@fishjam-cloud/webrtc-client 0.26.0-rc.0 → 0.26.0-rc.1

package/dist/index.d.mts

@@ -633,10 +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}.
@@ -671,7 +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);
     }
@@ -4999,10 +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}.
@@ -5037,25 +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
       });
@@ -5066,10 +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": "0.26.0-rc.0",
+  "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",