@unwanted/matrix-sdk-mini 34.12.0-1

package/lib/webrtc/call.d.ts

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+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

@@ -0,0 +1 @@
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