* 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

Moves the logics which deal with JitsiConference events to a separate module.

JitsiConference should emit JitsiConference events instead of RTC events
when adding/removing tracks on jvb<->p2p switch. This behaviour was mixing RTC layer
with the conference one. When conference switches from jvb to p2p it wants to make
the application switch to use p2p JitsiTracks, but it doesn't mean that the JVB tracks
have been removed on the RTC/peer connection level.

* feat: use getSynchronizationSources on the receiver for remote audio levels
Use getSynchronizationSources if it is supported, fallback to using getStats otherwise.
* feat/ref: Use the local audio levels from LocalStatsCollector
When using getSynchronizationSources, use the audio levels from LocalStatsCollector for NoAudioSignalDetection.js
Remove obsolete code - TalkMutedDetection feature using audio levels is not used anymore

Used for extending owner privileges to other participants in the chat room.

This is useful for testing insertable streams related issues.

This was an experimental feature introduced for calculating the RTT to STUN servers, no longer needed.

Adds extra 15 seconds before emitting ICE failed event. If the OS is capable of
reporting internet offline status then will start counting when the internet
comes back online.

The reason for that is when ICE restarts are disabled a user will get the page reload
screen immediately, but there's a chance to recover as long as the channels have not
expired on the bridge.

Refactors the way send/receive video constraints are managed and puts the high level
logic in a separate module. See QualityController.js for high level overview on how
the constraints are managed now.

Adds events to JitsiConference fired whenever it starts new jvb/p2p session.

Adds event to JingleSessionPC.js when remote party signals receive max frame height.
Also adds signaling of the local recv preference for the p2p mode(only existed for JVB).

VAD processing causes audio input to fail on mobiles running latest versions of iOS/iPadOS.