Handle track changes when call was started without audio or video

When the call was started without audio nor video (either explicitly or
because there was an error when getting the media) the media device
changes were not listened to, so further device changes were ignored and
"localTrackReplaced" events were not emitted.

Moreover, "localTrackReplaced" events were handled by Peer objects to
replace or add tracks as needed, but if a call is started without media
some Peer objects will be missing (the ownPeer object if the HPB is
used, Peer objects for any other participant that is not sending media
when the HPB is not used). Therefore now a forced reconnection is
triggered when needed to establish the missing sender connections.

Signed-off-by: Daniel Calviño Sánchez <danxuliu@gmail.com>

src/utils/webrtc/simplewebrtc/localmedia.js

@@ -144,6 +144,10 @@ LocalMedia.prototype.start = function(mediaConstraints, cb, context) {
 		webrtcIndex.mediaDevicesManager.disableDeviceEvents()
 	}
 
+	// The handlers for "change:audioInputId" and "change:videoInputId" events
+	// expect the initial "getUserMedia" call to have been completed before
+	// being used, so they must be set when the promise is resolved or rejected.
+
 	webrtcIndex.mediaDevicesManager.getUserMedia(constraints).then(function(stream) {
 		// Although the promise should be resolved only if all the constraints
 		// are met Edge resolves it if both audio and video are requested but
@@ -197,6 +201,9 @@ LocalMedia.prototype.start = function(mediaConstraints, cb, context) {
 
 		self.emit('localStreamRequestFailed', constraints)
 
+		webrtcIndex.mediaDevicesManager.on('change:audioInputId', self._handleAudioInputIdChangedBound)
+		webrtcIndex.mediaDevicesManager.on('change:videoInputId', self._handleVideoInputIdChangedBound)
+
 		if (cb) {
 			return cb(err, null)
 		}
@@ -484,10 +491,8 @@ LocalMedia.prototype.stop = function(stream) {
 	this.stopStream(stream)
 	this.stopScreenShare(stream)
 
-	if (!this.localStreams.length) {
-		webrtcIndex.mediaDevicesManager.off('change:audioInputId', this._handleAudioInputIdChangedBound)
-		webrtcIndex.mediaDevicesManager.off('change:videoInputId', this._handleVideoInputIdChangedBound)
-	}
+	webrtcIndex.mediaDevicesManager.off('change:audioInputId', this._handleAudioInputIdChangedBound)
+	webrtcIndex.mediaDevicesManager.off('change:videoInputId', this._handleVideoInputIdChangedBound)
 }
 
 LocalMedia.prototype.stopStream = function(stream) {

src/utils/webrtc/webrtc.js

@@ -49,6 +49,7 @@ let callParticipantCollection = null
 let localCallParticipantModel = null
 let showedTURNWarning = false
 let sendCurrentStateWithRepetitionTimeout = null
+let startedWithMedia
 
 function arrayDiff(a, b) {
 	return a.filter(function(i) {
@@ -552,6 +553,8 @@ export default function initWebRTC(signaling, _callParticipantCollection, _local
 	})
 
 	webrtc.startMedia = function(token) {
+		startedWithMedia = undefined
+
 		webrtc.joinCall(token)
 	}
 
@@ -872,6 +875,8 @@ export default function initWebRTC(signaling, _callParticipantCollection, _local
 		// reconnection is forced to start sending it.
 		signaling.setSendVideoIfAvailable(true)
 
+		startedWithMedia = true
+
 		let flags = signaling.getCurrentCallFlags()
 		flags |= PARTICIPANT.CALL_FLAG.WITH_VIDEO
 
@@ -937,9 +942,38 @@ export default function initWebRTC(signaling, _callParticipantCollection, _local
 		}
 	})
 
+	webrtc.on('localTrackReplaced', function(newTrack /*, oldTrack, stream */) {
+		// Device disabled, nothing to do here.
+		if (!newTrack) {
+			return
+		}
+
+		// If the call was started with media the connections will be already
+		// established. If it has not started yet the connections will be
+		// established once started.
+		if (startedWithMedia || startedWithMedia === undefined) {
+			return
+		}
+
+		// If the call was originally started without media the participant
+		// needs to reconnect to establish the sender connections.
+		startedWithMedia = true
+
+		let flags = signaling.getCurrentCallFlags()
+		if (newTrack.kind === 'audio') {
+			flags |= PARTICIPANT.CALL_FLAG.WITH_AUDIO
+		} else if (newTrack.kind === 'video') {
+			flags |= PARTICIPANT.CALL_FLAG.WITH_VIDEO
+		}
+
+		forceReconnect(signaling, flags)
+	})
+
 	webrtc.on('localMediaStarted', function(/* configuration */) {
 		console.info('localMediaStarted')
 
+		startedWithMedia = true
+
 		clearLocalStreamRequestedTimeoutAndHideNotification()
 
 		if (signaling.hasFeature('mcu')) {
@@ -950,6 +984,8 @@ export default function initWebRTC(signaling, _callParticipantCollection, _local
 	webrtc.on('localMediaError', function(error) {
 		console.warn('Access to microphone & camera failed', error)
 
+		startedWithMedia = false
+
 		clearLocalStreamRequestedTimeoutAndHideNotification()
 
 		let message