@unwanted/matrix-sdk-mini 34.12.0-1 → 34.12.0-3

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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