@fishjam-cloud/ts-client 0.25.3 → 0.26.0-rc.1

package/dist/index.d.mts

@@ -4,6 +4,7 @@ import TypedEmitter from 'typed-emitter';
 declare const AUTH_ERROR_REASONS: readonly ["missing token", "invalid token", "expired token", "room not found", "peer not found"];
 type AuthErrorReason = (typeof AUTH_ERROR_REASONS)[number];
 declare const isAuthError: (error: string) => error is AuthErrorReason;
+declare const normalizeCloseReason: (reason: string) => string;
 
 declare class TrackTypeError extends Error {
     constructor();
@@ -603,6 +604,24 @@ declare class WebRTCEndpoint extends WebRTCEndpoint_base {
      * ```
      */
     connect: (metadata: unknown) => void;
+    /**
+     * Returns the current audio level for a local audio track, if available.
+     *
+     * This method only works for local **audio** tracks that have been negotiated
+     * with the remote peer and for which an underlying `RTCRtpSender` and
+     * statistics are available.
+     *
+     * @param trackId - Identifier of the local track to query, as used when
+     * adding or managing local tracks on this endpoint.
+     * @returns A promise that resolves to `{ level: number }` when an audio
+     * level can be determined for the given track, or `null` if:
+     * - the track does not exist,
+     * - the track is not an audio track,
+     * - the track has not yet been negotiated / no sender exists
+     */
+    getLocalTrackAudioLevel(trackId: string): Promise<{
+        level: number;
+    } | null>;
     /**
      * Feeds media event received from RTC Engine to {@link WebRTCEndpoint}.
      * This function should be called whenever some media event from RTC Engine
@@ -655,10 +674,8 @@ declare class WebRTCEndpoint extends WebRTCEndpoint_base {
      * @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 parameter. **Currently ignored** - simulcast is disabled
-     * regardless of the value passed. This is a temporary change until bandwidth estimation is implemented or
-     * manual track selection support is added. For more information refer to {@link SimulcastConfig}.
-     * @param _maxBandwidth - maximal bandwidth this track can use. **Currently processed with a threshold check**:
+     * @param simulcastConfig - Simulcast configuration. For more information refer to {@link SimulcastConfig}.
+     * @param maxBandwidth - maximal bandwidth this track can use. **Currently processed with a threshold check**:
      * if the value is a positive number, it will be used; otherwise, it defaults to 0 (unlimited).
      * This option has no effect for simulcast and audio tracks.
      * For simulcast tracks use `{@link WebRTCEndpoint.setTrackBandwidth}.
@@ -693,7 +710,7 @@ declare class WebRTCEndpoint extends WebRTCEndpoint_base {
      *  .forEach((track) => webrtc.addTrack(track, localStream));
      * ```
      */
-    addTrack(track: MediaStreamTrack, trackMetadata?: unknown, _simulcastConfig?: MediaEvent_Track_SimulcastConfig, _maxBandwidth?: TrackBandwidthLimit): Promise<string>;
+    addTrack(track: MediaStreamTrack, trackMetadata?: unknown, simulcastConfig?: MediaEvent_Track_SimulcastConfig, maxBandwidth?: TrackBandwidthLimit, stream?: MediaStream): Promise<string>;
     /**
      * Replaces a track that is being sent to the RTC Engine.
      * @param trackId - Audio or video track.
@@ -916,7 +933,7 @@ declare class WebRTCEndpoint extends WebRTCEndpoint_base {
     private onIceConnectionStateChange;
 }
 
-declare const JOIN_ERRORS: readonly ["reached peers limit", "room not found", "node not found", "Invalid SDK version"];
+declare const JOIN_ERRORS: readonly ["reached peers limit", "room not found", "node not found", "invalid sdk version"];
 type JoinErrorReason = (typeof JOIN_ERRORS)[number];
 declare const isJoinError: (error: string) => error is JoinErrorReason;
 
@@ -1094,6 +1111,7 @@ type MessageEvents<P, S> = {
      * by the server. It's measured in bits per second.
      */
     bandwidthEstimationChanged: (estimation: bigint) => void;
+    encodingChanged: TrackContextEvents['encodingChanged'];
     targetTrackEncodingRequested: (event: Parameters<WebRTCEndpointEvents['targetTrackEncodingRequested']>[0]) => void;
     localTrackAdded: (event: Parameters<WebRTCEndpointEvents['localTrackAdded']>[0]) => void;
     localTrackRemoved: (event: Parameters<WebRTCEndpointEvents['localTrackRemoved']>[0]) => void;
@@ -1190,6 +1208,9 @@ declare class FishjamClient<PeerMetadata = GenericMetadata, ServerMetadata = Gen
     private reconnectManager;
     private peerMessageQueue;
     private sendStatisticsInterval;
+    private cameraStream;
+    private screenShareStream;
+    private trackIdToTrack;
     constructor(config?: CreateConfig);
     /**
      * Uses the WebSocket connection and {@link !WebRTCEndpoint | WebRTCEndpoint} to join to the room. Registers the callbacks to
@@ -1572,6 +1593,23 @@ declare class FishjamClient<PeerMetadata = GenericMetadata, ServerMetadata = Gen
      */
     disconnect(): void;
     cleanup(): void;
+    /**
+     * Returns the current audio level for a local track.
+     *
+     * The `level` represents a normalized audio level in the range 0.0–1.0,
+     * derived from WebRTC statistics for the given local audio track.
+     *
+     * This method returns `null` when the WebRTC layer is not initialized, when the track
+     * cannot be found among local tracks, or when audio statistics are not yet or no longer
+     * available for the track.
+     *
+     * @param trackId - The ID of the local track to query.
+     * @returns A promise resolving to an object containing the audio `level`, or `null`
+     *          if the track is unknown or stats are not available.
+     */
+    getLocalTrackAudioLevel(trackId: string): Promise<{
+        level: number;
+    } | null>;
 }
 
 type ReceiveLivestreamResult = {
@@ -1595,4 +1633,4 @@ type LivestreamCallbacks = {
 declare function receiveLivestream(url: string, token?: string, callbacks?: LivestreamCallbacks): Promise<ReceiveLivestreamResult>;
 declare function publishLivestream(stream: MediaStream, url: string, token: string, callbacks?: LivestreamCallbacks): Promise<PublishLivestreamResult>;
 
-export { AUTH_ERROR_REASONS, type AuthErrorReason, type BandwidthLimit, type ClientType, type Component, type ConnectConfig, type CreateConfig, type DataCallback, type DataChannelMessagePayload, type DataChannelOptions, type DataChannelType, type EncodingReason, type Endpoint, FishjamClient, type FishjamTrackContext, type GenericMetadata, JOIN_ERRORS, type JoinErrorReason, type LivestreamCallbacks, LivestreamError, type Logger, type MediaEvent, type MessageEvents, type Metadata, type Peer, type PublishLivestreamResult, type ReceiveLivestreamResult, type ReconnectConfig, type ReconnectionStatus, type SerializedMediaEvent, type SimulcastBandwidthLimit, MediaEvent_Track_SimulcastConfig as SimulcastConfig, type TrackBandwidthLimit, type TrackContext, type TrackContextEvents, type TrackKind, type TrackMetadata, TrackTypeError, type VadStatus, Variant, WebRTCEndpoint, type WebRTCEndpointEvents, getLogger, isAuthError, isJoinError, publishLivestream, receiveLivestream };
+export { AUTH_ERROR_REASONS, type AuthErrorReason, type BandwidthLimit, type ClientType, type Component, type ConnectConfig, type CreateConfig, type DataCallback, type DataChannelMessagePayload, type DataChannelOptions, type DataChannelType, type EncodingReason, type Endpoint, FishjamClient, type FishjamTrackContext, type GenericMetadata, JOIN_ERRORS, type JoinErrorReason, type LivestreamCallbacks, LivestreamError, type Logger, type MediaEvent, type MessageEvents, type Metadata, type Peer, type PublishLivestreamResult, type ReceiveLivestreamResult, type ReconnectConfig, type ReconnectionStatus, type SerializedMediaEvent, type SimulcastBandwidthLimit, MediaEvent_Track_SimulcastConfig as SimulcastConfig, type TrackBandwidthLimit, type TrackContext, type TrackContextEvents, type TrackKind, type TrackMetadata, TrackTypeError, type VadStatus, Variant, WebRTCEndpoint, type WebRTCEndpointEvents, getLogger, isAuthError, isJoinError, normalizeCloseReason, publishLivestream, receiveLivestream };