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).

* Added video mute participant

* Trigger mute event

* Optimized mute type checks

* Fixed event name

* Fixed some linter issues

* Fixed more linter issues

* And even more linter issues fixed

* And more linter fixes

* Added media type to analytics event

* feat: Drops unused code about handling incoming caps changes.

* fix: Drop using private fields outside JitsiParticipant.

All the send video constraints for the client, i.e., what simulcast streams will be enabled based on constraints received on the bridge channel, will be handled by the SendVideoController class.
The receive video constraints like lastN, selectEndpoints and receive video resolution will be handled by the ReceiveVideoController class.

"Pinned" endpoints are deprecated, use "selected" instead.

Force the client to reload when the bridge that is handling the media goes down.
This mitigates issues seen on the bridge because of a client re-joining the call with the same endpointId, BWE issues, etc.
This behavior is configurable through 'enableForcedReload' setting in config.js.

Determine the preferred codec for a given endpoint based on the config.js settings and the codecs supported by the endpoint.
The preferred codec is published in presence and then used by the other endpoints in the call during join/leave to determine
if the codec needs to be changed on the fly. Different codecs can be configuered for p2p/jvb connections.
The preferredCodec/disabledCodec settings under videoQuality will have precedence over the older settins like preferH264/disableH264.

* fix: Drops unused parameter of join and sendPresence.

* feat: Skips using disco-info for features.

Uses presence to publish features added externally.
Recognizes jigasi participants using a specific presence extension used by jigasi.

* squash: Fixes tests.

* squash: Adds e2ee to the features in presence.

* squash: Fix function name and docs.

* squash: Drops detecting jigasi from initiator.

Using the newly added presence features.

* squash: Drops unused var.

* squash: Fix comments.

* squash: Adds a constant and for E2EEE feature.

There are cases where we start p2p on the first participant received presence, where there are more presences coming. This starts the process of p2p resulting and few iq errors while terminating it quickly after starting it.
On a normal joining as third participant we see around 38 send or received iq messages, in the problem scenario we see 60.

Only one audio or video track is correctly supported by lib-jitsi-meet.
This is an attempt to mitigate an issue reported by the community
where the user's audio is played in the conference even though the UI
shows the user as muted.

Also return early to avoid weird errors if the initialization step fails.
Everything is broken at that stage pretty much.

Adds a JitsiConference.getSsrcByTrack() method that finds the SSRC of a JitsiTrack by its SSRC. This method is currently used to get (and display) the first audio/video SSRC in the GSM popover in Jitsi Meet.

Co-authored-by: George Politis <gp@jitsi.org>

This the second stage in our E2EE journey.

Instead of using a single pre-shared passphrase for deriving the key used for
E2EE, we now establish a secure E2EE communication channel amongst peers.

This channel is implemented using libolm, using XMPP groupchat or JVB channels
as the transport.

Once the secure E2EE channel has been established each participant will generate
a random 32 byte key and exchange it over this channel.

Keys are rotated (well, just re-created at the moment) when a participant joins
or leaves.

...which contains conference duration in seconds and
performance stats (currently long tasks).

* ref: Moves xmpp logs to be accessed from connection.

In cases where there is no room like pre-join and lobby screen we still want to be able to debug xmpp messages.

* squash: Remove getLogs from Conference.

* squash: Fix catch.

* squash: Print the error.

Add a performance stat around long tasks. Chrome supports PerformanceObserver API that lets us
register for long tasks event. Any task that takes longer than 50ms is considered a long task.

With the introduction of lobby the ChatRoom KICKED event is now also emitted for
ourselves (the kicker) so we want to avoid emitting an event where `undefined`
kicked someone.

Delays e2e encryption initialization until any encryption key is set.
Recreates peerconnections in order to enable insertable streams
only if the e2ee is used. This is to avoid bug around insertable streams
which may cause audio issue when the main JavaScript thread is loaded:

https://bugs.chromium.org/p/chromium/issues/detail?id=1103280