@opentok/client 2.27.6 → 2.28.0

package/dist/js/opentok.d.ts

@@ -277,6 +277,10 @@ declare namespace OT {
       track: MediaStreamTrack | undefined
     };
 
+    mediaStreamAvailable: Event<'mediaStreamAvailable', Publisher> & {
+      mediaStream: MediaStream;
+    };
+
     streamCreated: Event<'streamCreated', Publisher> & {
       stream: Stream;
     };
@@ -549,6 +553,10 @@ declare namespace OT {
 
     encryptionSecretMatch: Event<'encryptionSecretMatch', Subscriber>;
 
+    mediaStreamAvailable: Event<'mediaStreamAvailable', Subscriber> & {
+      mediaStream: MediaStream;
+    };
+
     videoDimensionsChanged: VideoDimensionsChangedEvent<Subscriber>;
 
     videoDisabled: Event<'videoDisabled', Subscriber> & {

package/dist/js/opentok.js

@@ -1,11 +1,11 @@
 /**
- * @license OpenTok.js 2.27.6 cf306f8
+ * @license OpenTok.js 2.28.0 1ea3fa6
  *
  * Copyright (c) 2010-2024 TokBox, Inc.
  * Subject to the applicable Software Development Kit (SDK) License Agreement:
  * https://www.vonage.com/legal/communications-apis/terms-of-use/
  *
- * Date: Wed, 05 Jun 2024 14:03:04 GMT
+ * Date: Wed, 26 Jun 2024 15:49:08 GMT
  */
 
 (function webpackUniversalModuleDefinition(root, factory) {
@@ -6753,6 +6753,29 @@ function EventsFactory(deps) {
       captionsEnabled
     });
   };
+
+  /**
+   * This is a <b>beta</b> feature.
+   * Defines an event object for the <code>mediaStreamAvailable</code> event dispatched by both
+   * the Publisher and Subscriber objects. The Publisher and Subscriber objects both dispatch a <code>mediaStreamAvailable</code>
+   * event when the MediaStream is available to be used in custom video elements. The Publisher and Subscriber objects should still be used to
+   * control all other aspects of the video API such as muting, disconnecting, and more.
+   *
+   * <p>Note that the audio will play through both the custom element and the Publisher or Subscriber widget.
+   * The custom elements should be muted to avoid echo.</p>
+   *
+   * @property {MediaStream} mediaStream The MediaStream object.
+   * @see <a href="Publisher.html#event:mediaStreamAvailable">Publisher mediaStreamAvailable
+   *   event</a>
+   * @see <a href="Subscriber.html#event:mediaStreamAvailable">Subscriber mediaStreamAvailable
+   *   event</a>
+   * @augments Event
+   */
+  Events.MediaStreamAvailable = function MediaStreamAvailable(mediaStream) {
+    return new _Event4.default('mediaStreamAvailable', false, {
+      mediaStream
+    });
+  };
   Events.StreamUpdatedEvent = function StreamUpdatedEvent(stream, key, oldValue, newValue) {
     return new _Event4.default('updated', false, {
       target: stream,
@@ -7922,7 +7945,7 @@ const logging = (0, _log.default)('StaticConfig');
 */
 
 /** @type builtInConfig */
-const builtInConfig = (0, _cloneDeep.default)({"version":"v2.27.6","buildHash":"cf306f8","minimumVersion":{"firefox":52,"chrome":49},"debug":false,"websiteURL":"http://www.tokbox.com","configURL":"https://config.opentok.com","ipWhitelistConfigURL":"","cdnURL":"","loggingURL":"https://hlg.tokbox.com/prod","apiURL":"https://anvil.opentok.com"});
+const builtInConfig = (0, _cloneDeep.default)({"version":"v2.28.0","buildHash":"1ea3fa6","minimumVersion":{"firefox":52,"chrome":49},"debug":false,"websiteURL":"http://www.tokbox.com","configURL":"https://config.opentok.com","ipWhitelistConfigURL":"","cdnURL":"","loggingURL":"https://hlg.tokbox.com/prod","apiURL":"https://anvil.opentok.com"});
 const whitelistAllowedRuntimeProperties = (0, _pick.default)(['apiURL', 'assetURL', 'cdnURL', 'sessionInfoOverrides', 'loggingURL']);
 const liveConfigMap = {
   apiUrl: 'apiURL',
@@ -18472,6 +18495,7 @@ function PublisherFactory(_ref) {
           }
         });
         widgetView.on('videoElementCreated', element => {
+          this.dispatchEvent(new Events.MediaStreamAvailable(webRTCStream));
           const event = new Events.VideoElementCreatedEvent(element);
           this.dispatchEvent(event);
         });
@@ -22210,6 +22234,15 @@ function PublisherFactory(_ref) {
   * @memberof Publisher
   */
 
+  /**
+  * This is a <b>beta</b> feature.
+  * Dispatched when the MediaStream is available.
+  * @see Event
+  * @name mediaStreamAvailable
+  * @event
+  * @memberof Publisher
+  */
+
   /**
   * Dispatched when the Allow/Deny box is closed. (This is the dialog box in which the
   * user can grant the app access to the camera and microphone.)
@@ -33380,7 +33413,7 @@ var _amrVideoBufferError = __webpack_require__(322);
 var _updateTrackOnStream = _interopRequireDefault(__webpack_require__(331));
 var _time = _interopRequireDefault(__webpack_require__(110));
 var _amrStates = _interopRequireDefault(__webpack_require__(325));
-const _excluded = ["AudioLevelMeter", "AudioLevelTransformer", "BackingBar", "Chrome", "env", "Errors", "Events", "ExceptionCodes", "audioLevelSamplerFactory", "getStatsHelpers", "hasAudioOutputLevelStatCapability", "hasRemoteStreamsWithWebAudio", "interpretPeerConnectionError", "logging", "MuteButton", "NamePanel", "otError", "OTErrorClass", "OTHelpers", "StylableComponent", "SubscriberPeerConnection", "SubscribingState", "VideoDisabledIndicator", "AudioBlockedIndicator", "VideoUnsupportedIndicator", "watchFrameRate", "createSendMethod", "parseIceServers", "document", "WidgetView", "createAudioConnector", "ResizeObserverPolyfill", "AMRStateDurations"];
+const _excluded = ["AudioLevelMeter", "AudioLevelTransformer", "BackingBar", "Chrome", "env", "Errors", "Events", "ExceptionCodes", "audioLevelSamplerFactory", "getStatsHelpers", "hasAudioOutputLevelStatCapability", "hasRemoteStreamsWithWebAudio", "interpretPeerConnectionError", "logging", "MuteButton", "NamePanel", "otError", "OTErrorClass", "OTHelpers", "StylableComponent", "SubscriberPeerConnection", "SubscribingState", "VideoDisabledIndicator", "AudioBlockedIndicator", "VideoUnsupportedIndicator", "watchFrameRate", "createSendMethod", "parseIceServers", "document", "WidgetView", "createAudioConnector", "ResizeObserverPolyfill", "MediaStream", "AMRStateDurations"];
 const defaultWidgetView = (0, _widget_view.default)();
 const EventsDefault = (0, _events2.default)();
 const errorsDefault = _Errors.default;
@@ -33485,6 +33518,8 @@ function SubscriberFactory(_ref2) {
     createAudioConnector = _ref3$createAudioConn === void 0 ? _audioConnector2.default : _ref3$createAudioConn,
     _ref3$ResizeObserverP = _ref3.ResizeObserverPolyfill,
     ResizeObserverPolyfill = _ref3$ResizeObserverP === void 0 ? _resizeObserverPolyfill.default : _ref3$ResizeObserverP,
+    _ref3$MediaStream = _ref3.MediaStream,
+    MediaStream = _ref3$MediaStream === void 0 ? (typeof window !== undefined ? window : global).MediaStream : _ref3$MediaStream,
     AMRStateDurations = _ref3.AMRStateDurations,
     deps = (0, _objectWithoutPropertiesLoose2.default)(_ref3, _excluded);
   const BIND_VIDEO_DELAY_MAX = 30000;
@@ -33530,7 +33565,15 @@ function SubscriberFactory(_ref2) {
     const webRTCStreams = {};
     const hybridSessionTransitionStartTimes = {};
     const _peerConnectionEvents = {};
-    const _audioConnector = createAudioConnector(webRTCStreams);
+    const updateTrackOnStream = (mediaStream, oldTrack, newTrack) => {
+      if (_exposedWebRTCStream) {
+        (0, _updateTrackOnStream.default)(_exposedWebRTCStream, oldTrack, newTrack);
+      }
+      (0, _updateTrackOnStream.default)(mediaStream, oldTrack, newTrack);
+    };
+    const _audioConnector = createAudioConnector(webRTCStreams, {
+      updateTrackOnStream
+    });
 
     /** @type {defaultWidgetView|null} */
     let _widgetView;
@@ -33582,6 +33625,7 @@ function SubscriberFactory(_ref2) {
     let _originalVideoTrack;
     let _senderStats;
     let _resizeObserver;
+    let _exposedWebRTCStream;
 
     // The audio stats watcher is only supported on chromium-based browsers that supports
     // the standard version of the getStats API.
@@ -33655,6 +33699,10 @@ function SubscriberFactory(_ref2) {
         isSubscriber: true
       }));
     };
+    const syncUpMediaStreams = (targetStream, originalStream) => {
+      (0, _updateTrackOnStream.default)(targetStream, targetStream.getAudioTracks()[0], originalStream.getAudioTracks()[0]);
+      (0, _updateTrackOnStream.default)(targetStream, targetStream.getVideoTracks()[0], originalStream.getVideoTracks()[0]);
+    };
     const bindWebRTCStream = /*#__PURE__*/function () {
       var _ref6 = (0, _asyncToGenerator2.default)( /*#__PURE__*/_regenerator.default.mark(function _callee3(webRTCStream) {
         var _stream2;
@@ -33668,6 +33716,10 @@ function SubscriberFactory(_ref2) {
               }
               return _context3.abrupt("return");
             case 2:
+              // Sync the track state of the exposed stream with our current stream
+              if (_exposedWebRTCStream) {
+                syncUpMediaStreams(_exposedWebRTCStream, webRTCStream);
+              }
               videoContainerOptions = {
                 error: onVideoError,
                 audioVolume: _audioVolume,
@@ -33675,28 +33727,28 @@ function SubscriberFactory(_ref2) {
               }; // We want to catch any error in the prebuffering when transition to P2P.
               // Otherwise, we ignore it.
               throwIfBufferFails = _isAdaptiveEnabled && _activeSourceStreamId === 'P2P';
-              _context3.prev = 4;
-              _context3.next = 7;
+              _context3.prev = 5;
+              _context3.next = 8;
               return _widgetView.bindVideo(webRTCStream, videoContainerOptions, throwIfBufferFails);
-            case 7:
-              _context3.next = 15;
+            case 8:
+              _context3.next = 16;
               break;
-            case 9:
-              _context3.prev = 9;
-              _context3.t0 = _context3["catch"](4);
+            case 10:
+              _context3.prev = 10;
+              _context3.t0 = _context3["catch"](5);
               if (!(_context3.t0 instanceof _cancel.CancellationError || _state.isDestroyed())) {
-                _context3.next = 13;
+                _context3.next = 14;
                 break;
               }
               return _context3.abrupt("return");
-            case 13:
+            case 14:
               onVideoError(_context3.t0);
               throw _context3.t0;
-            case 15:
+            case 16:
             case "end":
               return _context3.stop();
           }
-        }, _callee3, null, [[4, 9]]);
+        }, _callee3, null, [[5, 10]]);
       }));
       return function bindWebRTCStream(_x3) {
         return _ref6.apply(this, arguments);
@@ -34296,6 +34348,10 @@ function SubscriberFactory(_ref2) {
               isSwappingStreams = Object.keys(webRTCStreams).length === 1;
               sourceStreamId = peerConnection == null ? void 0 : peerConnection.getSourceStreamId();
               _session2 = _session, sessionInfo = _session2.sessionInfo;
+              if (!_exposedWebRTCStream) {
+                _exposedWebRTCStream = new MediaStream(webRTCStream);
+                _this.dispatchEvent(new Events.MediaStreamAvailable(_exposedWebRTCStream));
+              }
               _webRTCStream = webRTCStream;
 
               // save a copy of the WebRTCStream
@@ -34305,21 +34361,21 @@ function SubscriberFactory(_ref2) {
               logging.debug('OT.Subscriber.onRemoteStreamAdded with sourceStreamId', sourceStreamId);
               _state.set('BindingRemoteStream');
               if (!_videoMediaProcessorConnector) {
-                _context7.next = 14;
+                _context7.next = 15;
                 break;
               }
               if (!_videoEnabled) {
-                _context7.next = 13;
+                _context7.next = 14;
                 break;
               }
-              _context7.next = 11;
+              _context7.next = 12;
               return applyVideoConnectorOnStream(sourceStreamId);
-            case 11:
-              _context7.next = 14;
+            case 12:
+              _context7.next = 15;
               break;
-            case 13:
-              _videoMediaProcessorConnector.destroy();
             case 14:
+              _videoMediaProcessorConnector.destroy();
+            case 15:
               // Disable the audio/video, if needed
               _this.subscribeToAudio(_isSubscribingToAudio);
               if (!(_lastSubscribeToVideoReason === 'auto' && !_properties.subscribeToVideo)) {
@@ -34351,17 +34407,17 @@ function SubscriberFactory(_ref2) {
                   }
                 });
               }
-              _context7.prev = 18;
-              _context7.next = 21;
+              _context7.prev = 19;
+              _context7.next = 22;
               return bindWebRTCStream(_webRTCStream);
-            case 21:
-              _context7.next = 32;
+            case 22:
+              _context7.next = 33;
               break;
-            case 23:
-              _context7.prev = 23;
-              _context7.t0 = _context7["catch"](18);
+            case 24:
+              _context7.prev = 24;
+              _context7.t0 = _context7["catch"](19);
               if (!(0, _amrVideoBufferError.isAmrVideoBufferError)(_context7.t0)) {
-                _context7.next = 31;
+                _context7.next = 32;
                 break;
               }
               logging.error(`OT.Subscriber: ${_context7.t0.message}`);
@@ -34374,17 +34430,17 @@ function SubscriberFactory(_ref2) {
                 socket.subscriberDestroy(_stream.id, _this.widgetId, _activeSourceStreamId);
               }
               return _context7.abrupt("return");
-            case 31:
-              throw _context7.t0;
             case 32:
+              throw _context7.t0;
+            case 33:
               startAudioStatsWatcher();
               if (!_state.isDestroyed()) {
-                _context7.next = 36;
+                _context7.next = 37;
                 break;
               }
               logging.error('Subscriber destroyed');
               return _context7.abrupt("return");
-            case 36:
+            case 37:
               if (peerConnection && (typeof window !== undefined ? window : global).webkitMediaStream) {
                 // Enable any video streams that we previously disabled for OPENTOK-27112
                 peerConnection._getVideoTracks().forEach(track => {
@@ -34412,12 +34468,12 @@ function SubscriberFactory(_ref2) {
                 _widgetView.once('videoElementCreated', resolve);
                 _subscriber.once('destroyed', reject);
               });
-              _context7.next = 44;
+              _context7.next = 45;
               return videoElementCreated;
-            case 44:
-              _context7.next = 46;
+            case 45:
+              _context7.next = 47;
               return _pcConnected.promise;
-            case 46:
+            case 47:
               if (!isSwappingStreams) {
                 onLoaded();
               } else {
@@ -34453,34 +34509,34 @@ function SubscriberFactory(_ref2) {
               // Given an AMR transition, and an audio connector, we apply the connector for the new
               // relayed stream and clear it for the routed stream.
               if (!_isAdaptiveEnabled) {
-                _context7.next = 62;
+                _context7.next = 63;
                 break;
               }
               amrState.completeTransitionTo(_activeSourceStreamId);
               if (!(_isSubscribingToAudio && _audioConnector.audioMediaProcessorConnector)) {
-                _context7.next = 62;
+                _context7.next = 63;
                 break;
               }
-              _context7.prev = 53;
-              _context7.next = 56;
+              _context7.prev = 54;
+              _context7.next = 57;
               return _audioConnector.applyAudioConnectorOnRoutedToRelayedTransition();
-            case 56:
+            case 57:
               logAnalyticsEvent('setAudioMediaProcessorConnector', 'Success', {
                 message: 'Audio connector applied on routed to relayed transition.'
               });
-              _context7.next = 62;
+              _context7.next = 63;
               break;
-            case 59:
-              _context7.prev = 59;
-              _context7.t1 = _context7["catch"](53);
+            case 60:
+              _context7.prev = 60;
+              _context7.t1 = _context7["catch"](54);
               logAnalyticsEvent('setAudioMediaProcessorConnector', 'Failure', {
                 message: _context7.t1.message
               });
-            case 62:
+            case 63:
             case "end":
               return _context7.stop();
           }
-        }, _callee7, null, [[18, 23], [53, 59]]);
+        }, _callee7, null, [[19, 24], [54, 60]]);
       }));
       return function onRemoteStreamAdded(_x9, _x10) {
         return _ref15.apply(this, arguments);
@@ -35046,8 +35102,9 @@ function SubscriberFactory(_ref2) {
               _context15.next = 29;
               return createAudioLevelSampler(peerConnection);
             case 29:
+              _audioLevelSampler.webRTCStream(_webRTCStream);
               logRelayedToRoutedTransition('Success');
-            case 30:
+            case 31:
             case "end":
               return _context15.stop();
           }
@@ -36661,7 +36718,7 @@ function SubscriberFactory(_ref2) {
               return _videoMediaProcessorConnector.setTrack(_originalVideoTrack);
             case 10:
               filteredVideoTrack = _context27.sent;
-              (0, _updateTrackOnStream.default)(_webRTCStream, _originalVideoTrack, filteredVideoTrack);
+              updateTrackOnStream(_webRTCStream, _originalVideoTrack, filteredVideoTrack);
               logAnalyticsEvent('setVideoMediaProcessorConnector', 'Success');
               _context27.next = 20;
               break;
@@ -36691,7 +36748,7 @@ function SubscriberFactory(_ref2) {
         // We need to switch to the original track so we can transform it later when resetting the connector.
         const _webRTCStream$getVide2 = _webRTCStream.getVideoTracks(),
           filteredVideoTrack = _webRTCStream$getVide2[0];
-        (0, _updateTrackOnStream.default)(_webRTCStream, filteredVideoTrack, _originalVideoTrack);
+        updateTrackOnStream(_webRTCStream, filteredVideoTrack, _originalVideoTrack);
         logAnalyticsEvent('setVideoMediaProcessorConnector', 'Success', {
           message: 'reset the video track'
         });
@@ -37056,6 +37113,15 @@ function SubscriberFactory(_ref2) {
     * @memberof Subscriber
     */
 
+    /**
+    * This is a <b>beta</b> feature.
+    * Dispatched when the MediaStream is available.
+    * @see Event
+    * @name mediaStreamAvailable
+    * @event
+    * @memberof Subscriber
+    */
+
     /**
     * Dispatched when the subscriber receives a caption from the transcription service.
     * You must first call the <a href="Subscriber.html#subscribeToCaptions">Subscriber.subscribeToCaptions()</a>
@@ -66861,7 +66927,7 @@ function SessionFactory(deps) {
   /**
    * The client has disconnected from the session. This event may be dispatched asynchronously
    * in response to a successful call to the <code>disconnect()</code> method of the Session object.
-   * The event may also be disptached if a session connection is lost inadvertantly, as in the case
+   * The event may also be dispatched if a session connection is lost inadvertantly, as in the case
    * of a lost network connection.
    * <p>
    * The default behavior is that all Subscriber objects are unsubscribed and removed from the
@@ -94223,6 +94289,16 @@ var _default = _ref => {
       subscriber.dispatchEvent(new Events.AudioLevelUpdatedEvent(loudness));
     }
   });
+  const audioLevelUpdatedAdded = () => audioLevelRunner.start();
+  const subscriberDestroyed = () => audioLevelRunner.stop();
+  const addListeners = () => {
+    subscriber.on('audioLevelUpdated:added', audioLevelUpdatedAdded);
+    subscriber.once(_eventNames.default.SUBSCRIBER_DESTROYED, subscriberDestroyed);
+  };
+  const removeListeners = () => {
+    subscriber.off('audioLevelUpdated:added', audioLevelUpdatedAdded);
+    subscriber.off(_eventNames.default.SUBSCRIBER_DESTROYED, subscriberDestroyed);
+  };
   Object.defineProperty(subscriber, 'loudness', {
     get() {
       audioLevelRunner.start();
@@ -94237,16 +94313,14 @@ var _default = _ref => {
   if (subscriber.listenerCount(_eventNames.default.AUDIO_LEVEL_UPDATED)) {
     audioLevelRunner.start();
   }
-  subscriber.on('audioLevelUpdated:added', () => {
-    audioLevelRunner.start();
-  });
-  subscriber.once(_eventNames.default.SUBSCRIBER_DESTROYED, () => {
-    audioLevelRunner.stop();
-  });
+  addListeners();
   return {
     destroy() {
       audioLevelSampler.destroy();
       audioLevelRunner.stop();
+      removeListeners();
+      // eslint-disable-next-line no-param-reassign
+      subscriber = undefined;
     }
   };
 };
@@ -103310,6 +103384,8 @@ let SinglePeerConnectionController = /*#__PURE__*/function () {
   };
   _proto._parseOptions = function _parseOptions(options) {
     return Object.assign({}, options, {
+      // Create only one RTCDtlsTransport
+      bundlePolicy: 'max-bundle',
       // We always remove unused codecs after iceRestart in SPC
       removeUnusedCodecs: true,
       sendMessage: (type, payload) => {
@@ -103461,7 +103537,7 @@ let SinglePeerConnectionController = /*#__PURE__*/function () {
 
   // Singleton to make sure we are using only one PC when SPC. If SPC, we will add Subscriber
   // specific options to handle analytics and negotiation per Subscriber. This will take care
-  // to send the answer back to Rumor by its respective Subcriber and not multiple answers.
+  // to send the answer back to Rumor by its respective Subscriber and not multiple answers.
   // It will instantiate a new regular PC for all other cases.
   ;
   _proto.getPeerConnection = function getPeerConnection(opt, subscriberPc) {