More than one VP9 payload type is produced in the offer, more experimental ones if the local endpoint has no local video source. Munge the SDP to remove all codecs besides the one with profile-id=0 so that only one VP9 codec is negotiated with p2p peer.

Recommit 17ade96a70 with the jvb leak fix.

This is much faster than changing direction on the transceiver and doesn't need a renegotiation for the suspending/resuming media. Also fixes an issue with audio-only mode in p2p where media doesn't resume when audio-only mode is disabled if it was enabled while the jvb connection was active.

For renegotiations triggered by the application, it is ok to execute sRD->cA-sLD cycle on the p2p initiator instead of executing cO->sLD->sRD cycle. Chrome sometimes creates a third m-line for p2p when createOffer is called which is unexpected and therefore it breaks the client's renegotiation cycle. This has been done in other places already, i.e., when remote sources are signaled, etc. so making it uniform across all renegotiations.

Just moving them to the end of the list guarantees that they are not picked for encoding.

This reverts commit 17ade96a70.

It causes setParameters failures which can cause media being leaved to
JVB when in a P2P connection.

* fix(codec-selection) Use a configurable preferred codecs list to select the codecs.
Allow asymmetric codecs to be configured on the endpoints. This means that Firefox and Safari which have bugs with VP9 encode will now encode VP8 but decode VP9 coming in from Chromium endpoints.

* fix(codec-selection) Add unit tests for the codec selection logic.

* feat(codec-selection): Introduce mobileCodecPreferenceOrder setting in config.js.

* fix(codec-selection) Avoid unnecessary renegotiations.
On participant join/leave, check if the new intersection of codecs are already configured to be the top n codecs.

* Address review comments

* fix: Strip the codecs that are not in the codec list from the video m-lines.
Also, ignore remote codecs published in presence for p2p connections.

* fix: Define default codec order for mobile and desktop.
Fix an issue where p2p between mobile and desktop was broken.

* fix: Add default codecs for both p2p and jvb on mobile devices.
Munge the initial offer sent out by RN clients since RN doesn't support RTCRtpTransceiver#setCodecPreferences.

Fixes a bug where the remote video freezes if the remote user does an ICE restart using session-terminate. In this case, a new SSRC is assigned to the remote source which will have the same source-name.

Fixes an issue with Spot screensharing since Spot uses a fake conference to create a connection for screensharing and codecSettings are undefined.

* fix(stats) Obtain resolution/fps from 'outbound-rtp' stats.
'Track' based stats were dropped in Chrome 112, therefore send resolution and fps for the simulcast case needs to be calculated based on the 'outbound-rtp' streams that are currently active.

* squash: remove an unwanted log

* Squash: Address review comments.
Use outbound-rtp stats for both Firefox and Safari.
Firefox - Ignore active field if not present in the stats and calc fps using 'framesSent'.

Also add all the orphan jvb tracks to the conference when the call switches over from p2p to jvb.

* fix(TPC) Avoid applying constraints on desktop tracks if not needed.
Ignore sender constraints if the client is already sending video of the requested resolution. For desktop tracks, max resolution will be the height of the window being captured irrespective of the height being requested by the peer. Therfore, check if the configured resolution is equal to the track height for all requested heights > 0.
Fixes an issue where the track addition fails because of setParameters failing on the video track. This is only seen the torture tests because this is very timing specific.

* squash: check correct scaleResolutionDownBy value for p2p conn.

This fixes an issue where p2p clients (with different codec preferences) fail to decode video because the negotiated codecs are removed from the supported codecs list after the media session is established. The codec preferences will be applied when the first offer/answer is created.

This should avoid unnecessary renegotiations if disabledCodec setting is present in config.js and should improve time to receive media when joining big calls.

* 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(TPC): make screen share bitrate configurable

* fix(TPC): make local stream encodings configurable

* fix(TPC): fix lint error

* fix(TPC): alpha sort methods and update check for VP8

* 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.

Simulcast for SS is always on in unified plan mode, the lower spatial resolution streams are turned off for low fps sharing.

It now supports the sender encodings API.

(Rather than moving streams to inactive in SDP.)

Revert this change as this is causing a dummy track to be created when the media resumes on jvb connection after the p2p peer has left the call.

Currently, remote tracks are not created if presence for the endpoint is not received before source signaling. With ssrc-rewriting, the source information will be received on the bridge channel and presence on the prosody ws so there are chances that they can be out of sync. We do not want to skip remote track creation when that happens.

* 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.

Instead of Jicofo signaling all the remote sources available in the call, the bridge now signals only a limited set of SSRCs and then rewrites the SSRC on the outgoing media streams. The SSRC mapping is done based on the sources requested by the clients through the receiver constraints. This limits the number of m-lines in the remote/local SDPs on the client and therefore results in better performance in large calls.

* Handle source remapping messages from bridge
* Added track_owner_changed events
* don't process an invalid rtx ssrc.
* keep track of remote ssrcs, only renegotiate on new ones.
* Change source name on remote track on ssrc remapping.
* Don't remove tracks on member leave.
* Remove (orphaned) tracks on session terminated.
* Use serial number (per media type) to create msid attribute.
* Update videoType on remapping.

Co-authored-by: James A <jqdrqgnq@users.noreply.github.com>

* fix(codec-selection) Munge all related m-lines for multi-stream.
Munge all the video m-lines to apply the preferred codec settings in multi-stream mode.

* fix(TPC) Cache transceiver mids for all local video tracks.
There will be multiple video tracks in multi-stream, we need to cache the mids for all the the associated transceivers.

* fix(sender-constraints) Do not apply sender constraints if value hasn't changed for a given source.

* fix(TPC) Configure the direction correctly for all m-lines.
Ignore negotiationneeded events for p2p to avoid going into a renegotiation loop.

* squash: Address review comments.

Adds a helper function getSourceIndexFromSourceName to
have track index parsing in one place.

In _processLocalSSRCsMap there was an assumption that a track
index can be a single digit only.

Use "last index of" or search from the end of the collection to
make parsing a bit more universal (does not matter as much for
Jitsi endpoint IDs, but if fixes issues if someone has
a customization there).

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.

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

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.

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.

* feat: Add videoType to sources signaled in jingle.

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.