@fishjam-cloud/ts-client 0.9.0 → 0.10.1

package/dist/src/webrtc/webRTCEndpoint.d.ts

@@ -1,297 +0,0 @@
-import type { MediaEvent, SerializedMediaEvent } from './mediaEvent';
-import type TypedEmitter from 'typed-emitter';
-import type { BandwidthLimit, Config, Encoding, SimulcastConfig, TrackBandwidthLimit, TrackContext, WebRTCEndpointEvents } from './types';
-import type { EndpointWithTrackContext } from './internal';
-import { ConnectionManager } from './ConnectionManager';
-declare const WebRTCEndpoint_base: new <EndpointMetadata_1, TrackMetadata_1>() => TypedEmitter<Required<WebRTCEndpointEvents<EndpointMetadata_1, TrackMetadata_1>>>;
-/**
- * Main class that is responsible for connecting to the RTC Engine, sending and receiving media.
- */
-export declare class WebRTCEndpoint<EndpointMetadata = any, TrackMetadata = any> extends WebRTCEndpoint_base<EndpointMetadata, TrackMetadata> {
-    private readonly endpointMetadataParser;
-    private readonly trackMetadataParser;
-    private readonly localTrackManager;
-    private readonly remote;
-    private readonly local;
-    private readonly commandsQueue;
-    bandwidthEstimation: bigint;
-    connectionManager?: ConnectionManager;
-    private clearConnectionCallbacks;
-    constructor(config?: Config<EndpointMetadata, TrackMetadata>);
-    /**
-     * Tries to connect to the RTC Engine. If user is successfully connected then {@link WebRTCEndpointEvents.connected}
-     * will be emitted.
-     *
-     * @param metadata - Any information that other endpoints will receive in {@link WebRTCEndpointEvents.endpointAdded}
-     * after accepting this endpoint
-     *
-     * @example
-     * ```ts
-     * let webrtc = new WebRTCEndpoint();
-     * webrtc.connect({displayName: "Bob"});
-     * ```
-     */
-    connect: (metadata: EndpointMetadata) => void;
-    /**
-     * Feeds media event received from RTC Engine to {@link WebRTCEndpoint}.
-     * This function should be called whenever some media event from RTC Engine
-     * was received and can result in {@link WebRTCEndpoint} generating some other
-     * media events.
-     *
-     * @param mediaEvent - String data received over custom signalling layer.
-     *
-     * @example
-     * This example assumes phoenix channels as signalling layer.
-     * As phoenix channels require objects, RTC Engine encapsulates binary data into
-     * map with one field that is converted to object with one field on the TS side.
-     * ```ts
-     * webrtcChannel.on("mediaEvent", (event) => webrtc.receiveMediaEvent(event.data));
-     * ```
-     */
-    receiveMediaEvent: (mediaEvent: SerializedMediaEvent) => Promise<void>;
-    private getEndpointId;
-    private onTrackReady;
-    /**
-     * Retrieves statistics related to the RTCPeerConnection.
-     * These statistics provide insights into the performance and status of the connection.
-     *
-     * @return {Promise<RTCStatsReport>}
-     *
-     * @external RTCPeerConnection#getStats()
-     * @see {@link https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/getStats | MDN Web Docs: RTCPeerConnection.getStats()}
-     */
-    getStatistics(selector?: MediaStreamTrack | null): Promise<RTCStatsReport>;
-    /**
-     * Returns a snapshot of currently received remote tracks.
-     *
-     * @example
-     * if (webRTCEndpoint.getRemoteTracks()[trackId]?.simulcastConfig?.enabled) {
-     *   webRTCEndpoint.setTargetTrackEncoding(trackId, encoding);
-     * }
-     */
-    getRemoteTracks(): Record<string, TrackContext<EndpointMetadata, TrackMetadata>>;
-    /**
-     * Returns a snapshot of currently received remote endpoints.
-     */
-    getRemoteEndpoints(): Record<string, EndpointWithTrackContext<EndpointMetadata, TrackMetadata>>;
-    getLocalEndpoint(): EndpointWithTrackContext<EndpointMetadata, TrackMetadata>;
-    getBandwidthEstimation(): bigint;
-    private handleMediaEvent;
-    private onSdpAnswer;
-    /**
-     * Adds track that will be sent to the RTC Engine.
-     * @param track - Audio or video track e.g. from your microphone or camera.
-     * @param trackMetadata - Any information about this track that other endpoints will
-     * receive in {@link WebRTCEndpointEvents.endpointAdded}. E.g. this can source of the track - whether it's
-     * screensharing, webcam or some other media device.
-     * @param simulcastConfig - Simulcast configuration. By default simulcast is disabled.
-     * For more information refer to {@link SimulcastConfig}.
-     * @param maxBandwidth - maximal bandwidth this track can use.
-     * Defaults to 0 which is unlimited.
-     * This option has no effect for simulcast and audio tracks.
-     * For simulcast tracks use `{@link WebRTCEndpoint.setTrackBandwidth}.
-     * @returns {string} Returns id of added track
-     * @example
-     * ```ts
-     * let localStream: MediaStream = new MediaStream();
-     * try {
-     *   localAudioStream = await navigator.mediaDevices.getUserMedia(
-     *     AUDIO_CONSTRAINTS
-     *   );
-     *   localAudioStream
-     *     .getTracks()
-     *     .forEach((track) => localStream.addTrack(track));
-     * } catch (error) {
-     *   console.error("Couldn't get microphone permission:", error);
-     * }
-     *
-     * try {
-     *   localVideoStream = await navigator.mediaDevices.getUserMedia(
-     *     VIDEO_CONSTRAINTS
-     *   );
-     *   localVideoStream
-     *     .getTracks()
-     *     .forEach((track) => localStream.addTrack(track));
-     * } catch (error) {
-     *  console.error("Couldn't get camera permission:", error);
-     * }
-     *
-     * localStream
-     *  .getTracks()
-     *  .forEach((track) => webrtc.addTrack(track, localStream));
-     * ```
-     */
-    addTrack(track: MediaStreamTrack, trackMetadata?: TrackMetadata, simulcastConfig?: SimulcastConfig, maxBandwidth?: TrackBandwidthLimit): Promise<string>;
-    /**
-     * Replaces a track that is being sent to the RTC Engine.
-     * @param trackId - Audio or video track.
-     * @param {string} trackId - Id of audio or video track to replace.
-     * @param {MediaStreamTrack} newTrack
-     * @returns {Promise<boolean>} success
-     * @example
-     * ```ts
-     * // setup camera
-     * let localStream: MediaStream = new MediaStream();
-     * try {
-     *   localVideoStream = await navigator.mediaDevices.getUserMedia(
-     *     VIDEO_CONSTRAINTS
-     *   );
-     *   localVideoStream
-     *     .getTracks()
-     *     .forEach((track) => localStream.addTrack(track));
-     * } catch (error) {
-     *   console.error("Couldn't get camera permission:", error);
-     * }
-     * let oldTrackId;
-     * localStream
-     *  .getTracks()
-     *  .forEach((track) => trackId = webrtc.addTrack(track, localStream));
-     *
-     * // change camera
-     * const oldTrack = localStream.getVideoTracks()[0];
-     * let videoDeviceId = "abcd-1234";
-     * navigator.mediaDevices.getUserMedia({
-     *      video: {
-     *        ...(VIDEO_CONSTRAINTS as {}),
-     *        deviceId: {
-     *          exact: videoDeviceId,
-     *        },
-     *      }
-     *   })
-     *   .then((stream) => {
-     *     let videoTrack = stream.getVideoTracks()[0];
-     *     webrtc.replaceTrack(oldTrackId, videoTrack);
-     *   })
-     *   .catch((error) => {
-     *     console.error('Error switching camera', error);
-     *   })
-     * ```
-     */
-    replaceTrack(trackId: string, newTrack: MediaStreamTrack | null): Promise<void>;
-    /**
-     * Updates maximum bandwidth for the track identified by trackId.
-     * This value directly translates to quality of the stream and, in case of video, to the amount of RTP packets being sent.
-     * In case trackId points at the simulcast track bandwidth is split between all of the variant streams proportionally to their resolution.
-     *
-     * @param {string} trackId
-     * @param {BandwidthLimit} bandwidth in kbps
-     * @returns {Promise<boolean>} success
-     */
-    setTrackBandwidth(trackId: string, bandwidth: BandwidthLimit): Promise<void>;
-    /**
-     * Updates maximum bandwidth for the given simulcast encoding of the given track.
-     *
-     * @param {string} trackId - id of the track
-     * @param {string} rid - rid of the encoding
-     * @param {BandwidthLimit} bandwidth - desired max bandwidth used by the encoding (in kbps)
-     * @returns
-     */
-    setEncodingBandwidth(trackId: string, rid: string, bandwidth: BandwidthLimit): Promise<void>;
-    /**
-     * Removes a track from connection that was sent to the RTC Engine.
-     * @param {string} trackId - Id of audio or video track to remove.
-     * @example
-     * ```ts
-     * // setup camera
-     * let localStream: MediaStream = new MediaStream();
-     * try {
-     *   localVideoStream = await navigator.mediaDevices.getUserMedia(
-     *     VIDEO_CONSTRAINTS
-     *   );
-     *   localVideoStream
-     *     .getTracks()
-     *     .forEach((track) => localStream.addTrack(track));
-     * } catch (error) {
-     *   console.error("Couldn't get camera permission:", error);
-     * }
-     *
-     * let trackId
-     * localStream
-     *  .getTracks()
-     *  .forEach((track) => trackId = webrtc.addTrack(track, localStream));
-     *
-     * // remove track
-     * webrtc.removeTrack(trackId)
-     * ```
-     */
-    removeTrack(trackId: string): Promise<void>;
-    /**
-     * Sets track variant that server should send to the client library.
-     *
-     * The variant will be sent whenever it is available.
-     * If chosen variant is temporarily unavailable, some other variant
-     * will be sent until the chosen variant becomes active again.
-     *
-     * @param {string} trackId - id of track
-     * @param {Encoding} variant - variant to receive
-     * @example
-     * ```ts
-     * webrtc.setTargetTrackEncoding(incomingTrackCtx.trackId, "l")
-     * ```
-     */
-    setTargetTrackEncoding(trackId: string, variant: Encoding): void;
-    /**
-     * Enables track encoding so that it will be sent to the server.
-     * @param {string} trackId - id of track
-     * @param {Encoding} encoding - encoding that will be enabled
-     * @example
-     * ```ts
-     * const trackId = webrtc.addTrack(track, stream, {}, {enabled: true, activeEncodings: ["l", "m", "h"]});
-     * webrtc.disableTrackEncoding(trackId, "l");
-     * // wait some time
-     * webrtc.enableTrackEncoding(trackId, "l");
-     * ```
-     */
-    enableTrackEncoding: (trackId: string, encoding: Encoding) => Promise<void>;
-    /**
-     * Disables track encoding so that it will be no longer sent to the server.
-     * @param {string} trackId - id of track
-     * @param {Encoding} encoding - encoding that will be disabled
-     * @example
-     * ```ts
-     * const trackId = webrtc.addTrack(track, stream, {}, {enabled: true, activeEncodings: ["l", "m", "h"]});
-     * webrtc.disableTrackEncoding(trackId, "l");
-     * ```
-     */
-    disableTrackEncoding: (trackId: string, encoding: Encoding) => Promise<void>;
-    /**
-     * Updates the metadata for the current endpoint.
-     * @param metadata - Data about this endpoint that other endpoints will receive upon being added.
-     *
-     * If the metadata is different from what is already tracked in the room, the optional
-     * event `endpointUpdated` will be emitted for other endpoint in the room.
-     */
-    updateEndpointMetadata: (metadata: any) => void;
-    /**
-     * Updates the metadata for a specific track.
-     * @param trackId - trackId (generated in addTrack) of audio or video track.
-     * @param trackMetadata - Data about this track that other endpoint will receive upon being added.
-     *
-     * If the metadata is different from what is already tracked in the room, the optional
-     * event `trackUpdated` will be emitted for other endpoints in the room.
-     */
-    updateTrackMetadata: (trackId: string, trackMetadata: any) => void;
-    /**
-     * Disconnects from the room. This function should be called when user disconnects from the room
-     * in a clean way e.g. by clicking a dedicated, custom button `disconnect`.
-     * As a result there will be generated one more media event that should be
-     * sent to the RTC Engine. Thanks to it each other endpoint will be notified
-     * that endpoint was removed in {@link WebRTCEndpointEvents.endpointRemoved},
-     */
-    disconnect: () => void;
-    /**
-     * Cleans up {@link WebRTCEndpoint} instance.
-     */
-    cleanUp: () => void;
-    private getTrackId;
-    sendMediaEvent: (mediaEvent: MediaEvent) => void;
-    private createAndSendOffer;
-    private onOfferData;
-    private setConnection;
-    private onRemoteCandidate;
-    private onLocalCandidate;
-    private onIceCandidateError;
-    private onConnectionStateChange;
-    private onIceConnectionStateChange;
-}
-export {};