones them internally).
    setTracksEnabled(stream.getAudioTracks(), true);
    setTracksEnabled(stream.getVideoTracks(), true);
    if (this.getFeedByStreamId(stream.id)) {
      _logger.logger.warn(`Call ${this.callId} pushNewLocalFeed() ignoring stream because we already have a feed for it (streamId=${stream.id})`);
      return;
    }
    this.pushLocalFeed(new _callFeed.CallFeed({
      client: this.client,
      roomId: this.roomId,
      audioMuted: false,
      videoMuted: false,
      userId,
      deviceId: this.getOpponentDeviceId(),
      stream,
      purpose
    }), addToPeerConnection);
  }

  /**
   * Pushes supplied feed to the call
   * @param callFeed - to push
   * @param addToPeerConnection - whether to add the tracks to the peer connection
   */
  pushLocalFeed(callFeed, addToPeerConnection = true) {
    if (this.feeds.some(feed => callFeed.stream.id === feed.stream.id)) {
      _logger.logger.info(`Call ${this.callId} pushLocalFeed() ignoring duplicate local stream (streamId=${callFeed.stream.id})`);
      return;
    }
    this.feeds.push(callFeed);
    if (addToPeerConnection) {
      for (const track of callFeed.stream.getTracks()) {
        _logger.logger.info(`Call ${this.callId} pushLocalFeed() adding track to peer connection (id=${track.id}, kind=${track.kind}, streamId=${callFeed.stream.id}, streamPurpose=${callFeed.purpose}, enabled=${track.enabled})`);
        const tKey = getTransceiverKey(callFeed.purpose, track.kind);
        if (this.transceivers.has(tKey)) {
          // we already have a sender, so we re-use it. We try to re-use transceivers as much
          // as possible because they can't be removed once added, so otherwise they just
          // accumulate which makes the SDP very large very quickly: in fact it only takes
          // about 6 video tracks to exceed the maximum size of an Olm-encrypted
          // Matrix event.
          const transceiver = this.transceivers.get(tKey);
          transceiver.sender.replaceTrack(track);
          // set the direction to indicate we're going to start sending again
          // (this will trigger the re-negotiation)
          transceiver.direction = transceiver.direction === "inactive" ? "sendonly" : "sendrecv";
        } else {
          // create a new one. We need to use addTrack rather addTransceiver for this because firefox
          // doesn't yet implement RTCRTPSender.setStreams()
          // (https://bugzilla.mozilla.org/show_bug.cgi?id=1510802) so we'd have no way to group the
          // two tracks together into a stream.
          const newSender = this.peerConn.addTrack(track, callFeed.stream);

          // now go & fish for the new transceiver
          const newTransceiver = this.peerConn.getTransceivers().find(t => t.sender === newSender);
          if (newTransceiver) {
            this.transceivers.set(tKey, newTransceiver);
          } else {
            _logger.logger.warn(`Call ${this.callId} pushLocalFeed() didn't find a matching transceiver after adding track!`);
          }
        }
      }
    }
    _logger.logger.info(`Call ${this.callId} pushLocalFeed() pushed stream (id=${callFeed.stream.id}, active=${callFeed.stream.active}, purpose=${callFeed.purpose})`);
    this.emit(CallEvent.FeedsChanged, this.feeds, this);
  }

  /**
   * Removes local call feed from the call and its tracks from the peer
   * connection
   * @param callFeed - to remove
   */
  removeLocalFeed(callFeed) {
    const audioTransceiverKey = getTransceiverKey(callFeed.purpose, "audio");
    const videoTransceiverKey = getTransceiverKey(callFeed.purpose, "video");
    for (const transceiverKey of [audioTransceiverKey, videoTransceiverKey]) {
      // this is slightly mixing the track and transceiver API but is basically just shorthand.
      // There is no way to actually remove a transceiver, so this just sets it to inactive
      // (or recvonly) and replaces the source with nothing.
      if (this.transceivers.has(transceiverKey)) {
        const transceiver = this.transceivers.get(transceiverKey);
        if (transceiver.sender) this.peerConn.removeTrack(transceiver.sender);
      }
    }
    if (callFeed.purpose === _callEventTypes.SDPStreamMetadataPurpose.Screenshare) {
      this.client.getMediaHandler().stopScreensharingStream(callFeed.stream);
    }
    this.deleteFeed(callFeed);
  }
  deleteAllFeeds() {
    for (const feed of this.feeds) {
      if (!feed.isLocal() || !this.groupCallId) {
        feed.dispose();
      }
    }
    this.feeds = [];
    this.emit(CallEvent.FeedsChanged, this.feeds, this);
  }
  deleteFeedByStream(stream) {
    const feed = this.getFeedByStreamId(stream.id);
    if (!feed) {
      _logger.logger.warn(`Call ${this.callId} deleteFeedByStream() didn't find the feed to delete (streamId=${stream.id})`);
      return;
    }
    this.deleteFeed(feed);
  }
  deleteFeed(feed) {
    feed.dispose();
    this.feeds.splice(this.feeds.indexOf(feed), 1);
    this.emit(CallEvent.FeedsChanged, this.feeds, this);
  }

  // The typescript definitions have this type as 'any' :(
  async getCurrentCallStats() {
    if (this.callHasEnded()) {
      return this.callStatsAtEnd;
    }
    return this.collectCallStats();
  }
  async collectCallStats() {
    // This happens when the call fails before it starts.
    // For example when we fail to get capture sources
    if (!this.peerConn) return;
    const statsReport = await this.peerConn.getStats();
    const stats = [];
    statsReport.forEach(item => {
      stats.push(item);
    });
    return stats;
  }

  /**
   * Configure this call from an invite event. Used by MatrixClient.
   * @param event - The m.call.invite event
   */
  async initWithInvite(event) {
    var _this$feeds$find2;
    const invite = event.getContent();
    this.direction = CallDirection.Inbound;

    // make sure we have valid turn creds. Unless something's gone wrong, it should
    // poll and keep the credentials valid so this should be instant.
    const haveTurnCreds = await this.client.checkTurnServers();
    if (!haveTurnCreds) {
      _logger.logger.warn(`Call ${this.callId} initWithInvite() failed to get TURN credentials! Proceeding with call anyway...`);
    }
    const sdpStreamMetadata = invite[_callEventTypes.SDPStreamMetadataKey];
    if (sdpStreamMetadata) {
      this.updateRemoteSDPStreamMetadata(sdpStreamMetadata);
    } else {
      _logger.logger.debug(`Call ${this.callId} initWithInvite() did not get any SDPStreamMetadata! Can not send/receive multiple streams`);
    }
    this.peerConn = this.createPeerConnection();
    // we must set the party ID before await-ing on anything: the call event
    // handler will start giving us more call events (eg. candidates) so if
    // we haven't set the party ID, we'll ignore them.
    this.chooseOpponent(event);
    await this.initOpponentCrypto();
    try {
      await this.peerConn.setRemoteDescription(invite.offer);
      await this.addBufferedIceCandidates();
    } catch (e) {
      _logger.logger.debug(`Call ${this.callId} initWithInvite() failed to set remote description`, e);
      this.terminate(CallParty.Local, CallErrorCode.SetRemoteDescription, false);
      return;
    }
    const remoteStream = (_this$feeds$find2 = this.feeds.find(feed => !feed.isLocal())) === null || _this$feeds$find2 === void 0 ? void 0 : _this$feeds$find2.stream;

    // According to previous comments in this file, firefox at some point did not
    // add streams until media started arriving on them. Testing latest firefox
    // (81 at time of writing), this is no longer a problem, so let's do it the correct way.
    //
    // For example in case of no media webrtc connections like screen share only call we have to allow webrtc
    // connections without remote media. In this case we always use a data channel. At the moment we allow as well
    // only data channel as media in the WebRTC connection with this setup here.
    if (!this.isOnlyDataChannelAllowed && (!remoteStream || remoteStream.getTracks().length === 0)) {
      _logger.logger.error(`Call ${this.callId} initWithInvite() no remote stream or no tracks after setting remote description!`);
      this.terminate(CallParty.Local, CallErrorCode.SetRemoteDescription, false);
      return;
    }
    this.state = CallState.Ringing;
    if (event.getLocalAge()) {
      // Time out the call if it's ringing for too long
      const ringingTimer = setTimeout(() => {
        if (this.state == CallState.Ringing) {
          var _this$stats;
          _logger.logger.debug(`Call ${this.callId} initWithInvite() invite has expired. Hanging up.`);
          this.hangupParty = CallParty.Remote; // effectively
          this.state = CallState.Ended;
          this.stopAllMedia();
          if (this.peerConn.signalingState != "closed") {
            this.peerConn.close();
          }
          (_this$stats = this.stats) === null || _this$stats === void 0 ? void 0 : _this$stats.removeStatsReportGatherer(this.callId);
          this.emit(CallEvent.Hangup, this);
        }
      }, invite.lifetime - event.getLocalAge());
      const onState = state => {
        if (state !== CallState.Ringing) {
          clearTimeout(ringingTimer);
          this.off(CallEvent.State, onState);
        }
      };
      this.on(CallEvent.State, onState);
    }
  }

  /**
   * Configure this call from a hangup or reject event. Used by MatrixClient.
   * @param event - The m.call.hangup event
   */
  initWithHangup(event) {
    // perverse as it may seem, sometimes we want to instantiate a call with a
    // hangup message (because when getting the state of the room on load, events
    // come in reverse order and we want to remember that a call has been hung up)
    this.state = CallState.Ended;
  }
  shouldAnswerWithMediaType(wantedValue, valueOfTheOtherSide, type) {
    if (wantedValue && !valueOfTheOtherSide) {
      // TODO: Figure out how to do this
      _logger.logger.warn(`Call ${this.callId} shouldAnswerWithMediaType() unable to answer with ${type} because the other side isn't sending it either.`);
      return false;
    } else if (!utils.isNullOrUndefined(wantedValue) && wantedValue !== valueOfTheOtherSide && !this.opponentSupportsSDPStreamMetadata()) {
      _logger.logger.warn(`Call ${this.callId} shouldAnswerWithMediaType() unable to answer with ${type}=${wantedValue} because the other side doesn't support it. Answering with ${type}=${valueOfTheOtherSide}.`);
      return valueOfTheOtherSide;
    }
    return wantedValue !== null && wantedValue !== void 0 ? wantedValue : valueOfTheOtherSide;
  }

  /**
   * Answer a call.
   */
  async answer(audio, video) {
    if (this.inviteOrAnswerSent) return;
    // TODO: Figure out how to do this
    if (audio === false && video === false) throw new Error("You CANNOT answer a call without media");
    if (!this.localUsermediaStream && !this.waitForLocalAVStream) {
      const prevState = this.state;
      const answerWithAudio = this.shouldAnswerWithMediaType(audio, this.hasRemoteUserMediaAudioTrack, "audio");
      const answerWithVideo = this.shouldAnswerWithMediaType(video, this.hasRemoteUserMediaVideoTrack, "video");
      this.state = CallState.WaitLocalMedia;
      this.waitForLocalAVStream = true;
      try {
        var _this$client$getDevic;
        const stream = await this.client.getMediaHandler().getUserMediaStream(answerWithAudio, answerWithVideo);
        this.waitForLocalAVStream = false;
        const usermediaFeed = new _callFeed.CallFeed({
          client: this.client,
          roomId: this.roomId,
          userId: this.client.getUserId(),
          deviceId: (_this$client$getDevic = this.client.getDeviceId()) !== null && _this$client$getDevic !== void 0 ? _this$client$getDevic : undefined,
          stream,
          purpose: _callEventTypes.SDPStreamMetadataPurpose.Usermedia,
          audioMuted: false,
          videoMuted: false
        });
        const feeds = [usermediaFeed];
        if (this.localScreensharingFeed) {
          feeds.push(this.localScreensharingFeed);
        }
        this.answerWithCallFeeds(feeds);
      } catch (e) {
        if (answerWithVideo) {
          // Try to answer without video
          _logger.logger.warn(`Call ${this.callId} answer() failed to getUserMedia(), trying to getUserMedia() without video`);
          this.state = prevState;
          this.waitForLocalAVStream = false;
          await this.answer(answerWithAudio, false);
        } else {
          this.getUserMediaFailed(e);
          return;
        }
      }
    } else if (this.waitForLocalAVStream) {
      this.state = CallState.WaitLocalMedia;
    }
  }
  answerWithCallFeeds(callFeeds) {
    if (this.inviteOrAnswerSent) return;
    this.queueGotCallFeedsForAnswer(callFeeds);
  }

  /**
   * Replace this call with a new call, e.g. for glare resolution. Used by
   * MatrixClient.
   * @param newCall - The new call.
   */
  replacedBy(newCall) {
    _logger.logger.debug(`Call ${this.callId} replacedBy() running (newCallId=${newCall.callId})`);
    if (this.state === CallState.WaitLocalMedia) {
      _logger.logger.debug(`Call ${this.callId} replacedBy() telling new call to wait for local media (newCallId=${newCall.callId})`);
      newCall.waitForLocalAVStream = true;
    } else if ([CallState.CreateOffer, CallState.InviteSent].includes(this.state)) {
      if (newCall.direction === CallDirection.Outbound) {
        newCall.queueGotCallFeedsForAnswer([]);
      } else {
        _logger.logger.debug(`Call ${this.callId} replacedBy() handing local stream to new call(newCallId=${newCall.callId})`);
        newCall.queueGotCallFeedsForAnswer(this.getLocalFeeds().map(feed => feed.clone()));
      }
    }
    this.successor = newCall;
    this.emit(CallEvent.Replaced, newCall, this);
    this.hangup(CallErrorCode.Replaced, true);
  }

  /**
   * Hangup a call.
   * @param reason - The reason why the call is being hung up.
   * @param suppressEvent - True to suppress emitting an event.
   */
  hangup(reason, suppressEvent) {
    if (this.callHasEnded()) return;
    _logger.logger.debug(`Call ${this.callId} hangup() ending call (reason=${reason})`);
    this.terminate(CallParty.Local, reason, !suppressEvent);
    // We don't want to send hangup here if we didn't even get to sending an invite
    if ([CallState.Fledgling, CallState.WaitLocalMedia].includes(this.state)) return;
    const content = {};
    // Don't send UserHangup reason to older clients
    if (this.opponentVersion && this.opponentVersion !== 0 || reason !== CallErrorCode.UserHangup) {
      content["reason"] = reason;
    }
    this.sendVoipEvent(_event.EventType.CallHangup, content);
  }

  /**
   * Reject a call
   * This used to be done by calling hangup, but is a separate method and protocol
   * event as of MSC2746.
   */
  reject() {
    if (this.state !== CallState.Ringing) {
      throw Error("Call must be in 'ringing' state to reject!");
    }
    if (this.opponentVersion === 0) {
      _logger.logger.info(`Call ${this.callId} reject() opponent version is less than 1: sending hangup instead of reject (opponentVersion=${this.opponentVersion})`);
      this.hangup(CallErrorCode.UserHangup, true);
      return;
    }
    _logger.logger.debug("Rejecting call: " + this.callId);
    this.terminate(CallParty.Local, CallErrorCode.UserHangup, true);
    this.sendVoipEvent(_event.EventType.CallReject, {});
  }

  /**
   * Adds an audio and/or video track - upgrades the call
   * @param audio - should add an audio track
   * @param video - should add an video track
   */
  async upgradeCall(audio, video) {
    // We don't do call downgrades
    if (!audio && !video) return;
    if (!this.opponentSupportsSDPStreamMetadata()) return;
    try {
      _logger.logger.debug(`Call ${this.callId} upgradeCall() upgrading call (audio=${audio}, video=${video})`);
      const getAudio = audio || this.hasLocalUserMediaAudioTrack;
      const getVideo = video || this.hasLocalUserMediaVideoTrack;

      // updateLocalUsermediaStream() will take the tracks, use them as
      // replacement and throw the stream away, so it isn't reusable
      const stream = await this.client.getMediaHandler().getUserMediaStream(getAudio, getVideo, false);
      await this.updateLocalUsermediaStream(stream, audio, video);
    } catch (error) {
      _logger.logger.error(`Call ${this.callId} upgradeCall() failed to upgrade the call`, error);
      this.emit(CallEvent.Error, new CallError(CallErrorCode.NoUserMedia, "Failed to get camera access: ", error), this);
    }
  }

  /**
   * Returns true if this.remoteSDPStreamMetadata is defined, otherwise returns false
   * @returns can screenshare
   */
  opponentSupportsSDPStreamMetadata() {
    return Boolean(this.remoteSDPStreamMetadata);
  }

  /**
   * If there is a screensharing stream returns true, otherwise returns false
   * @returns is screensharing
   */
  isScreensharing() {
    return Boolean(this.localScreensharingStream);
  }

  /**
   * Starts/stops screensharing
   * @param enabled - the desired screensharing state
   * @param desktopCapturerSourceId - optional id of the desktop capturer source to use
   * @returns new screensharing state
   */
  async setScreensharingEnabled(enabled, opts) {
    // Skip if there is nothing to do
    if (enabled && this.isScreensharing()) {
      _logger.logger.warn(`Call ${this.callId} setScreensharingEnabled() there is already a screensharing stream - there is nothing to do!`);
      return true;
    } else if (!enabled && !this.isScreensharing()) {
      _logger.logger.warn(`Call ${this.callId} setScreensharingEnabled() there already isn't a screensharing stream - there is nothing to do!`);
      return false;
    }

    // Fallback to replaceTrack()
    if (!this.opponentSupportsSDPStreamMetadata()) {
      return this.setScreensharingEnabledWithoutMetadataSupport(enabled, opts);
    }
    _logger.logger.debug(`Call ${this.callId} setScreensharingEnabled() running (enabled=${enabled})`);
    if (enabled) {
      try {
        const stream = await this.client.getMediaHandler().getScreensharingStream(opts);
        if (!stream) return false;
        this.pushNewLocalFeed(stream, _callEventTypes.SDPStreamMetadataPurpose.Screenshare);
        return true;
      } catch (err) {
        _logger.logger.error(`Call ${this.callId} setScreensharingEnabled() failed to get screen-sharing stream:`, err);
        return false;
      }
    } else {
      const audioTransceiver = this.transceivers.get(getTransceiverKey(_callEventTypes.SDPStreamMetadataPurpose.Screenshare, "audio"));
      const videoTransceiver = this.transceivers.get(getTransceiverKey(_callEventTypes.SDPStreamMetadataPurpose.Screenshare, "video"));
      for (const transceiver of [audioTransceiver, videoTransceiver]) {
        // this is slightly mixing the track and transceiver API but is basically just shorthand
        // for removing the sender.
        if (transceiver && transceiver.sender) this.peerConn.removeTrack(transceiver.sender);
      }
      this.client.getMediaHandler().stopScreensharingStream(this.localScreensharingStream);
      this.deleteFeedByStream(this.localScreensharingStream);
      return false;
    }
  }

  /**
   * Starts/stops screensharing
   * Should be used ONLY if the opponent doesn't support SDPStreamMetadata
   * @param enabled - the desired screensharing state
   * @param desktopCapturerSourceId - optional id of the desktop capturer source to use
   * @returns new screensharing state
   */
  async setScreensharingEnabledWithoutMetadataSupport(enabled, opts) {
    _logger.logger.debug(`Call ${this.callId} setScreensharingEnabledWithoutMetadataSupport() running (enabled=${enabled})`);
    if (enabled) {
      try {
        var _this$transceivers$ge3;
        const stream = await this.client.getMediaHandler().getScreensharingStream(opts);
        if (!stream) return false;
        const track = stream.getTracks().find(track => track.kind === "video");
        const sender = (_this$transceivers$ge3 = this.transceivers.get(getTransceiverKey(_callEventTypes.SDPStreamMetadataPurpose.Usermedia, "video"))) === null || _this$transceivers$ge3 === void 0 ? void 0 : _this$transceivers$ge3.sender;
        sender === null || sender === void 0 ? void 0 : sender.replaceTrack(track !== null && track !== void 0 ? track : null);
        this.pushNewLocalFeed(stream, _callEventTypes.SDPStreamMetadataPurpose.Screenshare, false);
        return true;
      } catch (err) {
        _logger.logger.error(`Call ${this.callId} setScreensharingEnabledWithoutMetadataSupport() failed to get screen-sharing stream:`, err);
        return false;
      }
    } else {
      var _this$localUsermediaS3, _this$transceivers$ge4;
      const track = (_this$localUsermediaS3 = this.localUsermediaStream) === null || _this$localUsermediaS3 === void 0 ? void 0 : _this$localUsermediaS3.getTracks().find(track => track.kind === "video");
      const sender = (_this$transceivers$ge4 = this.transceivers.get(getTransceiverKey(_callEventTypes.SDPStreamMetadataPurpose.Usermedia, "video"))) === null || _this$transceivers$ge4 === void 0 ? void 0 : _this$transceivers$ge4.sender;
      sender === null || sender === void 0 ? void 0 : sender.replaceTrack(track !== null && track !== void 0 ? track : null);
      this.client.getMediaHandler().stopScreensharingStream(this.localScreensharingStream);
      this.deleteFeedByStream(this.localScreensharingStream);
      return false;
    }
  }

  /**
   * Replaces/adds the tracks from the passed stream to the localUsermediaStream
   * @param stream - to use a replacement for the local usermedia stream
   */
  async updateLocalUsermediaStream(stream, forceAudio = false, forceVideo = false) {
    const callFeed = this.localUsermediaFeed;
    const audioEnabled = forceAudio || !callFeed.isAudioMuted() && !this.remoteOnHold;
    const videoEnabled = forceVideo || !callFeed.isVideoMuted() && !this.remoteOnHold;
    _logger.logger.log(`Call ${this.callId} updateLocalUsermediaStream() running (streamId=${stream.id}, audio=${audioEnabled}, video=${videoEnabled})`);
    setTracksEnabled(stream.getAudioTracks(), audioEnabled);
    setTracksEnabled(stream.getVideoTracks(), videoEnabled);

    // We want to keep the same stream id, so we replace the tracks rather
    // than the whole stream.

    // Firstly, we replace the tracks in our localUsermediaStream.
    for (const track of this.localUsermediaStream.getTracks()) {
      this.localUsermediaStream.removeTrack(track);
      track.stop();
    }
    for (const track of stream.getTracks()) {
      this.localUsermediaStream.addTrack(track);
    }

    // Then replace the old tracks, if possible.
    for (const track of stream.getTracks()) {
      const tKey = getTransceiverKey(_callEventTypes.SDPStreamMetadataPurpose.Usermedia, track.kind);
      const transceiver = this.transceivers.get(tKey);
      const oldSender = transceiver === null || transceiver === void 0 ? void 0 : transceiver.sender;
      let added = false;
      if (oldSender) {
        try {
          _logger.logger.info(`Call ${this.callId} updateLocalUsermediaStream() replacing track (id=${track.id}, kind=${track.kind}, streamId=${stream.id}, streamPurpose=${callFeed.purpose})`);
          await oldSender.replaceTrack(track);
          // Set the direction to indicate we're going to be sending.
          // This is only necessary in the cases where we're upgrading
          // the call to video after downgrading it.
          transceiver.direction = transceiver.direction === "inactive" ? "sendonly" : "sendrecv";
          added = true;
        } catch (error) {
          _logger.logger.warn(`Call ${this.callId} updateLocalUsermediaStream() replaceTrack failed: adding new transceiver instead`, error);
        }
      }
      if (!added) {
        _logger.logger.info(`Call ${this.callId} updateLocalUsermediaStream() adding track to peer connection (id=${track.id}, kind=${track.kind}, streamId=${stream.id}, streamPurpose=${callFeed.purpose})`);
        const newSender = this.peerConn.addTrack(track, this.localUsermediaStream);
        const newTransceiver = this.peerConn.getTransceivers().find(t => t.sender === newSender);
        if (newTransceiver) {
          this.transceivers.set(tKey, newTransceiver);
        } else {
          _logger.logger.warn(`Call ${this.callId} updateLocalUsermediaStream() couldn't find matching transceiver for newly added track!`);
        }
      }
    }
  }

  /**
   * Set whether our outbound video should be muted or not.
   * @param muted - True to mute the outbound video.
   * @returns the new mute state
   */
  async setLocalVideoMuted(muted) {
    var _this$localUsermediaF3;
    _logger.logger.log(`Call ${this.callId} setLocalVideoMuted() running ${muted}`);

    // if we were still thinking about stopping and removing the video
    // track: don't, because we want it back.
    if (!muted && this.stopVideoTrackTimer !== undefined) {
      clearTimeout(this.stopVideoTrackTimer);
      this.stopVideoTrackTimer = undefined;
    }
    if (!(await this.client.getMediaHandler().hasVideoDevice())) {
      return this.isLocalVideoMuted();
    }
    if (!this.hasUserMediaVideoSender && !muted) {
      var _this$localUsermediaF2;
      (_this$localUsermediaF2 = this.localUsermediaFeed) === null || _this$localUsermediaF2 === void 0 ? void 0 : _this$localUsermediaF2.setAudioVideoMuted(null, muted);
      await this.upgradeCall(false, true);
      return this.isLocalVideoMuted();
    }

    // we may not have a video track - if not, re-request usermedia
    if (!muted && this.localUsermediaStream.getVideoTracks().length === 0) {
      const stream = await this.client.getMediaHandler().getUserMediaStream(true, true);
      await this.updateLocalUsermediaStream(stream);
    }
    (_this$localUsermediaF3 = this.localUsermediaFeed) === null || _this$localUsermediaF3 === void 0 ? void 0 : _this$localUsermediaF3.setAudioVideoMuted(null, muted);
    this.updateMuteStatus();
    await this.sendMetadataUpdate();

    // if we're muting video, set a timeout to stop & remove the video track so we release
    // the camera. We wait a short time to do this because when we disable a track, WebRTC
    // will send black video for it. If we just stop and remove it straight away, the video
    // will just freeze which means that when we unmute video, the other side will briefly
    // get a static frame of us from before we muted. This way, the still frame is just black.
    // A very small delay is not always enough so the theory here is that it needs to be long
    // enough for WebRTC to encode a frame: 120ms should be long enough even if we're only
    // doing 10fps.
    if (muted) {
      this.stopVideoTrackTimer = setTimeout(() => {
        for (const t of this.localUsermediaStream.getVideoTracks()) {
          t.stop();
          this.localUsermediaStream.removeTrack(t);
        }
      }, 120);
    }
    return this.isLocalVideoMuted();
  }

  /**
   * Check if local video is muted.
   *
   * If there are multiple video tracks, <i>all</i> of the tracks need to be muted
   * for this to return true. This means if there are no video tracks, this will
   * return true.
   * @returns True if the local preview video is muted, else false
   * (including if the call is not set up yet).
   */
  isLocalVideoMuted() {
    var _this$localUsermediaF4, _this$localUsermediaF5;
    return (_this$localUsermediaF4 = (_this$localUsermediaF5 = this.localUsermediaFeed) === null || _this$localUsermediaF5 === void 0 ? void 0 : _this$localUsermediaF5.isVideoMuted()) !== null && _this$localUsermediaF4 !== void 0 ? _this$localUsermediaF4 : false;
  }

  /**
   * Set whether the microphone should be muted or not.
   * @param muted - True to mute the mic.
   * @returns the new mute state
   */
  async setMicrophoneMuted(muted) {
    var _this$localUsermediaF6;
    _logger.logger.log(`Call ${this.callId} setMicrophoneMuted() running ${muted}`);
    if (!(await this.client.getMediaHandler().hasAudioDevice())) {
      return this.isMicrophoneMuted();
    }
    if (!muted && (!this.hasUserMediaAudioSender || !this.hasLocalUserMediaAudioTrack)) {
      await this.upgradeCall(true, false);
      return this.isMicrophoneMuted();
    }
    (_this$localUsermediaF6 = this.localUsermediaFeed) === null || _this$localUsermediaF6 === void 0 ? void 0 : _this$localUsermediaF6.setAudioVideoMuted(muted, null);
    this.updateMuteStatus();
    await this.sendMetadataUpdate();
    return this.isMicrophoneMuted();
  }

  /**
   * Check if the microphone is muted.
   *
   * If there are multiple audio tracks, <i>all</i> of the tracks need to be muted
   * for this to return true. This means if there are no audio tracks, this will
   * return true.
   * @returns True if the mic is muted, else false (including if the call
   * is not set up yet).
   */
  isMicrophoneMuted() {
    var _this$localUsermediaF7, _this$localUsermediaF8;
    return (_this$localUsermediaF7 = (_this$localUsermediaF8 = this.localUsermediaFeed) === null || _this$localUsermediaF8 === void 0 ? void 0 : _this$localUsermediaF8.isAudioMuted()) !== null && _this$localUsermediaF7 !== void 0 ? _this$localUsermediaF7 : false;
  }

  /**
   * @returns true if we have put the party on the other side of the call on hold
   * (that is, we are signalling to them that we are not listening)
   */
  isRemoteOnHold() {
    return this.remoteOnHold;
  }
  setRemoteOnHold(onHold) {
    if (this.isRemoteOnHold() === onHold) return;
    this.remoteOnHold = onHold;
    for (const transceiver of this.peerConn.getTransceivers()) {
      // We don't send hold music or anything so we're not actually
      // sending anything, but sendrecv is fairly standard for hold and
      // it makes it a lot easier to figure out who's put who on hold.
      transceiver.direction = onHold ? "sendonly" : "sendrecv";
    }
    this.updateMuteStatus();
    this.sendMetadataUpdate();
    this.emit(CallEvent.RemoteHoldUnhold, this.remoteOnHold, this);
  }

  /**
   * Indicates whether we are 'on hold' to the remote party (ie. if true,
   * they cannot hear us).
   * @returns true if the other party has put us on hold
   */
  isLocalOnHold() {
    if (this.state !== CallState.Connected) return false;
    let callOnHold = true;

    // We consider a call to be on hold only if *all* the tracks are on hold
    // (is this the right thing to do?)
    for (const transceiver of this.peerConn.getTransceivers()) {
      const trackOnHold = ["inactive", "recvonly"].includes(transceiver.currentDirection);
      if (!trackOnHold) callOnHold = false;
    }
    return callOnHold;
  }

  /**
   * Sends a DTMF digit to the other party
   * @param digit - The digit (nb. string - '#' and '*' are dtmf too)
   */
  sendDtmfDigit(digit) {
    for (const sender of this.peerConn.getSenders()) {
      var _sender$track;
      if (((_sender$track = sender.track) === null || _sender$track === void 0 ? void 0 : _sender$track.kind) === "audio" && sender.dtmf) {
        sender.dtmf.insertDTMF(digit);
        return;
      }
    }
    throw new Error("Unable to find a track to send DTMF on");
  }
  updateMuteStatus() {
    const micShouldBeMuted = this.isMicrophoneMuted() || this.remoteOnHold;
    const vidShouldBeMuted = this.isLocalVideoMuted() || this.remoteOnHold;
    _logger.logger.log(`Call ${this.callId} updateMuteStatus stream ${this.localUsermediaStream.id} micShouldBeMuted ${micShouldBeMuted} vidShouldBeMuted ${vidShouldBeMuted}`);
    setTracksEnabled(this.localUsermediaStream.getAudioTracks(), !micShouldBeMuted);
    setTracksEnabled(this.localUsermediaStream.getVideoTracks(), !vidShouldBeMuted);
  }
  async sendMetadataUpdate() {
    await this.sendVoipEvent(_event.EventType.CallSDPStreamMetadataChangedPrefix, {
      [_callEventTypes.SDPStreamMetadataKey]: this.getLocalSDPStreamMetadata()
    });
  }
  gotCallFeedsForInvite(callFeeds, requestScreenshareFeed = false) {
    if (this.successor) {
      this.successor.queueGotCallFeedsForAnswer(callFeeds);
      return;
    }
    if (this.callHasEnded()) {
      this.stopAllMedia();
      return;
    }
    for (const feed of callFeeds) {
      this.pushLocalFeed(feed);
    }
    if (requestScreenshareFeed) {
      this.peerConn.addTransceiver("video", {
        direction: "recvonly"
      });
    }
    this.state = CallState.CreateOffer;
    _logger.logger.debug(`Call ${this.callId} gotUserMediaForInvite() run`);
    // Now we wait for the negotiationneeded event
  }

  async sendAnswer() {
    const answerContent = {
      answer: {
        sdp: this.peerConn.localDescription.sdp,
        // type is now deprecated as of Matrix VoIP v1, but
        // required to still be sent for backwards compat
        type: this.peerConn.localDescription.type
      },
      [_callEventTypes.SDPStreamMetadataKey]: this.getLocalSDPStreamMetadata(true)
    };
    answerContent.capabilities = {
      "m.call.transferee": this.client.supportsCallTransfer,
      "m.call.dtmf": false
    };

    // We have just taken the local description from the peerConn which will
    // contain all the local candidates added so far, so we can discard any candidates
    // we had queued up because they'll be in the answer.
    const discardCount = this.discardDuplicateCandidates();
    _logger.logger.info(`Call ${this.callId} sendAnswer() discarding ${discardCount} candidates that will be sent in answer`);
    try {
      await this.sendVoipEvent(_event.EventType.CallAnswer, answerContent);
      // If this isn't the first time we've tried to send the answer,
      // we may have candidates queued up, so send them now.
      this.inviteOrAnswerSent = true;
    } catch (error) {
      // We've failed to answer: back to the ringing state
      this.state = CallState.Ringing;
      if (error instanceof _httpApi.MatrixError && error.event) this.client.cancelPendingEvent(error.event);
      let code = CallErrorCode.SendAnswer;
      let message = "Failed to send answer";
      if (error.name == "UnknownDeviceError") {
        code = CallErrorCode.UnknownDevices;
        message = "Unknown devices present in the room";
      }
      this.emit(CallEvent.Error, new CallError(code, message, error), this);
      throw error;
    }

    // error handler re-throws so this won't happen on error, but
    // we don't want the same error handling on the candidate queue
    this.sendCandidateQueue();
  }
  queueGotCallFeedsForAnswer(callFeeds) {
    // Ensure only one negotiate/answer event is being processed at a time.
    if (this.responsePromiseChain) {
      this.responsePromiseChain = this.responsePromiseChain.then(() => this.gotCallFeedsForAnswer(callFeeds));
    } else {
      this.responsePromiseChain = this.gotCallFeedsForAnswer(callFeeds);
    }
  }

  // Enables DTX (discontinuous transmission) on the given session to reduce
  // bandwidth when transmitting silence
  mungeSdp(description, mods) {
    // The only way to enable DTX at this time is through SDP munging
    const sdp = (0, _sdpTransform.parse)(description.sdp);
    sdp.media.forEach(media => {
      const payloadTypeToCodecMap = new Map();
      const codecToPayloadTypeMap = new Map();
      for (const rtp of media.rtp) {
        payloadTypeToCodecMap.set(rtp.payload, rtp.codec);
        codecToPayloadTypeMap.set(rtp.codec, rtp.payload);
      }
      for (const mod of mods) {
        if (mod.mediaType !== media.type) continue;
        if (!codecToPayloadTypeMap.has(mod.codec)) {
          _logger.logger.info(`Call ${this.callId} mungeSdp() ignoring SDP modifications for ${mod.codec} as it's not present.`);
          continue;
        }
        const extraConfig = [];
        if (mod.enableDtx !== undefined) {
          extraConfig.push(`usedtx=${mod.enableDtx ? "1" : "0"}`);
        }
        if (mod.maxAverageBitrate !== undefined) {
          extraConfig.push(`maxaveragebitrate=${mod.maxAverageBitrate}`);
        }
        let found = false;
        for (const fmtp of media.fmtp) {
          if (payloadTypeToCodecMap.get(fmtp.payload) === mod.codec) {
            found = true;
            fmtp.config += ";" + extraConfig.join(";");
          }
        }
        if (!found) {
          media.fmtp.push({
            payload: codecToPayloadTypeMap.get(mod.codec),
            config: extraConfig.join(";")
          });
        }
      }
    });
    description.sdp = (0, _sdpTransform.write)(sdp);
  }
  async createOffer() {
    const offer = await this.peerConn.createOffer();
    this.mungeSdp(offer, getCodecParamMods(this.isPtt));
    return offer;
  }
  async createAnswer() {
    const answer = await this.peerConn.createAnswer();
    this.mungeSdp(answer, getCodecParamMods(this.isPtt));
    return answer;
  }
  async gotCallFeedsForAnswer(callFeeds) {
    if (this.callHasEnded()) return;
    this.waitForLocalAVStream = false;
    for (const feed of callFeeds) {
      this.pushLocalFeed(feed);
    }
    this.state = CallState.CreateAnswer;
    let answer;
    try {
      this.getRidOfRTXCodecs();
      answer = await this.createAnswer();
    } catch (err) {
      _logger.logger.debug(`Call ${this.callId} gotCallFeedsForAnswer() failed to create answer: `, err);
      this.terminate(CallParty.Local, CallErrorCode.CreateAnswer, true);
      return;
    }
    try {
      await this.peerConn.setLocalDescription(answer);

      // make sure we're still going
      if (this.callHasEnded()) return;
      this.state = CallState.Connecting;

      // Allow a short time for initial candidates to be gathered
      await new Promise(resolve => {
        setTimeout(resolve, 200);
      });

      // make sure the call hasn't ended before we continue
      if (this.callHasEnded()) return;
      this.sendAnswer();
    } catch (err) {
      _logger.logger.debug(`Call ${this.callId} gotCallFeedsForAnswer() error setting local description!`, err);
      this.terminate(CallParty.Local, CallErrorCode.SetLocalDescription, true);
      return;
    }
  }

  /**
   * Internal
   */

  async onRemoteIceCandidatesReceived(ev) {
    if (this.callHasEnded()) {
      //debuglog("Ignoring remote ICE candidate because call has ended");
      return;
    }
    const content = ev.getContent();
    const candidates = content.candidates;
    if (!candidates) {
      _logger.logger.info(`Call ${this.callId} onRemoteIceCandidatesReceived() ignoring candidates event with no candidates!`);
      return;
    }
    const fromPartyId = content.version === 0 ? null : content.party_id || null;
    if (this.opponentPartyId === undefined) {
      // we haven't picked an opponent yet so save the candidates
      if (fromPartyId) {
        _logger.logger.info(`Call ${this.callId} onRemoteIceCandidatesReceived() buffering ${candidates.length} candidates until we pick an opponent`);
        const bufferedCandidates = this.remoteCandidateBuffer.get(fromPartyId) || [];
        bufferedCandidates.push(...candidates);
        this.remoteCandidateBuffer.set(fromPartyId, bufferedCandidates);
      }
      return;
    }
    if (!this.partyIdMatches(content)) {
      _logger.logger.info(`Call ${this.callId} onRemoteIceCandidatesReceived() ignoring candidates from party ID ${content.party_id}: we have chosen party ID ${this.opponentPartyId}`);
      return;
    }
    await this.addIceCandidates(candidates);
  }

  /**
   * Used by MatrixClient.
   */
  async onAnswerReceived(event) {
    const content = event.getContent();
    _logger.logger.debug(`Call ${this.callId} onAnswerReceived() running (hangupParty=${content.party_id})`);
    if (this.callHasEnded()) {
      _logger.logger.debug(`Call ${this.callId} onAnswerReceived() ignoring answer because call has ended`);
      return;
    }
    if (this.opponentPartyId !== undefined) {
      _logger.logger.info(`Call ${this.callId} onAnswerReceived() ignoring answer from party ID ${content.party_id}: we already have an answer/reject from ${this.opponentPartyId}`);
      return;
    }
    this.chooseOpponent(event);
    await this.addBufferedIceCandidates();
    this.state = CallState.Connecting;
    const sdpStreamMetadata = content[_callEventTypes.SDPStreamMetadataKey];
    if (sdpStreamMetadata) {
      this.updateRemoteSDPStreamMetadata(sdpStreamMetadata);
    } else {
      _logger.logger.warn(`Call ${this.callId} onAnswerReceived() did not get any SDPStreamMetadata! Can not send/receive multiple streams`);
    }
    try {
      await this.peerConn.setRemoteDescription(content.answer);
    } catch (e) {
      _logger.logger.debug(`Call ${this.callId} onAnswerReceived() failed to set remote description`, e);
      this.terminate(CallParty.Local, CallErrorCode.SetRemoteDescription, false);
      return;
    }

    // If the answer we selected has a party_id, send a select_answer event
    // We do this after setting the remote description since otherwise we'd block
    // call setup on it
    if (this.opponentPartyId !== null) {
      try {
        await this.sendVoipEvent(_event.EventType.CallSelectAnswer, {
          selected_party_id: this.opponentPartyId
        });
      } catch (err) {
        // This isn't fatal, and will just mean that if another party has raced to answer
        // the call, they won't know they got rejected, so we carry on & don't retry.
        _logger.logger.warn(`Call ${this.callId} onAnswerReceived() failed to send select_answer event`, err);
      }
    }
  }
  async onSelectAnswerReceived(event) {
    if (this.direction !== CallDirection.Inbound) {
      _logger.logger.warn(`Call ${this.callId} onSelectAnswerReceived() got select_answer for an outbound call: ignoring`);
      return;
    }
    const selectedPartyId = event.getContent().selected_party_id;
    if (selectedPartyId === undefined || selectedPartyId === null) {
      _logger.logger.warn(`Call ${this.callId} onSelectAnswerReceived() got nonsensical select_answer with null/undefined selected_party_id: ignoring`);
      return;
    }
    if (selectedPartyId !== this.ourPartyId) {
      _logger.logger.info(`Call ${this.callId} onSelectAnswerReceived() got select_answer for party ID ${selectedPartyId}: we are party ID ${this.ourPartyId}.`);
      // The other party has picked somebody else's answer
      await this.terminate(CallParty.Remote, CallErrorCode.AnsweredElsewhere, true);
    }
  }
  async onNegotiateReceived(event) {
    const content = event.getContent();
    const description = content.description;
    if (!description || !description.sdp || !description.type) {
      _logger.logger.info(`Call ${this.callId} onNegotiateReceived() ignoring invalid m.call.negotiate event`);
      return;
    }
    // Politeness always follows the direction of the call: in a glare situation,
    // we pick either the inbound or outbound call, so one side will always be
    // inbound and one outbound
    const polite = this.direction === CallDirection.Inbound;

    // Here we follow the perfect negotiation logic from
    // https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/Perfect_negotiation
    const offerCollision = description.type === "offer" && (this.makingOffer || this.peerConn.signalingState !== "stable");
    this.ignoreOffer = !polite && offerCollision;
    if (this.ignoreOffer) {
      _logger.logger.info(`Call ${this.callId} onNegotiateReceived() ignoring colliding negotiate event because we're impolite`);
      return;
    }
    const prevLocalOnHold = this.isLocalOnHold();
    const sdpStreamMetadata = content[_callEventTypes.SDPStreamMetadataKey];
    if (sdpStreamMetadata) {
      this.updateRemoteSDPStreamMetadata(sdpStreamMetadata);
    } else {
      _logger.logger.warn(`Call ${this.callId} onNegotiateReceived() received negotiation event without SDPStreamMetadata!`);
    }
    try {
      await this.peerConn.setRemoteDescription(description);
      if (description.type === "offer") {
        var _localDescription;
        let answer;
        try {
          this.getRidOfRTXCodecs();
          answer = await this.createAnswer();
        } catch (err) {
          _logger.logger.debug(`Call ${this.callId} onNegotiateReceived() failed to create answer: `, err);
          this.terminate(CallParty.Local, CallErrorCode.CreateAnswer, true);
          return;
        }
        await this.peerConn.setLocalDescription(answer);
        this.sendVoipEvent(_event.EventType.CallNegotiate, {
          description: (_localDescription = this.peerConn.localDescription) === null || _localDescription === void 0 ? void 0 : _localDescription.toJSON(),
          [_callEventTypes.SDPStreamMetadataKey]: this.getLocalSDPStreamMetadata(true)
        });
      }
    } catch (err) {
      _logger.logger.warn(`Call ${this.callId} onNegotiateReceived() failed to complete negotiation`, err);
    }
    const newLocalOnHold = this.isLocalOnHold();
    if (prevLocalOnHold !== newLocalOnHold) {
      this.emit(CallEvent.LocalHoldUnhold, newLocalOnHold, this);
      // also this one for backwards compat
      this.emit(CallEvent.HoldUnhold, newLocalOnHold);
    }
  }
  updateRemoteSDPStreamMetadata(metadata) {
    this.remoteSDPStreamMetadata = utils.recursivelyAssign(this.remoteSDPStreamMetadata || {}, metadata, true);
    for (const feed of this.getRemoteFeeds()) {
      var _streamId;
      const streamId = feed.stream.id;
      const metadata = this.remoteSDPStreamMetadata[streamId];
      feed.setAudioVideoMuted(metadata === null || metadata === void 0 ? void 0 : metadata.audio_muted, metadata === null || metadata === void 0 ? void 0 : metadata.video_muted);
      feed.purpose = (_streamId = this.remoteSDPStreamMetadata[streamId]) === null || _streamId === void 0 ? void 0 : _streamId.purpose;
    }
  }
  onSDPStreamMetadataChangedReceived(event) {
    const content = event.getContent();
    const metadata = content[_callEventTypes.SDPStreamMetadataKey];
    this.updateRemoteSDPStreamMetadata(metadata);
  }
  async onAssertedIdentityReceived(event) {
    const content = event.getContent();
    if (!content.asserted_identity) return;
    this.remoteAssertedIdentity = {
      id: content.asserted_identity.id,
      displayName: content.asserted_identity.display_name
    };
    this.emit(CallEvent.AssertedIdentityChanged, this);
  }
  callHasEnded() {
    // This exists as workaround to typescript trying to be clever and erroring
    // when putting if (this.state === CallState.Ended) return; twice in the same
    // function, even though that function is async.
    return this.state === CallState.Ended;
  }
  queueGotLocalOffer() {
    // Ensure only one negotiate/answer event is being processed at a time.
    if (this.responsePromiseChain) {
      this.responsePromiseChain = this.responsePromiseChain.then(() => this.wrappedGotLocalOffer());
    } else {
      this.responsePromiseChain = this.wrappedGotLocalOffer();
    }
  }
  async wrappedGotLocalOffer() {
    this.makingOffer = true;
    try {
      // XXX: in what situations do we believe gotLocalOffer actually throws? It appears
      // to handle most of its exceptions itself and terminate the call. I'm not entirely
      // sure it would ever throw, so I can't add a test for these lines.
      // Also the tense is different between "gotLocalOffer" and "getLocalOfferFailed" so
      // it's not entirely clear whether getLocalOfferFailed is just misnamed or whether
      // they've been cross-polinated somehow at some point.
      await this.gotLocalOffer();
    } catch (e) {
      this.getLocalOfferFailed(e);
      return;
    } finally {
      this.makingOffer = false;
    }
  }
  async gotLocalOffer() {
    _logger.logger.debug(`Call ${this.callId} gotLocalOffer() running`);
    if (this.callHasEnded()) {
      _logger.logger.debug(`Call ${this.callId} gotLocalOffer() ignoring newly created offer because the call has ended"`);
      return;
    }
    let offer;
    try {
      this.getRidOfRTXCodecs();
      offer = await this.createOffer();
    } catch (err) {
      _logger.logger.debug(`Call ${this.callId} gotLocalOffer() failed to create offer: `, err);
      this.terminate(CallParty.Local, CallErrorCode.CreateOffer, true);
      return;
    }
    try {
      await this.peerConn.setLocalDescription(offer);
    } catch (err) {
      _logger.logger.debug(`Call ${this.callId} gotLocalOffer() error setting local description!`, err);
      this.terminate(CallParty.Local, CallErrorCode.SetLocalDescription, true);
      return;
    }
    if (this.peerConn.iceGatheringState === "gathering") {
      // Allow a short time for initial candidates to be gathered
      await new Promise(resolve => {
        setTimeout(resolve, 200);
      });
    }
    if (this.callHasEnded()) return;
    const eventType = this.state === CallState.CreateOffer ? _event.EventType.CallInvite : _event.EventType.CallNegotiate;
    const content = {
      lifetime: CALL_TIMEOUT_MS
    };
    if (eventType === _event.EventType.CallInvite && this.invitee) {
      content.invitee = this.invitee;
    }

    // clunky because TypeScript can't follow the types through if we use an expression as the key
    if (this.state === CallState.CreateOffer) {
      var _localDescription2;
      content.offer = (_localDescription2 = this.peerConn.localDescription) === null || _localDescription2 === void 0 ? void 0 : _localDescription2.toJSON();
    } else {
      var _localDescription3;
      content.description = (_localDescription3 = this.peerConn.localDescription) === null || _localDescription3 === void 0 ? void 0 : _localDescription3.toJSON();
    }
    content.capabilities = {
      "m.call.transferee": this.client.supportsCallTransfer,
      "m.call.dtmf": false
    };
    content[_callEventTypes.SDPStreamMetadataKey] = this.getLocalSDPStreamMetadata(true);

    // Get rid of any candidates waiting to be sent: they'll be included in the local
    // description we just got and will send in the offer.
    const discardCount = this.discardDuplicateCandidates();
    _logger.logger.info(`Call ${this.callId} gotLocalOffer() discarding ${discardCount} candidates that will be sent in offer`);
    try {
      await this.sendVoipEvent(eventType, content);
    } catch (error) {
      _logger.logger.error(`Call ${this.callId} gotLocalOffer() failed to send invite`, error);
      if (error instanceof _httpApi.MatrixError && error.event) this.client.cancelPendingEvent(error.event);
      let code = CallErrorCode.SignallingFailed;
      let message = "Signalling failed";
      if (this.state === CallState.CreateOffer) {
        code = CallErrorCode.SendInvite;
        message = "Failed to send invite";
      }
      if (error.name == "UnknownDeviceError") {
        code = CallErrorCode.UnknownDevices;
        message = "Unknown devices present in the room";
      }
      this.emit(CallEvent.Error, new CallError(code, message, error), this);
      this.terminate(CallParty.Local, code, false);

      // no need to carry on & send the candidate queue, but we also
      // don't want to rethrow the error
      return;
    }
    this.sendCandidateQueue();
    if (this.state === CallState.CreateOffer) {
      this.inviteOrAnswerSent = true;
      this.state = CallState.InviteSent;
      this.inviteTimeout = setTimeout(() => {
        this.inviteTimeout = undefined;
        if (this.state === CallState.InviteSent) {
          this.hangup(CallErrorCode.InviteTimeout, false);
        }
      }, CALL_TIMEOUT_MS);
    }
  }
  /**
   * This method removes all video/rtx codecs from screensharing video
   * transceivers. This is necessary since they can cause problems. Without
   * this the following steps should produce an error:
   *   Chromium calls Firefox
   *   Firefox answers
   *   Firefox starts screen-sharing
   *   Chromium starts screen-sharing
   *   Call crashes for Chromium with:
   *       [96685:23:0518/162603.933321:ERROR:webrtc_video_engine.cc(3296)] RTX codec (PT=97) mapped to PT=96 which is not in the codec list.
   *       [96685:23:0518/162603.933377:ERROR:webrtc_video_engine.cc(1171)] GetChangedRecvParameters called without any video codecs.
   *       [96685:23:0518/162603.933430:ERROR:sdp_offer_answer.cc(4302)] Failed to set local video description recv parameters for m-section with mid='2'. (INVALID_PARAMETER)
   */
  getRidOfRTXCodecs() {
    // RTCRtpReceiver.getCapabilities and RTCRtpSender.getCapabilities don't seem to be supported on FF
    if (!RTCRtpReceiver.getCapabilities || !RTCRtpSender.getCapabilities) return;
    const recvCodecs = RTCRtpReceiver.getCapabilities("video").codecs;
    const sendCodecs = RTCRtpSender.getCapabilities("video").codecs;
    const codecs = [...sendCodecs, ...recvCodecs];
    for (const codec of codecs) {
      if (codec.mimeType === "video/rtx") {
        const rtxCodecIndex = codecs.indexOf(codec);
        codecs.splice(rtxCodecIndex, 1);
      }
    }
    const screenshareVideoTransceiver = this.transceivers.get(getTransceiverKey(_callEventTypes.SDPStreamMetadataPurpose.Screenshare, "video"));
    if (screenshareVideoTransceiver) screenshareVideoTransceiver.setCodecPreferences(codecs);
  }
  /**
   * @internal
   */
  async sendVoipEvent(eventType, content) {
    const realContent = Object.assign({}, content, {
      version: VOIP_PROTO_VERSION,
      call_id: this.callId,
      party_id: this.ourPartyId,
      conf_id: this.groupCallId
    });
    if (this.opponentDeviceId) {
      var _this$getOpponentMemb2;
      const toDeviceSeq = this.toDeviceSeq++;
      const content = _objectSpread(_objectSpread({}, realContent), {}, {
        device_id: this.client.deviceId,
        sender_session_id: this.client.getSessionId(),
        dest_session_id: this.opponentSessionId,
        seq: toDeviceSeq,
        [_event.ToDeviceMessageId]: (0, _uuid.v4)()
      });
      this.emit(CallEvent.SendVoipEvent, {
        type: "toDevice",
        eventType,
        userId: this.invitee || ((_this$getOpponentMemb2 = this.getOpponentMember()) === null || _this$getOpponentMemb2 === void 0 ? void 0 : _this$getOpponentMemb2.userId),
        opponentDeviceId: this.opponentDeviceId,
        content
      }, this);
      const userId = this.invitee || this.getOpponentMember().userId;
      if (this.client.getUseE2eForGroupCall()) {
        if (!this.opponentDeviceInfo) {
          _logger.logger.warn(`Call ${this.callId} sendVoipEvent() failed: we do not have opponentDeviceInfo`);
          return;
        }
        await this.client.encryptAndSendToDevices([{
          userId,
          deviceInfo: this.opponentDeviceInfo
        }], {
          type: eventType,
          content
        });
      } else {
        await this.client.sendToDevice(eventType, new Map([[userId, new Map([[this.opponentDeviceId, content]])]]));
      }
    } else {
      var _this$getOpponentMemb3;
      this.emit(CallEvent.SendVoipEvent, {
        type: "sendEvent",
        eventType,
        roomId: this.roomId,
        content: realContent,
        userId: this.invitee || ((_this$getOpponentMemb3 = this.getOpponentMember()) === null || _this$getOpponentMemb3 === void 0 ? void 0 : _this$getOpponentMemb3.userId)
      }, this);
      await this.client.sendEvent(this.roomId, eventType, realContent);
    }
  }

  /**
   * Queue a candidate to be sent
   * @param content - The candidate to queue up, or null if candidates have finished being generated
   *                and end-of-candidates should be signalled
   */
  queueCandidate(content) {
    // We partially de-trickle candidates by waiting for `delay` before sending them
    // amalgamated, in order to avoid sending too many m.call.candidates events and hitting
    // rate limits in Matrix.
    // In practice, it'd be better to remove rate limits for m.call.*

    // N.B. this deliberately lets you queue and send blank candidates, which MSC2746
    // currently proposes as the way to indicate that candidate gathering is complete.
    // This will hopefully be changed to an explicit rather than implicit notification
    // shortly.
    if (content) {
      this.candidateSendQueue.push(content);
    } else {
      this.candidatesEnded = true;
    }

    // Don't send the ICE candidates yet if the call is in the ringing state: this
    // means we tried to pick (ie. started generating candidates) and then failed to
    // send the answer and went back to the ringing state. Queue up the candidates
    // to send if we successfully send the answer.
    // Equally don't send if we haven't yet sent the answer because we can send the
    // first batch of candidates along with the answer
    if (this.state === CallState.Ringing || !this.inviteOrAnswerSent) return;

    // MSC2746 recommends these values (can be quite long when calling because the
    // callee will need a while to answer the call)
    const delay = this.direction === CallDirection.Inbound ? 500 : 2000;
    if (this.candidateSendTries === 0) {
      setTimeout(() => {
        this.sendCandidateQueue();
      }, delay);
    }
  }

  // Discard all non-end-of-candidates messages
  // Return the number of candidate messages that were discarded.
  // Call this method before sending an invite or answer message
  discardDuplicateCandidates() {
    let discardCount = 0;
    const newQueue = [];
    for (let i = 0; i < this.candidateSendQueue.length; i++) {
      const candidate = this.candidateSendQueue[i];
      if (candidate.candidate === "") {
        newQueue.push(candidate);
      } else {
        discardCount++;
      }
    }
    this.candidateSendQueue = newQueue;
    return discardCount;
  }

  /*
   * Transfers this call to another user
   */
  async transfer(targetUserId) {
    // Fetch the target user's global profile info: their room avatar / displayname
    // could be different in whatever room we share with them.
    const profileInfo = await this.client.getProfileInfo(targetUserId);
    const replacementId = genCallID();
    const body = {
      replacement_id: genCallID(),
      target_user: {
        id: targetUserId,
        display_name: profileInfo.displayname,
        avatar_url: profileInfo.avatar_url
      },
      create_call: replacementId
    };
    await this.sendVoipEvent(_event.EventType.CallReplaces, body);
    await this.terminate(CallParty.Local, CallErrorCode.Transferred, true);
  }

  /*
   * Transfers this call to the target call, effectively 'joining' the
   * two calls (so the remote parties on each call are connected together).
   */
  async transferToCall(transferTargetCall) {
    var _transferTargetCall$g, _this$getOpponentMemb4;
    const targetUserId = (_transferTargetCall$g = transferTargetCall.getOpponentMember()) === null || _transferTargetCall$g === void 0 ? void 0 : _transferTargetCall$g.userId;
    const targetProfileInfo = targetUserId ? await this.client.getProfileInfo(targetUserId) : undefined;
    const opponentUserId = (_this$getOpponentMemb4 = this.getOpponentMember()) === null || _this$getOpponentMemb4 === void 0 ? void 0 : _this$getOpponentMemb4.userId;
    const transfereeProfileInfo = opponentUserId ? await this.client.getProfileInfo(opponentUserId) : undefined;
    const newCallId = genCallID();
    const bodyToTransferTarget = {
      // the replacements on each side have their own ID, and it's distinct from the
      // ID of the new call (but we can use the same function to generate it)
      replacement_id: genCallID(),
      target_user: {
        id: opponentUserId,
        display_name: transfereeProfileInfo === null || transfereeProfileInfo === void 0 ? void 0 : transfereeProfileInfo.displayname,
        avatar_url: transfereeProfileInfo === null || transfereeProfileInfo === void 0 ? void 0 : transfereeProfileInfo.avatar_url
      },
      await_call: newCallId
    };
    await transferTargetCall.sendVoipEvent(_event.EventType.CallReplaces, bodyToTransferTarget);
    const bodyToTransferee = {
      replacement_id: genCallID(),
      target_user: {
        id: targetUserId,
        display_name: targetProfileInfo === null || targetProfileInfo === void 0 ? void 0 : targetProfileInfo.displayname,
        avatar_url: targetProfileInfo === null || targetProfileInfo === void 0 ? void 0 : targetProfileInfo.avatar_url
      },
      create_call: newCallId
    };
    await this.sendVoipEvent(_event.EventType.CallReplaces, bodyToTransferee);
    await this.terminate(CallParty.Local, CallErrorCode.Transferred, true);
    await transferTargetCall.terminate(CallParty.Local, CallErrorCode.Transferred, true);
  }
  async terminate(hangupParty, hangupReason, shouldEmit) {
    var _this$stats2;
    if (this.callHasEnded()) return;
    this.hangupParty = hangupParty;
    this.hangupReason = hangupReason;
    this.state = CallState.Ended;
    if (this.inviteTimeout) {
      clearTimeout(this.inviteTimeout);
      this.inviteTimeout = undefined;
    }
    if (this.iceDisconnectedTimeout !== undefined) {
      clearTimeout(this.iceDisconnectedTimeout);
      this.iceDisconnectedTimeout = undefined;
    }
    if (this.callLengthInterval) {
      clearInterval(this.callLengthInterval);
      this.callLengthInterval = undefined;
    }
    if (this.stopVideoTrackTimer !== undefined) {
      clearTimeout(this.stopVideoTrackTimer);
      this.stopVideoTrackTimer = undefined;
    }
    for (const [stream, listener] of this.removeTrackListeners) {
      stream.removeEventListener("removetrack", listener);
    }
    this.removeTrackListeners.clear();
    this.callStatsAtEnd = await this.collectCallStats();

    // Order is important here: first we stopAllMedia() and only then we can deleteAllFeeds()
    this.stopAllMedia();
    this.deleteAllFeeds();
    if (this.peerConn && this.peerConn.signalingState !== "closed") {
      this.peerConn.close();
    }
    (_this$stats2 = this.stats) === null || _this$stats2 === void 0 ? void 0 : _this$stats2.removeStatsReportGatherer(this.callId);
    if (shouldEmit) {
      this.emit(CallEvent.Hangup, this);
    }
    this.client.callEventHandler.calls.delete(this.callId);
  }
  stopAllMedia() {
    _logger.logger.debug(`Call ${this.callId} stopAllMedia() running`);
    for (const feed of this.feeds) {
      // Slightly awkward as local feed need to go via the correct method on
      // the MediaHandler so they get removed from MediaHandler (remote tracks
      // don't)
      // NB. We clone local streams when passing them to individual calls in a group
      // call, so we can (and should) stop the clones once we no longer need them:
      // the other clones will continue fine.
      if (feed.isLocal() && feed.purpose === _callEventTypes.SDPStreamMetadataPurpose.Usermedia) {
        this.client.getMediaHandler().stopUserMediaStream(feed.stream);
      } else if (feed.isLocal() && feed.purpose === _callEventTypes.SDPStreamMetadataPurpose.Screenshare) {
        this.client.getMediaHandler().stopScreensharingStream(feed.stream);
      } else if (!feed.isLocal()) {
        _logger.logger.debug(`Call ${this.callId} stopAllMedia() stopping stream (streamId=${feed.stream.id})`);
        for (const track of feed.stream.getTracks()) {
          track.stop();
        }
      }
    }
  }
  checkForErrorListener() {
    if (this.listeners(_typedEventEmitter.EventEmitterEvents.Error).length === 0) {
      throw new Error("You MUST attach an error listener using call.on('error', function() {})");
    }
  }
  async sendCandidateQueue() {
    if (this.candidateSendQueue.length === 0 || this.callHasEnded()) {
      return;
    }
    const candidates = this.candidateSendQueue;
    this.candidateSendQueue = [];
    ++this.candidateSendTries;
    const content = {
      candidates: candidates.map(candidate => candidate.toJSON())
    };
    if (this.candidatesEnded) {
      // If there are no more candidates, signal this by adding an empty string candidate
      content.candidates.push({
        candidate: ""
      });
    }
    _logger.logger.debug(`Call ${this.callId} sendCandidateQueue() attempting to send ${candidates.length} candidates`);
    try {
      await this.sendVoipEvent(_event.EventType.CallCandidates, content);
      // reset our retry count if we have successfully sent our candidates
      // otherwise queueCandidate() will refuse to try to flush the queue
      this.candidateSendTries = 0;

      // Try to send candidates again just in case we received more candidates while sending.
      this.sendCandidateQueue();
    } catch (error) {
      // don't retry this event: we'll send another one later as we might
      // have more candidates by then.
      if (error instanceof _httpApi.MatrixError && error.event) this.client.cancelPendingEvent(error.event);

      // put all the candidates we failed to send back in the queue
      this.candidateSendQueue.push(...candidates);
      if (this.candidateSendTries > 5) {
        _logger.logger.debug(`Call ${this.callId} sendCandidateQueue() failed to send candidates on attempt ${this.candidateSendTries}. Giving up on this call.`, error);
        const code = CallErrorCode.SignallingFailed;
        const message = "Signalling failed";
        this.emit(CallEvent.Error, new CallError(code, message, error), this);
        this.hangup(code, false);
        return;
      }
      const delayMs = 500 * Math.pow(2, this.candidateSendTries);
      ++this.candidateSendTries;
      _logger.logger.debug(`Call ${this.callId} sendCandidateQueue() failed to send candidates. Retrying in ${delayMs}ms`, error);
      setTimeout(() => {
        this.sendCandidateQueue();
      }, delayMs);
    }
  }

  /**
   * Place a call to this room.
   * @throws if you have not specified a listener for 'error' events.
   * @throws if have passed audio=false.
   */
  async placeCall(audio, video) {
    if (!audio) {
      throw new Error("You CANNOT start a call without audio");
    }
    this.state = CallState.WaitLocalMedia;
    try {
      var _this$client$getDevic2;
      const stream = await this.client.getMediaHandler().getUserMediaStream(audio, video);

      // make sure all the tracks are enabled (same as pushNewLocalFeed -
      // we probably ought to just have one code path for adding streams)
      setTracksEnabled(stream.getAudioTracks(), true);
      setTracksEnabled(stream.getVideoTracks(), true);
      const callFeed = new _callFeed.CallFeed({
        client: this.client,
        roomId: this.roomId,
        userId: this.client.getUserId(),
        deviceId: (_this$client$getDevic2 = this.client.getDeviceId()) !== null && _this$client$getDevic2 !== void 0 ? _this$client$getDevic2 : undefined,
        stream,
        purpose: _callEventTypes.SDPStreamMetadataPurpose.Usermedia,
        audioMuted: false,
        videoMuted: false
      });
      await this.placeCallWithCallFeeds([callFeed]);
    } catch (e) {
      this.getUserMediaFailed(e);
      return;
    }
  }

  /**
   * Place a call to this room with call feed.
   * @param callFeeds - to use
   * @throws if you have not specified a listener for 'error' events.
   * @throws if have passed audio=false.
   */
  async placeCallWithCallFeeds(callFeeds, requestScreenshareFeed = false) {
    this.checkForErrorListener();
    this.direction = CallDirection.Outbound;
    await this.initOpponentCrypto();

    // XXX Find a better way to do this
    this.client.callEventHandler.calls.set(this.callId, this);

    // make sure we have valid turn creds. Unless something's gone wrong, it should
    // poll and keep the credentials valid so this should be instant.
    const haveTurnCreds = await this.client.checkTurnServers();
    if (!haveTurnCreds) {
      _logger.logger.warn(`Call ${this.callId} placeCallWithCallFeeds() failed to get TURN credentials! Proceeding with call anyway...`);
    }

    // create the peer connection now so it can be gathering candidates while we get user
    // media (assuming a candidate pool size is configured)
    this.peerConn = this.createPeerConnection();
    this.gotCallFeedsForInvite(callFeeds, requestScreenshareFeed);
  }
  createPeerConnection() {
    var _this$stats3;
    const pc = new window.RTCPeerConnection({
      iceTransportPolicy: this.forceTURN ? "relay" : undefined,
      iceServers: this.turnServers,
      iceCandidatePoolSize: this.client.iceCandidatePoolSize,
      bundlePolicy: "max-bundle"
    });

    // 'connectionstatechange' would be better, but firefox doesn't implement that.
    pc.addEventListener("iceconnectionstatechange", this.onIceConnectionStateChanged);
    pc.addEventListener("signalingstatechange", this.onSignallingStateChanged);
    pc.addEventListener("icecandidate", this.gotLocalIceCandidate);
    pc.addEventListener("icegatheringstatechange", this.onIceGatheringStateChange);
    pc.addEventListener("track", this.onTrack);
    pc.addEventListener("negotiationneeded", this.onNegotiationNeeded);
    pc.addEventListener("datachannel", this.onDataChannel);
    (_this$stats3 = this.stats) === null || _this$stats3 === void 0 ? void 0 : _this$stats3.addStatsReportGatherer(this.callId, "unknown", pc);
    return pc;
  }
  partyIdMatches(msg) {
    // They must either match or both be absent (in which case opponentPartyId will be null)
    // Also we ignore party IDs on the invite/offer if the version is 0, so we must do the same
    // here and use null if the version is 0 (woe betide any opponent sending messages in the
    // same call with different versions)
    const msgPartyId = msg.version === 0 ? null : msg.party_id || null;
    return msgPartyId === this.opponentPartyId;
  }

  // Commits to an opponent for the call
  // ev: An invite or answer event
  chooseOpponent(ev) {
    var _getMember;
    // I choo-choo-choose you
    const msg = ev.getContent();
    _logger.logger.debug(`Call ${this.callId} chooseOpponent() running (partyId=${msg.party_id})`);
    this.opponentVersion = msg.version;
    if (this.opponentVersion === 0) {
      // set to null to indicate that we've chosen an opponent, but because
      // they're v0 they have no party ID (even if they sent one, we're ignoring it)
      this.opponentPartyId = null;
    } else {
      // set to their party ID, or if they're naughty and didn't send one despite
      // not being v0, set it to null to indicate we picked an opponent with no
      // party ID
      this.opponentPartyId = msg.party_id || null;
    }
    this.opponentCaps = msg.capabilities || {};
    this.opponentMember = (_getMember = this.client.getRoom(this.roomId).getMember(ev.getSender())) !== null && _getMember !== void 0 ? _getMember : undefined;
  }
  async addBufferedIceCandidates() {
    const bufferedCandidates = this.remoteCandidateBuffer.get(this.opponentPartyId);
    if (bufferedCandidates) {
      _logger.logger.info(`Call ${this.callId} addBufferedIceCandidates() adding ${bufferedCandidates.length} buffered candidates for opponent ${this.opponentPartyId}`);
      await this.addIceCandidates(bufferedCandidates);
    }
    this.remoteCandidateBuffer.clear();
  }
  async addIceCandidates(candidates) {
    for (const candidate of candidates) {
      if ((candidate.sdpMid === null || candidate.sdpMid === undefined) && (candidate.sdpMLineIndex === null || candidate.sdpMLineIndex === undefined)) {
        _logger.logger.debug(`Call ${this.callId} addIceCandidates() got remote ICE end-of-candidates`);
      } else {
        _logger.logger.debug(`Call ${this.callId} addIceCandidates() got remote ICE candidate (sdpMid=${candidate.sdpMid}, candidate=${candidate.candidate})`);
      }
      try {
        await this.peerConn.addIceCandidate(candidate);
      } catch (err) {
        if (!this.ignoreOffer) {
          _logger.logger.info(`Call ${this.callId} addIceCandidates() failed to add remote ICE candidate`, err);
        }
      }
    }
  }
  get hasPeerConnection() {
    return Boolean(this.peerConn);
  }
  initStats(stats, peerId = "unknown") {
    this.stats = stats;
    this.stats.start();
  }
}
exports.MatrixCall = MatrixCall;
function setTracksEnabled(tracks, enabled) {
  for (const track of tracks) {
    track.enabled = enabled;
  }
}
function supportsMatrixCall() {
  // typeof prevents Node from erroring on an undefined reference
  if (typeof window === "undefined" || typeof document === "undefined") {
    // NB. We don't log here as apps try to create a call object as a test for
    // whether calls are supported, so we shouldn't fill the logs up.
    return false;
  }

  // Firefox throws on so little as accessing the RTCPeerConnection when operating in a secure mode.
  // There's some information at https://bugzilla.mozilla.org/show_bug.cgi?id=1542616 though the concern
  // is that the browser throwing a SecurityError will brick the client creation process.
  try {
    const supported = Boolean(window.RTCPeerConnection || window.RTCSessionDescription || window.RTCIceCandidate || navigator.mediaDevices);
    if (!supported) {
      /* istanbul ignore if */ // Adds a lot of noise to test runs, so disable logging there.
      if (process.env.NODE_ENV !== "test") {
        _logger.logger.error("WebRTC is not supported in this browser / environment");
      }
      return false;
    }
  } catch (e) {
    _logger.logger.error("Exception thrown when trying to access WebRTC", e);
    return false;
  }
  return true;
}

/**
 * DEPRECATED
 * Use client.createCall()
 *
 * Create a new Matrix call for the browser.
 * @param client - The client instance to use.
 * @param roomId - The room the call is in.
 * @param options - DEPRECATED optional options map.
 * @returns the call or null if the browser doesn't support calling.
 */
function createNewMatrixCall(client, roomId, options) {
  if (!supportsMatrixCall()) return null;
  const optionsForceTURN = options ? options.forceTURN : false;
  const opts = {
    client: client,
    roomId: roomId,
    invitee: options === null || options === void 0 ? void 0 : options.invitee,
    turnServers: client.getTurnServers(),
    // call level options
    forceTURN: client.forceTURN || optionsForceTURN,
    opponentDeviceId: options === null || options === void 0 ? void 0 : options.opponentDeviceId,
    opponentSessionId: options === null || options === void 0 ? void 0 : options.opponentSessionId,
    groupCallId: options === null || options === void 0 ? void 0 : options.groupCallId
  };
  const call = new MatrixCall(opts);
  client.reEmitter.reEmit(call, Object.values(CallEvent));
  return call;
}
//# sourceMappingURL=call.js.map