Check if all the relevant fields are correctly configured instead of checking for just the encode resolution on the local video sender.

Chrome on macOS recently started offering encoder for higher level (5.2) but decoder only for level 3.1. See https://issues.chromium.org/issues/324930413
Therefore, filter out all H.264 payload types with main and high profiles. Also, sort all H.264 payload types so that same pt is picked for both H.264 encoder and decoder. Fixes random black tile issues across different browsers when H.264 is the preferred codec for p2p.

We do not want non Chromium browsers to negotiate DD ext headers when VP8/VP9 is the selected codec. Fixes poor video quality issue for Safari when Av1 is offered by Jicofo.

Require cycle: lib/lib-jitsi-meet/modules/RTC/RTC.js -> lib/lib-jitsi-meet/modules/RTC/TraceablePeerConnection.js -> lib/lib-jitsi-meet/modules/RTC/RTC.js

Set the scalabilityMode to undefined for VP8 after a VP9->VP8 switch even though the browser returns L1T2 in the RTCRtpSender.getParameters() call. Fixes an issue where the bridge stops forwarding video for some participants after vp8->vp9 switch.

The browser returns height as 0 at track creation time. Fixes an issue where Safari's screenshare shows as black.

Check the first payload type of the m-line in remote description to get the codec used for encoding the local video source.

Fixes https://bugzilla.mozilla.org/show_bug.cgi?id=1877675.

If MediaStreamTrack.getSettings() doesn't return the height of the track, use the constraints passed to gUM for calculating the height. Fixes an issue for Firefox/Linux which could result in media not displayed.

When a local video track is muted, getSettings() on MediaStreamTrack doesn't return capture height. Fixes random torture test failures on beta.

* fix(video-quality): Adjust encodings bitrates/scalefactor for high capture resolutions.
When the client is configured to capture camera sources at 1080p or higher, adjust the bitrates and the encodings scale factor accordingly.

* squash: Address review comments
Fix the bitrates set on the remote description and also bump up the bitrates for 1080p, 4k and screenshare.

* squash: Address some more review comments.

This check is not needed anymore since we now set the correct scaleResolutionDownBy value everytime the encodings are configured. Fixes a regression on Safari where it can end up sending low resolution to a p2p peer in some cases. Also, add more unit tests for p2p desktop share cases.

* fix(RTC): Remove support for plan-b SDP.
Since unified plan mode becomes the default mode, so does multi-stream mode.

* squash: Remove plan-b unit tests and fix comments

* squash: address review comments

When these are configured, the browser sends only the lowest resolution layer even when all the encodings are active. Fixes a quality issue on older versions of Chromium that do not support the scalabilityMode API.

There is a lot of additional noise when log level is set to debug, the SDP descriptions should be set to trace (As it seems it was originally intended, based on the name of the variable)

This fixes an issue when encodings for video streams are configured incorrectly after every codec switch

Add SVC and Simulcast support for AV1 and VP9.

The default mode is the full SVC mode for VP9. It works as expected with the current version of JVB.
AV1 in the client is only supported when the bridge negotiates the Dependency Descriptor extension headers which are needed for reading the layer information for AV1 and H.264 video frames.

* feat: Enable H.264 simulcast support.
Multi-encoding simulcast for H.264 is supported now because of the DD header ext support added on the bridge side.

* fix(video-quality): Add default bitrates for all codecs.
Expect the videoQuality settings in the new format, deprecated configs are still supported.
videoQuality: {
    AV1: {
        maxBitratesVideo: {
            low: 100000,
            standard: 300000,
            high: 1000000,
            ssHigh: 1500000
        },
        useScalabilityModeAPI: {
            enabled: true,
            useSimulcast: true, //(defaults to SVC),
            useKSVC: true //(defaults to L3T3_KEY)
        },
    },
    H264: {
        maxBitratesVideo: {
            low: 200000,
            standard: 500000,
            high: 1500000,
            ssHigh: 2500000
        }
    },
    VP8: {
        maxBitratesVideo: {
            low: 200000,
            standard: 500000,
            high: 1500000,
            ssHigh: 2500000
        }
    },
    VP9: {
        maxBitratesVideo: {
            low: 100000,
            standard: 300000,
            high: 1200000,
            ssHigh: 2500000
        },
        useScalabilityModeAPI: {
            enabled: true,
            useSimulcast: true, //(defaults to SVC),
            useKSVC: true //(defaults to L3T3_KEY)
        },
    }
}

* Disable encodings when requested height for screenshare is 0.

* feat: add unit tests for scalability modes.

* ref: Move all encoding configuration calculation to TPCUtils.

* Add more unit tests for VP9 K-SVC mode.

* Make codec name in settings case insensitive.
Also read the deprecated max bitrates correctly, add unit tests to test it.

* ref(TPC): isSimulcastOn -> isSpatialScalabilityOn.
It makes more sense to call it spatial scalability than simulcast now that full SVC support is available.

* fix: Negotiate AV1 DD header exts only for AV1 and H.264.

* fix: Rename setting useL3T3Mode->useKSVC and adjust the defaults.

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.