* feat(multi-stream-support) Add TrackStreamingStatus class
update JitsiRemoteTrack to init and dispose TrackStreamingStatus
stop emitting LASTN_ENDPOINT_CHANGED event when source name signaling is enabled
convert TrackStreamingStatus class to typescript

* rename methods and use TrackStreamingStatus enum

* update jdocs for JitsiConferenceEvents.FORWARDED_SOURCES_CHANGED

* feat(multi-stream-support) Adjust presence and videoType message.
Adjust the sourceInfo for all the available local tracks. Do not add handlers for videoType changes on the peer since the videoType doesn't change when multiple stream support is enabled.

* fix(presence) Send the presence in old format for backwards compat.
Send the presence messages in both the old and new format when source name signaling is enabled and sending multiple streams is disabled.

* fix(JitsiConference) Send the videoType bridge msg after mute/unmute op is complete.

* ref(multi-stream-support) Throw if a track is being replaced with a track of different videoType.
This is not supported in the multi sender enabled mode.

* Address review comments.

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.

* 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

Other minor formatting changes.

After the refactor in
5a4232f908
the succesful terminating reason is expressed by the 'success' value in the
`reason`parameter, rather than having it undefined.

Ref: https://github.com/jitsi/lib-jitsi-meet/issues/1793#issuecomment-989913626

* 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.

Rewrite the function using async / await and reverse the logic to make i easier
to read / understand.

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.

They are companion rooms created in a separate MUC. The room relationship is
maintained by a Prosody plugin.

All signalling happens through the breakout rooms MUC component.
Co-authored-by: Saúl Ibarra Corretgé <saghul@jitsi.org>

* feat: facial expression in speaker stats

* feat: send xmpp message with facial expression to server

* fix: rebase conflicts

* feat: facial expression in speaker stats

* feat: facial expression in speaker stats

* fchore(facial-expressions): remove the send facial expression call from update facial expression function

* feat(facial-expressions): store expresions as a timeline

* refactor(facial-expressions): store expressions by counting them in a map

* feat(facial-expressions): camera time tracker

* add(facial-expression): increase facial expression with duration parameter from payload

* add(facial-expressions): the disgusted expression

* refactor(facial-expressions): remove camera time tracker

* refactor(facial-expressions): change data channel message handel position for facial expressions and renamed some types

* fix(facial-expressions): move facial expression endpoint message handler from statistics.js to ConnectionQuality.js

* fix(facial-expressions): remove unused type

This is a stopgap measure. Ideally we'd support it, but the interactions between
both JVB and P2P sessions are complex and very timing sensitive and there are a
number of corner cases that lead to not having media after toggling e2ee.

Make sure the P2P tracks are not added to the JVB session when we are restarting
the media sessions, since the PC has not been created with the encoded streams
constraint yet.

The _MEDIA_SESSION_STARTED would be emitted with `null` as the session, instead
of the original one.

Sends the source name to video type mapping to the jvb
in a new bridge channel message.

* fix: Avoid sending two presences if start muted and then screen share.

* squash: Drop some changes and simplify sending presence.

* squash: Always send presence when sourceNameSignalingEnabled.

...to advertise track's muted state and the video type.

For now, if the source name signaling flag is enabled, both legacy and
new <SourceInfo> element will be used at the same time. This is to be
able to interoperate with legacy clients and start testing the new
format at the same time. Whenever possible <SourceInfo> will be used
as main source of truth with the fallback to legacy <audiomuted/>,
<videomuted/> and <videotype/> elements.

We do not have to wait for the removal of the ssrcs from the remote description for removing the remote tracks associated with a participant that left the call. This speeds up removal of the participant from call even if the JingleSession modification queue is backed up.

Increase the RTC mute timeout from 500ms to 2500ms for p2p connections. This fixes an issue with Chrome tab sharing where the application keeps switching between the avatar and the share contnuously because of a chrome bug https://bugs.chromium.org/p/chromium/issues/detail?id=1258034

We need to make sure the audio track is added to the JVB connection, or we won't
be able to unmute.

Why this happens is a mystery wrapped in an enigma, it started happening with
iOS 15.

Same thing applies to Safari for macOS.

Fixes: https://github.com/jitsi/jitsi-meet/issues/10104

If a second JitsiConference instance for the same room is created we were
throwing an error but some listeners were already attached. This commit
makes sure that we throw the error as soon as possible and no listeners
are added.

* feat(av-moderation) Remove from moderation whitelist functionality

* Fix comment

Credit to Stephen Donner.

This swallows the stack trace.

If oldTrack was not previously in the conference this will
lead to a crash in onLocalTrackRemoved where oldTrack doesn't
have `muteHandler` and `audioLevelHandler` listeners defined.

There is no need to check if we need to sswitch from P2P to JVB or vice versa if
we have already left the room. It creates a log flood.

On mobile Safari, if a user joins audio and video muted, the browser doesn't decode the incoming audio. Workaround is to always add the audio track to pc and mute it if needed.

Fixes error when trying to stop screen sharing and video moderation is on

If audio levels are disabled it's pointless to try and detect problems.

* enable config callstats configParams

* add comments

* wrap long comments

* fix format

* fix format

* fix format

* add jsDoc