* 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.

The tracks will not be added when the call switches from jvb to p2p for an endpoint that joins muted by focus.

* feat: Audio/Video moderation.

* squash: Fix docs.

* squash: Adds some warning logs when execution is rejected.

* squash: Changes a field name in the message for adding jid to whitelist.

* squash: Send to participants only message about approval.

Skips sending the whole list.

* squash: Fixes tests.

* squash: Adds more logs.

* feat: Separates enable/disable by media type.

Adds actor to the messages to inform who enabled it.

* squash: Fixes log line.

* squash: Fixes comments.

* squash: Fixes log messages.

* squash: Fixes comments.

* feat: Video type camera is default value of missing, skip it in presence.

We skip sending initial video type camera if it is already missing from presence, if it changes we send the last value.
We also are dropping the namespace as it is not used anywhere and just increases presence size.

* feat: A/V muted is default so we can skip adding it initially to the presence.

Dropping the namespace as it is not used anywhere and that reduces size.

* squash: Drops unused setting.

* squash: Adds a config option to enable skipping the muted state.

Use the new Colibri message "EndpointStats" for broadcasting the local stats. The bridge then will be able to filter the endpoint stats and send them only to the interested parties instead of broadcasting it to all the endpoints in the call.

* Feature: Moderator can revoke moderator role to others and himself

* Code review inputs

* Code review inputs

* fix(TPC): Configure degradation preference in RTCRtpSendParameters.
Properly configure degradation preference on RTCRtpSendParameters instead of RTCRtpEncodingParameters. Fixes https://github.com/jitsi/lib-jitsi-meet/issues/1510.

* feat(JingleSessionPC): Remove ssrcs from remote desc when a user leaves.
Remove the ssrcs (associated with remote sources) from the remote desc along with the removal of the remote tracks when an endpoint leaves the call. The source-remove signaling message from Jicofo will no longer be needed in this case and can be dropped.

Add the ability to send the bridge messages for the receiver video constraints in the new format directly.

Apple added getUserMedia support for WkWebview based browsers like chrome and Firefox on iOS 14.3. These browsers behave as Safari does on iOS. Therefore, extend the Safari checks to these webkit based browsers as well.

Do not add the muted audio/video tracks to the peerconnection on join. The tracks will be added when the user unmutes for the first time. This reduces the number of remote sources that will be added when a participant joins a large call where everyone joins muted (startAudioMuted/startVideoMuted setting).