@unwanted/matrix-sdk-mini 34.12.0-1 → 34.12.0-2

package/lib/webrtc/audioContext.js

@@ -1,46 +0,0 @@
-/*
-Copyright 2022 The Matrix.org Foundation C.I.C.
-
-Licensed under the Apache License, Version 2.0 (the "License");
-you may not use this file except in compliance with the License.
-You may obtain a copy of the License at
-
-    http://www.apache.org/licenses/LICENSE-2.0
-
-Unless required by applicable law or agreed to in writing, software
-distributed under the License is distributed on an "AS IS" BASIS,
-WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-See the License for the specific language governing permissions and
-limitations under the License.
-*/
-
-var audioContext = null;
-var refCount = 0;
-
-/**
- * Acquires a reference to the shared AudioContext.
- * It's highly recommended to reuse this AudioContext rather than creating your
- * own, because multiple AudioContexts can be problematic in some browsers.
- * Make sure to call releaseContext when you're done using it.
- * @returns The shared AudioContext
- */
-export var acquireContext = () => {
-  if (audioContext === null) audioContext = new AudioContext();
-  refCount++;
-  return audioContext;
-};
-
-/**
- * Signals that one of the references to the shared AudioContext has been
- * released, allowing the context and associated hardware resources to be
- * cleaned up if nothing else is using it.
- */
-export var releaseContext = () => {
-  refCount--;
-  if (refCount === 0) {
-    var _audioContext;
-    (_audioContext = audioContext) === null || _audioContext === void 0 || _audioContext.close();
-    audioContext = null;
-  }
-};
-//# sourceMappingURL=audioContext.js.map

package/lib/webrtc/audioContext.js.map

@@ -1 +0,0 @@
-{"version":3,"file":"audioContext.js","names":["audioContext","refCount","acquireContext","AudioContext","releaseContext","_audioContext","close"],"sources":["../../src/webrtc/audioContext.ts"],"sourcesContent":["/*\nCopyright 2022 The Matrix.org Foundation C.I.C.\n\nLicensed under the Apache License, Version 2.0 (the \"License\");\nyou may not use this file except in compliance with the License.\nYou may obtain a copy of the License at\n\n    http://www.apache.org/licenses/LICENSE-2.0\n\nUnless required by applicable law or agreed to in writing, software\ndistributed under the License is distributed on an \"AS IS\" BASIS,\nWITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.\nSee the License for the specific language governing permissions and\nlimitations under the License.\n*/\n\nlet audioContext: AudioContext | null = null;\nlet refCount = 0;\n\n/**\n * Acquires a reference to the shared AudioContext.\n * It's highly recommended to reuse this AudioContext rather than creating your\n * own, because multiple AudioContexts can be problematic in some browsers.\n * Make sure to call releaseContext when you're done using it.\n * @returns The shared AudioContext\n */\nexport const acquireContext = (): AudioContext => {\n    if (audioContext === null) audioContext = new AudioContext();\n    refCount++;\n    return audioContext;\n};\n\n/**\n * Signals that one of the references to the shared AudioContext has been\n * released, allowing the context and associated hardware resources to be\n * cleaned up if nothing else is using it.\n */\nexport const releaseContext = (): void => {\n    refCount--;\n    if (refCount === 0) {\n        audioContext?.close();\n        audioContext = null;\n    }\n};\n"],"mappings":"AAAA;AACA;AACA;AACA;AACA;AACA;AACA;AACA;AACA;AACA;AACA;AACA;AACA;AACA;AACA;;AAEA,IAAIA,YAAiC,GAAG,IAAI;AAC5C,IAAIC,QAAQ,GAAG,CAAC;;AAEhB;AACA;AACA;AACA;AACA;AACA;AACA;AACA,OAAO,IAAMC,cAAc,GAAGA,CAAA,KAAoB;EAC9C,IAAIF,YAAY,KAAK,IAAI,EAAEA,YAAY,GAAG,IAAIG,YAAY,CAAC,CAAC;EAC5DF,QAAQ,EAAE;EACV,OAAOD,YAAY;AACvB,CAAC;;AAED;AACA;AACA;AACA;AACA;AACA,OAAO,IAAMI,cAAc,GAAGA,CAAA,KAAY;EACtCH,QAAQ,EAAE;EACV,IAAIA,QAAQ,KAAK,CAAC,EAAE;IAAA,IAAAI,aAAA;IAChB,CAAAA,aAAA,GAAAL,YAAY,cAAAK,aAAA,eAAZA,aAAA,CAAcC,KAAK,CAAC,CAAC;IACrBN,YAAY,GAAG,IAAI;EACvB;AACJ,CAAC","ignoreList":[]}

package/lib/webrtc/call.d.ts

@@ -1,560 +0,0 @@
-import { MatrixEvent } from "../models/event.ts";
-import { EventType, TimelineEvents } from "../@types/event.ts";
-import { RoomMember } from "../models/room-member.ts";
-import { MCallAnswer, MCallHangupReject } from "./callEventTypes.ts";
-import { CallFeed } from "./callFeed.ts";
-import { MatrixClient } from "../client.ts";
-import { TypedEventEmitter } from "../models/typed-event-emitter.ts";
-import { IScreensharingOpts } from "./mediaHandler.ts";
-import { GroupCallStats } from "./stats/groupCallStats.ts";
-interface CallOpts {
-    roomId: string;
-    invitee?: string;
-    client: MatrixClient;
-    /**
-     * Whether relay through TURN should be forced.
-     * @deprecated use opts.forceTURN when creating the matrix client
-     * since it's only possible to set this option on outbound calls.
-     */
-    forceTURN?: boolean;
-    turnServers?: Array<TurnServer>;
-    opponentDeviceId?: string;
-    opponentSessionId?: string;
-    groupCallId?: string;
-}
-interface TurnServer {
-    urls: Array<string>;
-    username?: string;
-    password?: string;
-    ttl?: number;
-}
-interface AssertedIdentity {
-    id: string;
-    displayName: string;
-}
-export declare enum CallState {
-    Fledgling = "fledgling",
-    InviteSent = "invite_sent",
-    WaitLocalMedia = "wait_local_media",
-    CreateOffer = "create_offer",
-    CreateAnswer = "create_answer",
-    Connecting = "connecting",
-    Connected = "connected",
-    Ringing = "ringing",
-    Ended = "ended"
-}
-export declare enum CallType {
-    Voice = "voice",
-    Video = "video"
-}
-export declare enum CallDirection {
-    Inbound = "inbound",
-    Outbound = "outbound"
-}
-export declare enum CallParty {
-    Local = "local",
-    Remote = "remote"
-}
-export declare enum CallEvent {
-    Hangup = "hangup",
-    State = "state",
-    Error = "error",
-    Replaced = "replaced",
-    LocalHoldUnhold = "local_hold_unhold",
-    RemoteHoldUnhold = "remote_hold_unhold",
-    HoldUnhold = "hold_unhold",
-    FeedsChanged = "feeds_changed",
-    AssertedIdentityChanged = "asserted_identity_changed",
-    LengthChanged = "length_changed",
-    DataChannel = "datachannel",
-    SendVoipEvent = "send_voip_event",
-    PeerConnectionCreated = "peer_connection_created"
-}
-export declare enum CallErrorCode {
-    /** The user chose to end the call */
-    UserHangup = "user_hangup",
-    /** An error code when the local client failed to create an offer. */
-    LocalOfferFailed = "local_offer_failed",
-    /**
-     * An error code when there is no local mic/camera to use. This may be because
-     * the hardware isn't plugged in, or the user has explicitly denied access.
-     */
-    NoUserMedia = "no_user_media",
-    /**
-     * Error code used when a call event failed to send
-     * because unknown devices were present in the room
-     */
-    UnknownDevices = "unknown_devices",
-    /**
-     * Error code used when we fail to send the invite
-     * for some reason other than there being unknown devices
-     */
-    SendInvite = "send_invite",
-    /**
-     * An answer could not be created
-     */
-    CreateAnswer = "create_answer",
-    /**
-     * An offer could not be created
-     */
-    CreateOffer = "create_offer",
-    /**
-     * Error code used when we fail to send the answer
-     * for some reason other than there being unknown devices
-     */
-    SendAnswer = "send_answer",
-    /**
-     * The session description from the other side could not be set
-     */
-    SetRemoteDescription = "set_remote_description",
-    /**
-     * The session description from this side could not be set
-     */
-    SetLocalDescription = "set_local_description",
-    /**
-     * A different device answered the call
-     */
-    AnsweredElsewhere = "answered_elsewhere",
-    /**
-     * No media connection could be established to the other party
-     */
-    IceFailed = "ice_failed",
-    /**
-     * The invite timed out whilst waiting for an answer
-     */
-    InviteTimeout = "invite_timeout",
-    /**
-     * The call was replaced by another call
-     */
-    Replaced = "replaced",
-    /**
-     * Signalling for the call could not be sent (other than the initial invite)
-     */
-    SignallingFailed = "signalling_timeout",
-    /**
-     * The remote party is busy
-     */
-    UserBusy = "user_busy",
-    /**
-     * We transferred the call off to somewhere else
-     */
-    Transferred = "transferred",
-    /**
-     * A call from the same user was found with a new session id
-     */
-    NewSession = "new_session"
-}
-/** The fallback ICE server to use for STUN or TURN protocols. */
-export declare const FALLBACK_ICE_SERVER = "stun:turn.matrix.org";
-export declare class CallError extends Error {
-    readonly code: string;
-    constructor(code: CallErrorCode, msg: string, err: Error);
-}
-export declare function genCallID(): string;
-type CallEventType = EventType.CallReplaces | EventType.CallAnswer | EventType.CallSelectAnswer | EventType.CallNegotiate | EventType.CallInvite | EventType.CallCandidates | EventType.CallHangup | EventType.CallReject | EventType.CallSDPStreamMetadataChangedPrefix;
-export interface VoipEvent {
-    type: "toDevice" | "sendEvent";
-    eventType: string;
-    userId?: string;
-    opponentDeviceId?: string;
-    roomId?: string;
-    content: TimelineEvents[CallEventType];
-}
-/**
- * These now all have the call object as an argument. Why? Well, to know which call a given event is
- * about you have three options:
- *  1. Use a closure as the callback that remembers what call it's listening to. This can be
- *     a pain because you need to pass the listener function again when you remove the listener,
- *     which might be somewhere else.
- *  2. Use not-very-well-known fact that EventEmitter sets 'this' to the emitter object in the
- *     callback. This doesn't really play well with modern Typescript and eslint and doesn't work
- *     with our pattern of re-emitting events.
- *  3. Pass the object in question as an argument to the callback.
- *
- * Now that we have group calls which have to deal with multiple call objects, this will
- * become more important, and I think methods 1 and 2 are just going to cause issues.
- */
-export type CallEventHandlerMap = {
-    [CallEvent.DataChannel]: (channel: RTCDataChannel, call: MatrixCall) => void;
-    [CallEvent.FeedsChanged]: (feeds: CallFeed[], call: MatrixCall) => void;
-    [CallEvent.Replaced]: (newCall: MatrixCall, oldCall: MatrixCall) => void;
-    [CallEvent.Error]: (error: CallError, call: MatrixCall) => void;
-    [CallEvent.RemoteHoldUnhold]: (onHold: boolean, call: MatrixCall) => void;
-    [CallEvent.LocalHoldUnhold]: (onHold: boolean, call: MatrixCall) => void;
-    [CallEvent.LengthChanged]: (length: number, call: MatrixCall) => void;
-    [CallEvent.State]: (state: CallState, oldState: CallState, call: MatrixCall) => void;
-    [CallEvent.Hangup]: (call: MatrixCall) => void;
-    [CallEvent.AssertedIdentityChanged]: (call: MatrixCall) => void;
-    [CallEvent.HoldUnhold]: (onHold: boolean) => void;
-    [CallEvent.SendVoipEvent]: (event: VoipEvent, call: MatrixCall) => void;
-    [CallEvent.PeerConnectionCreated]: (peerConn: RTCPeerConnection, call: MatrixCall) => void;
-};
-export declare class MatrixCall extends TypedEventEmitter<CallEvent, CallEventHandlerMap> {
-    roomId: string;
-    callId: string;
-    invitee?: string;
-    hangupParty?: CallParty;
-    hangupReason?: string;
-    direction?: CallDirection;
-    ourPartyId: string;
-    peerConn?: RTCPeerConnection;
-    toDeviceSeq: number;
-    isPtt: boolean;
-    private _state;
-    private readonly client;
-    private readonly forceTURN?;
-    private readonly turnServers;
-    private candidateSendQueue;
-    private candidateSendTries;
-    private candidatesEnded;
-    private feeds;
-    private transceivers;
-    private inviteOrAnswerSent;
-    private waitForLocalAVStream;
-    private successor?;
-    private opponentMember?;
-    private opponentVersion?;
-    private opponentPartyId;
-    private opponentCaps?;
-    private iceDisconnectedTimeout?;
-    private iceReconnectionTimeOut?;
-    private inviteTimeout?;
-    private readonly removeTrackListeners;
-    private remoteOnHold;
-    private callStatsAtEnd?;
-    private makingOffer;
-    private ignoreOffer;
-    private isSettingRemoteAnswerPending;
-    private responsePromiseChain?;
-    private remoteCandidateBuffer;
-    private remoteAssertedIdentity?;
-    private remoteSDPStreamMetadata?;
-    private callLengthInterval?;
-    private callStartTime?;
-    private opponentDeviceId?;
-    private opponentDeviceInfo?;
-    private opponentSessionId?;
-    groupCallId?: string;
-    private stopVideoTrackTimer?;
-    private readonly isOnlyDataChannelAllowed;
-    private stats;
-    /**
-     * Construct a new Matrix Call.
-     * @param opts - Config options.
-     */
-    constructor(opts: CallOpts);
-    /**
-     * Place a voice call to this room.
-     * @throws If you have not specified a listener for 'error' events.
-     */
-    placeVoiceCall(): Promise<void>;
-    /**
-     * Place a video call to this room.
-     * @throws If you have not specified a listener for 'error' events.
-     */
-    placeVideoCall(): Promise<void>;
-    /**
-     * 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: string, options: RTCDataChannelInit | undefined): RTCDataChannel;
-    getOpponentMember(): RoomMember | undefined;
-    getOpponentDeviceId(): string | undefined;
-    getOpponentSessionId(): string | undefined;
-    opponentCanBeTransferred(): boolean;
-    opponentSupportsDTMF(): boolean;
-    getRemoteAssertedIdentity(): AssertedIdentity | undefined;
-    get state(): CallState;
-    private set state(value);
-    get type(): CallType;
-    get hasLocalUserMediaVideoTrack(): boolean;
-    get hasRemoteUserMediaVideoTrack(): boolean;
-    get hasLocalUserMediaAudioTrack(): boolean;
-    get hasRemoteUserMediaAudioTrack(): boolean;
-    private get hasUserMediaAudioSender();
-    private get hasUserMediaVideoSender();
-    get localUsermediaFeed(): CallFeed | undefined;
-    get localScreensharingFeed(): CallFeed | undefined;
-    get localUsermediaStream(): MediaStream | undefined;
-    get localScreensharingStream(): MediaStream | undefined;
-    get remoteUsermediaFeed(): CallFeed | undefined;
-    get remoteScreensharingFeed(): CallFeed | undefined;
-    get remoteUsermediaStream(): MediaStream | undefined;
-    get remoteScreensharingStream(): MediaStream | undefined;
-    private getFeedByStreamId;
-    /**
-     * Returns an array of all CallFeeds
-     * @returns CallFeeds
-     */
-    getFeeds(): Array<CallFeed>;
-    /**
-     * Returns an array of all local CallFeeds
-     * @returns local CallFeeds
-     */
-    getLocalFeeds(): Array<CallFeed>;
-    /**
-     * Returns an array of all remote CallFeeds
-     * @returns remote CallFeeds
-     */
-    getRemoteFeeds(): Array<CallFeed>;
-    private initOpponentCrypto;
-    /**
-     * Generates and returns localSDPStreamMetadata
-     * @returns localSDPStreamMetadata
-     */
-    private getLocalSDPStreamMetadata;
-    /**
-     * Returns true if there are no incoming feeds,
-     * otherwise returns false
-     * @returns no incoming feeds
-     */
-    noIncomingFeeds(): boolean;
-    private pushRemoteFeed;
-    /**
-     * This method is used ONLY if the other client doesn't support sending SDPStreamMetadata
-     */
-    private pushRemoteFeedWithoutMetadata;
-    private pushNewLocalFeed;
-    /**
-     * Pushes supplied feed to the call
-     * @param callFeed - to push
-     * @param addToPeerConnection - whether to add the tracks to the peer connection
-     */
-    pushLocalFeed(callFeed: CallFeed, addToPeerConnection?: boolean): void;
-    /**
-     * Removes local call feed from the call and its tracks from the peer
-     * connection
-     * @param callFeed - to remove
-     */
-    removeLocalFeed(callFeed: CallFeed): void;
-    private deleteAllFeeds;
-    private deleteFeedByStream;
-    private deleteFeed;
-    getCurrentCallStats(): Promise<any[] | undefined>;
-    private collectCallStats;
-    /**
-     * Configure this call from an invite event. Used by MatrixClient.
-     * @param event - The m.call.invite event
-     */
-    initWithInvite(event: MatrixEvent): Promise<void>;
-    /**
-     * Configure this call from a hangup or reject event. Used by MatrixClient.
-     * @param event - The m.call.hangup event
-     */
-    initWithHangup(event: MatrixEvent): void;
-    private shouldAnswerWithMediaType;
-    /**
-     * Answer a call.
-     */
-    answer(audio?: boolean, video?: boolean): Promise<void>;
-    answerWithCallFeeds(callFeeds: CallFeed[]): void;
-    /**
-     * Replace this call with a new call, e.g. for glare resolution. Used by
-     * MatrixClient.
-     * @param newCall - The new call.
-     */
-    replacedBy(newCall: MatrixCall): void;
-    /**
-     * Hangup a call.
-     * @param reason - The reason why the call is being hung up.
-     * @param suppressEvent - True to suppress emitting an event.
-     */
-    hangup(reason: CallErrorCode, suppressEvent: boolean): void;
-    /**
-     * Reject a call
-     * This used to be done by calling hangup, but is a separate method and protocol
-     * event as of MSC2746.
-     */
-    reject(): void;
-    /**
-     * Adds an audio and/or video track - upgrades the call
-     * @param audio - should add an audio track
-     * @param video - should add an video track
-     */
-    private upgradeCall;
-    /**
-     * Returns true if this.remoteSDPStreamMetadata is defined, otherwise returns false
-     * @returns can screenshare
-     */
-    opponentSupportsSDPStreamMetadata(): boolean;
-    /**
-     * If there is a screensharing stream returns true, otherwise returns false
-     * @returns is screensharing
-     */
-    isScreensharing(): boolean;
-    /**
-     * Starts/stops screensharing
-     * @param enabled - the desired screensharing state
-     * @param opts - screen sharing options
-     * @returns new screensharing state
-     */
-    setScreensharingEnabled(enabled: boolean, opts?: IScreensharingOpts): Promise<boolean>;
-    /**
-     * Starts/stops screensharing
-     * Should be used ONLY if the opponent doesn't support SDPStreamMetadata
-     * @param enabled - the desired screensharing state
-     * @param opts - screen sharing options
-     * @returns new screensharing state
-     */
-    private setScreensharingEnabledWithoutMetadataSupport;
-    /**
-     * Replaces/adds the tracks from the passed stream to the localUsermediaStream
-     * @param stream - to use a replacement for the local usermedia stream
-     */
-    updateLocalUsermediaStream(stream: MediaStream, forceAudio?: boolean, forceVideo?: boolean): Promise<void>;
-    /**
-     * Set whether our outbound video should be muted or not.
-     * @param muted - True to mute the outbound video.
-     * @returns the new mute state
-     */
-    setLocalVideoMuted(muted: boolean): Promise<boolean>;
-    /**
-     * 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(): boolean;
-    /**
-     * Set whether the microphone should be muted or not.
-     * @param muted - True to mute the mic.
-     * @returns the new mute state
-     */
-    setMicrophoneMuted(muted: boolean): Promise<boolean>;
-    /**
-     * 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(): boolean;
-    /**
-     * @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(): boolean;
-    setRemoteOnHold(onHold: boolean): void;
-    /**
-     * 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(): boolean;
-    /**
-     * Sends a DTMF digit to the other party
-     * @param digit - The digit (nb. string - '#' and '*' are dtmf too)
-     */
-    sendDtmfDigit(digit: string): void;
-    private updateMuteStatus;
-    sendMetadataUpdate(): Promise<void>;
-    private gotCallFeedsForInvite;
-    private sendAnswer;
-    private queueGotCallFeedsForAnswer;
-    private mungeSdp;
-    private createOffer;
-    private createAnswer;
-    private gotCallFeedsForAnswer;
-    /**
-     * Internal
-     */
-    private gotLocalIceCandidate;
-    private onIceGatheringStateChange;
-    onRemoteIceCandidatesReceived(ev: MatrixEvent): Promise<void>;
-    /**
-     * Used by MatrixClient.
-     */
-    onAnswerReceived(event: MatrixEvent): Promise<void>;
-    onSelectAnswerReceived(event: MatrixEvent): Promise<void>;
-    onNegotiateReceived(event: MatrixEvent): Promise<void>;
-    private updateRemoteSDPStreamMetadata;
-    onSDPStreamMetadataChangedReceived(event: MatrixEvent): void;
-    onAssertedIdentityReceived(event: MatrixEvent): Promise<void>;
-    callHasEnded(): boolean;
-    private queueGotLocalOffer;
-    private wrappedGotLocalOffer;
-    private gotLocalOffer;
-    private getLocalOfferFailed;
-    private getUserMediaFailed;
-    private placeCallFailed;
-    private onIceConnectionStateChanged;
-    private onSignallingStateChanged;
-    private onTrack;
-    private onDataChannel;
-    /**
-     * 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)
-     */
-    private getRidOfRTXCodecs;
-    private onNegotiationNeeded;
-    onHangupReceived: (msg: MCallHangupReject) => void;
-    onRejectReceived: (msg: MCallHangupReject) => void;
-    onAnsweredElsewhere: (msg: MCallAnswer) => void;
-    /**
-     * @internal
-     */
-    private sendVoipEvent;
-    /**
-     * 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
-     */
-    private queueCandidate;
-    private discardDuplicateCandidates;
-    transfer(targetUserId: string): Promise<void>;
-    transferToCall(transferTargetCall: MatrixCall): Promise<void>;
-    private terminate;
-    private stopAllMedia;
-    private checkForErrorListener;
-    private sendCandidateQueue;
-    /**
-     * Place a call to this room.
-     * @throws if you have not specified a listener for 'error' events.
-     * @throws if have passed audio=false.
-     */
-    placeCall(audio: boolean, video: boolean): Promise<void>;
-    /**
-     * 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.
-     */
-    placeCallWithCallFeeds(callFeeds: CallFeed[], requestScreenshareFeed?: boolean): Promise<void>;
-    private createPeerConnection;
-    private partyIdMatches;
-    private chooseOpponent;
-    private addBufferedIceCandidates;
-    private addIceCandidates;
-    get hasPeerConnection(): boolean;
-    initStats(stats: GroupCallStats, peerId?: string): void;
-}
-export declare function setTracksEnabled(tracks: Array<MediaStreamTrack>, enabled: boolean): void;
-export declare function supportsMatrixCall(): boolean;
-/**
- * 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.
- */
-export declare function createNewMatrixCall(client: MatrixClient, roomId: string, options?: Pick<CallOpts, "forceTURN" | "invitee" | "opponentDeviceId" | "opponentSessionId" | "groupCallId">): MatrixCall | null;
-export {};
-//# sourceMappingURL=call.d.ts.map

package/lib/webrtc/call.d.ts.map

@@ -1 +0,0 @@
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