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

Firefox lists stopped transceivers when getTranceivers is called, ignore these when picking a trancceiver for a screenshare track. Fixes an issue when starting screenshare fails if there are stopped tranceivers in the peerconnection (i.e., if some remote users with sources have left the call).

Fixes cases where SS fails in multi-stream mode with p2p->jvb->p2p switching.

In Unified-plan mode, we rely on the browser to fire 'negotiationneeded' event to start a O/A cycle. Also add muted tracks to PC. Fixes issues where the streams are not being signaled when call switches over from p2p to jvb in multi-stream mode.

This fixes an error that gets logged when no media is being sent/received by the endpoint.
Logger.js:154 2022-09-12T14:40:53.476Z [modules/statistics/AvgRTPStatsReporter.js] <xl.addNext>:  bandwidth_upload - invalid value for idx: 0

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

Avoid needing an external dependency.

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>

These have becomee necessary with the introduction of multi-stream.

> Possibly fixes a bug where SS appears black when the'unmute' and forwarded sources event for the track are received before a large-video update is scheduled.

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

* modules: add RFC 2198 audio redundancy encoder

which can be used together with WebRTC Insertable Streams to encode
a variable amount of audio redundancy

* port some tests

* port the rest of the tests but the comparison seems broken

* fix most lint

* with jasmine@4 actually fix the tests

* fix lint issues

* +jsdoc

* +nils review

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

It gets resolved during translation.

Fixes: https://github.com/jitsi/lib-jitsi-meet/issues/2055

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.

Since the track is only temporarily removed and added back, it can be treated like mute & unmute operation. Fixes https://github.com/jitsi/lib-jitsi-meet/issues/2048. Remove a TODO thats no longer needed, track replace operation don't force renegotiations anymore.

They are always required and the signaled features are ignored.

* feat: Add videoType to sources signaled in jingle.

* Add a receiveMultipleVideoStreams flag.

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 hack for making Chrome capture the highest resolution possible for desktop track breaks high fps screen capture.

* fix(SDP): don't separate fmtp parameter with spaces

* adjusted tests accordingly to not expect spaces anymore