The encodings for local video tracks are confiugured based on the SenderVideoConstraintsV2 constraints received from the bridge. The constraints are sourceName based. Also ignore enableLayerSuspension flag since it has been deprecated and layer suspension has been enabled by default. Also having it off makes screensharing on Chrome in unified plan not work as expected.

Subsequent attempts to join will show produce an error "[JitsiConference.js] : A coference with the same name has already been created!".

* feat(multi-stream-support) Add the support for multiple local video streams.
This feature is behind 'sendMultipleVideoStreams' config.js flag and is currently supported only on clients running in Unified plan mode.

* squash: Use the track index from source name to find the matching transceiver on unmute.

Messaging methods in lobby room
getLocalId to get the local lobby room id from jid
setMessageListener to listen for messages in lobby room

Reviewed-by: Fecri Kaan Ulubey <f.kaan93@gmail.com>

ref(lobby chat) refactor lobby chat api to conform to other existing lib-jitsi-meet apis

* feat(multi-stream-support) Handle SDP munging for multiple local/remote streams/ep.
A new SDPSimulcast class is added for handling SDP munging for local and remote descriptions to handle multiple streams for both local and remote endpoints. This new class will be used only for unified plan implentation. The sdp-simucast npm package will be used for plan-b and deprecated later when all the clients switch to unified plan.

* fix(build) fix building with TS code

Use the Babel preset for the webpack bundle since Babel 7 already
understands TS and ts-loader uses tsc instead, which we configure
differently.

* squash: Add unit test and address review comments.

Co-authored-by: Saúl Ibarra Corretgé <saghul@jitsi.org>

When a user leaves, the ssrcs associated with the ep are removed from the remote description and a renegotiation is forced. The browser then fires 'removetrack' event on the associated MediaStream when sRD is called with ssrcs removed. The remote track is then disposed of in the event handler. Therefore, there is no need to delete the tracks at the TPC level before renegotiation. This also prevents the 'Removed track not found for stream' events from showing up in the browser console log.

Add the _onPrivateMessage handler when breakout_rooms is found in disco info.

Regression introduced in
5c9b905111

Emitting declarations generated these errors:

~~~~
modules/transcription/audioRecorder.js:3:1 - error TS9005: Declaration emit for this file requires using private name 'TrackRecorder'. An explicit type annotation may unblock declaration emit.

3 const RecordingResult = require('./recordingResult');
  ~~~~~

modules/transcription/recordingResult.js:13:1 - error TS9005: Declaration emit for this file requires using private name 'RecordingResult'. An explicit type annotation may unblock declaration emit.

13 const RecordingResult = function(blob, name, startTime, wordArray) {
   ~~~~~

modules/transcription/transcriber.js:1:1 - error TS9006: Declaration emit for this file requires using private name 'SphinxService' from module '"/Users/saghul/work/jitsi/lib-jitsi-meet/modules/transcription/transcriptionServices/SphinxTranscriptionService"'. An explicit type annotation may unblock declaration emit.

1 const AudioRecorder = require('./audioRecorder');
  ~~~~~

modules/transcription/word.js:7:1 - error TS9005: Declaration emit for this file requires using private name 'Word'. An explicit type annotation may unblock declaration emit.

7 const Word = function(word, begin, end) {
  ~~~~~

Found 4 errors.
~~~~

Fixes https://github.com/jitsi/jitsi-meet/issues/10754.

Since the order of the ssrcs in the json-encoded message is not guaranteed to be in the correct SIM/FID order, log all the ssrcs.

* ref(JingleSession) Recycle m-lines by rejecting them on source removal.
By rejecting the m-lines on source removal (setting the port to 0 on the associated m-line), m-lines can be recycled for new sources instead of appending new ones.

* chore(deps) update sdp-interop@latest.

This fixes a regression introduced by the previous commit.

sLD is called immediately after createOffer/createAnswer, therefore the desc provided by createOffer/createAnswer can be directly passed to sLD without the need for converting it to unified plan format. This also fixes a warning seen on the browser console that says 'The description does not look like plan-b'.

The issue currently affects browsers based on Chromium 96.0.4664.110
including Brave, Chrome, Edge & Opera and Google Chrome 97.0.4692.56

* fix: Reads lobby jid from correct location in the error.

* squash: Add todo comment.

* feat: Adjust setReceiverConstraints to use new format

* add ReceiveVideoController spec

* rename prioritizedSources to onStageSources

* call FeatureFlags.init in specs and update description

* provide a default flags argument to FeatureFlags.init

* fix(RTC): fixed typo in function name

* fix(xmpp): codec renegotiation task needs to be queued

* fixed typos

* fix(tpc) disable ulpfec on chrome 96

* ref(tpc) add comment

Broken with previous commit 579a49.

* feat: Adds events for session-add, session-remove and session-accept.

* feat: Refactor updating presence for audio/video mute and video type.

This brings few changes and fixes. Thew initial presence will always miss audioMuted and videoMuted value, those will be added on session accept. All updates to presence are done on sessionAccept, on source-add or on source-remove and the only one not signalled when camera track is replaced by video track and vice versa.
This change brings more presence updates when number of participants are below startAudioMuted/startVideoMuted, but when participants are above those numbers we get less presence update. This is important for big meetings.
Fixes wrong videoMuted state, as replace track and mute are both executed in promise, sometimes the replace-track one finishes first and when the mute one is resolved there is no conference object in the track to be able to update the presence (hitting this when we pass the startAudioMuted threshold).

* squash: Put back the tracks for _setTrackMuteStatus and _setNewVideoType.

* squash: Fix line length.

* squash: Fix skip sending presence twice.

* squash: Adds the error to the error callback.

* squash: Fix newJingleErrorHandler.

Promise.all will be rejected with the first rejected Promise, without waiting
for the rest, we don't want that.

If leaving a room results in a timeout, we wouldn't be cleaning up the EMUC
state, which will prevent us from joining the room again, as it would detect it
as a duplicate.

Both of this features listen for events in the XMPP connection. Thus, if we join
another meeting over the same connection we might still receive some events on
the old event handler.

Stop listening for events from the components as soon as the leave operation is
triggered.

Jicofo sends a presence when the audio/video sender limit is reaced in the conference. The client can then proceed to disable the audio and video mute buttons when this occurs.

This is necessary when adding new properties to the payload.

This fixes an issue where a user is not able to unmute their video if the MediaStreamTrack associated with the camera stream returns a null value for the track height.