s/updateCurrentKey/updateCurrentMediaKey/g

Track based stats have been removed recently and RTCRtpReceiver:getSynchronizationSources has been available in the browsers for a while now.

* fix(TPC) Disable media instead of changing dir for p2p->jvb switch.
Resume or suspend the media on the jvb peerconnection by changing the RTCRtpEncodingParamters.active state instead of changing the direction on the transceiver. This avoids the needs to start a O/A renegotiation cycle for these operations. The media direction will be changed only for p2p lastn=0 case since video needs to be disabled on both the sender and the peer for p2p lastn=0 case.

* Address review comments

* Disable media after adding source while media is suspended on the jvb connection. Default 'active' state for stream encodings after the source is added is 'true'.

* Wait for all the promises to be settled before returning

* fix(SignalingLayer) Update SSRC owners on leave.
Update the SSRC owners in the following cases:
1. When a remote endpoint leaves the call.
2. When a source-remove is received.
3. When a source is remapped (with ssrc-rewriting enabled).
Create the remote track even if presence is not yet received. The ssrc owner check prevents the client from creating a dummy track when the call switches over from p2p to jvb when the last remote endpoint leaves the call.

* ref(SignalingLayer) alpha sort methods.
Clean up unused methods, _findEndpointSourceInfoForMediaType is not used anymore.

* squash: Address review comments.

When the application doesn't set receiver constraints, generate default constraints in the source-name mode.
Cleanup code related to the endpoint based video constraint handling for p2p connection.

Endpoint based receiver constraints and other endpoint based bridge signaling messages are no longer supported by latest JVB after the switch to source-name signaling.
Rename method names 'sendNewReceiverVideoConstraintsMessage'->'sendReceiverVideoConstraintsMessage', 'setNewReceiverVideoConstraints'->'setReceiverVideoConstraints'

Bumps [minimatch](https://github.com/isaacs/minimatch) from 3.0.4 to 3.1.2.
- [Release notes](https://github.com/isaacs/minimatch/releases)
- [Commits](https://github.com/isaacs/minimatch/compare/v3.0.4...v3.1.2)

---
updated-dependencies:
- dependency-name: minimatch
  dependency-type: indirect
...

Signed-off-by: dependabot[bot] <support@github.com>

Delete hasAnyVideoTrackWebRTCMuted method as it is not used anymore.

* Remove support for legacy endpoint based signaling and make source-name signaling as the only signaling mode. The legacy screensharing mode will no longer be supported. The client will still be able to process remote presence sent in the old format to support interop with very old mobile clients and jigasi that do not support source-name signaling.
* remove code related to presenter mode.

fix: corrected error message (see 6810839edc)

* feat(multi-stream) Enable multi-stream by default (if not disabled explicitly).

It's only used for the UMD bundle.

It does the minimum necessary to function without the complexity of the
main one.

Fixes cases where SS stream is not being sent when p2p->jvb switch happens.

Allows a moderator to end the conference for everyone.

Co-authored-by: Werner Fleischer <70745309+wernf@users.noreply.github.com>

This fixes an issue where the bitrates for screenshare were much higher than before for VP9 causing the JVB to suspend SS streams more often.

Co-authored-by: Дамян Минков <damencho@jitsi.org>

When an answer is received from the p2p peer, only set remote description instead of calling the full renegotiation cycle. Also, always initiate a sRD->cA->sLD cycle when adding remote sources irrespective of whether the local endpoint is an initiator or responder for the p2p connection. This lets us remove the hacks added for Chromium (unified-plan) bugs related to video not rendering in p2p cases.

fix up test failure by including extends Listenable

It does no harm on RN, since it doesn't detect it as a supported browser
and bails out.

Since Track effects and mute/unmute operations use the same flow, i.e., removing/adding the track from the RTCPeerConnection but not from TPC, make the names of the methods involved more generic.

* feat: Add videoType to sources signaled in jingle.

* Add a receiveMultipleVideoStreams flag.

It's not necessary to perform a full join, sending a conference IQ is
enough.

Set the bitrates on the video m-lines based on the associated track type. In multi-stream mode, there will be two separate m-lines for camera and desktop tracks. Also fallback to SDP munging for setting codec preferences on Chromium.

the editor will no longer complain about superfluous await when calling