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

* send transcripton_language only when necessary

* fix syntax

This lets the bridge adjust the bitrate allocation for this source so that layers with higher fps are prioritized over layers with higher resolution.
As a result, endpoints with restricted downlink will receive a high fps low resolution share as opposed to a high resolution low fps screenshare.

Provide a method for changing the capture fps for desktop tracks during the call. These changes to the lib are needed for making it configurable from the UI.

Set the initial video type state to NONE to cover cases when the endpoint joins video muted.

Disable cross browser p2p using unified plan until all the issues are fixed.

When started muted and no tracks are being created and we were muted by focus (av moderation) we need to unmute the channels on the bridge after creating the tracks.