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

package/dist/index.d.mts

@@ -563,6 +563,24 @@ 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
@@ -615,10 +633,8 @@ 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}.
@@ -653,7 +669,7 @@ 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.

package/dist/index.mjs

@@ -4152,9 +4152,9 @@ var LocalTrack = class {
     const stream = this.trackContext.stream;
     const oldTrack = this.trackContext.track;
     if (!this.sender) throw Error("There is no RTCRtpSender for this track id!");
-    stream?.getTracks().forEach((track) => {
-      stream?.removeTrack(track);
-    });
+    if (oldTrack) {
+      stream?.removeTrack(oldTrack);
+    }
     if (newTrack) {
       stream?.addTrack(newTrack);
     }
@@ -4236,6 +4236,18 @@ var LocalTrack = class {
       {}
     );
   };
+  getAudioLevel = async () => {
+    if (!this.sender) return null;
+    try {
+      const stats = await this.sender.getStats();
+      const source = [...stats.values()].find(
+        (r) => r.type === "media-source" && r.kind === "audio" && typeof r.audioLevel === "number"
+      );
+      return source ? { level: source.audioLevel } : null;
+    } catch {
+      return null;
+    }
+  };
   createTrackVariantBitratesEvent = () => {
   };
 };
@@ -4454,6 +4466,9 @@ var Local = class {
       localTrack.addTrackToConnection();
     });
   };
+  getLocalTrackAudioLevel = async (trackId) => {
+    return this.localTracks[trackId]?.getAudioLevel() ?? null;
+  };
 };
 
 // src/tracks/LocalTrackManager.ts
@@ -4785,6 +4800,24 @@ var WebRTCEndpoint = class extends EventEmitter3 {
     const connect = MediaEvent_Connect.create({ metadataJson: JSON.stringify(metadata) });
     this.sendMediaEvent({ connect });
   };
+  /**
+   * 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) {
+    return this.local.getLocalTrackAudioLevel(trackId);
+  }
   /**
    * Feeds media event received from RTC Engine to {@link WebRTCEndpoint}.
    * This function should be called whenever some media event from RTC Engine
@@ -4966,10 +4999,8 @@ var WebRTCEndpoint = class extends EventEmitter3 {
    * @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}.
@@ -5004,25 +5035,27 @@ var WebRTCEndpoint = class extends EventEmitter3 {
    *  .forEach((track) => webrtc.addTrack(track, localStream));
    * ```
    */
-  async addTrack(track, trackMetadata, _simulcastConfig = {
+  async addTrack(track, trackMetadata, simulcastConfig = {
     enabled: false,
     enabledVariants: [],
     disabledVariants: []
-  }, _maxBandwidth = 0) {
+  }, maxBandwidth = 0, stream) {
     const resolutionNotifier = new Deferred();
     const trackId = this.getTrackId(uuidv4());
-    const stream = new MediaStream();
-    const simulcastConfig = {
-      enabled: false,
-      enabledVariants: [],
-      disabledVariants: []
-    };
-    const maxBandwidth = typeof _maxBandwidth === "number" && _maxBandwidth > 0 ? _maxBandwidth : 0;
+    const trackStream = stream ?? new MediaStream();
+    const resolvedMaxBandwidth = typeof maxBandwidth === "number" && maxBandwidth > 0 ? maxBandwidth : 0;
     try {
-      stream.addTrack(track);
+      if (!stream) trackStream.addTrack(track);
       this.commandsQueue.pushCommand({
-        handler: async () => this.localTrackManager.addTrackHandler(trackId, track, stream, trackMetadata, simulcastConfig, maxBandwidth),
-        parse: () => this.localTrackManager.parseAddTrack(track, simulcastConfig, maxBandwidth),
+        handler: async () => this.localTrackManager.addTrackHandler(
+          trackId,
+          track,
+          trackStream,
+          trackMetadata,
+          simulcastConfig,
+          resolvedMaxBandwidth
+        ),
+        parse: () => this.localTrackManager.parseAddTrack(track, simulcastConfig, resolvedMaxBandwidth),
         resolve: "after-renegotiation",
         resolutionNotifier
       });
@@ -5033,10 +5066,10 @@ var WebRTCEndpoint = class extends EventEmitter3 {
     this.emit("localTrackAdded", {
       trackId,
       track,
-      stream,
+      stream: trackStream,
       trackMetadata,
       simulcastConfig,
-      maxBandwidth
+      maxBandwidth: resolvedMaxBandwidth
     });
     return trackId;
   }

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@fishjam-cloud/webrtc-client",
-  "version": "0.25.3",
+  "version": "0.26.0-rc.1",
   "description": "Typescript client library for ExWebRTC/WebRTC endpoint in Membrane RTC Engine",
   "license": "Apache-2.0",
   "author": "Fishjam Team",