The bridge is now able to use DD headers for VP8 and VP9 as well so there is no need to remove it from remote desc when the call switches from AV1 to VP8/VP9.

Disables VP9 for FF since the simulcast  is disabled with FF flag by default. Introduces a few fixes so that if the FF flag is enabled simulcast works. This should also help us to be ready to enable VP9 once the  FF config flag for simulcast for VP9 is enabled by default.

For FF 136+ where DD extension is supported,  removing DD extension from the SDP will lead to broken scalabilty (the temporal scalability information will be missing and also DD extension header wont be included).

It doesn't get translated in the TS build, for one.

Script I used:

```python
import os

for (dirpath, dirnames, filenames) in os.walk('.'):
    if '.git' in dirpath:
        continue
    if 'node_modules' in dirpath:
        continue
    if 'dist' in dirpath:
        continue
    if 'types' in dirpath:
        continue
    for filename in filenames:
        path = os.path.join(dirpath, filename)
        if not path.endswith('.js') and not path.endswith('.ts'):
            continue
        #print(path)
        with open(path, 'r+') as f:
            #print(f)
            data = f.read()
            if '__filename' in data:
                p, ext = os.path.splitext(path)
                txt = f"'{p[2:]}'"
                print(txt)
                data = data.replace('__filename', txt)  # Assign the result back to data
                f.seek(0)
                f.write(data)
                f.truncate()
```

Aparently we call this for all tracks including for the audio tracks and it is failing with JS error. This leads to "participant unable to unmute".

Firefox added support for AV1 with DD and VP9 SVC is also working as expected in Firefox 136.

* fix(codec) Debounce the call that calc codec intersection set.
Calculate codec intersection set only once per second even when there is a burst of joins/leaves. Also, check for current codec before chaining a codec change operation when codec selection API is used.

* squash: Address review comments

Regression was introduced in 07f037e12e/. Also process encodeTime stats when H.264 is selected

The constructor for RTCSessionDescription has been deprecated - https://developer.mozilla.org/en-US/docs/Web/API/RTCSessionDescription/RTCSessionDescription.

* ref(SDPDiffer) Convert to ES6 class.
Make it work directly with unified plan SDP that has multiple m-lines and add more unit tests.

* ref(xmpp) translate unified-plan SDP->Jingle directly.
Without having to run it through the SDPInterop.toPlanB cycle.

* ref(SDP) Always generate the MSID for signaling it to Jicofo.

* fix(SDPDiffer) Check explicitly for ssrc changes

* fix(SDP): Fix comments and cleanup.
Remove LOCAL_TRACK_SSRC_UPDATED event as the application ignores the event and no additional action needs to be taken when that event is fired.

* ref(SDP) Add a helper function for parsing the 'a=ssrc-group' line.

* squash: Address review comments

* ref(QualityController) Add recovery mechanism and adjust the resolution check.
Impl a recovery mechanism for the lastN to be increased if the cpu limitation goes away and doesn't return after increasing lastN. Also, additionally improve the calculation of the expected resolution taking simulcast stream resolutions into account.

* squash: Address review comments and add more unit tests

* squash: Address review comments

* feat(quality) Add a QualityController class for runtime adjustments.
Make run time adjustments to the client when adaptive mode is enabled.

* feat: Update lastN and receive resolution to improve quality.

* squash: Address review comments

* squash: Add more logging and address review comments.

Also fixes the issue with video freezing after video mute and unmute when the user joins the call with startSilent=true.

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

Firefox 123 started throwing an error when the transction ids for getParamters and setParameters don't match. This can result in Firefox not establishing a p2p connection with a remote peer when it is the offerer.

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

When both the camera and screenshare tracks are present. Add unit tests.

Set the min. required version for Firefox to 91 so that the last 3 ESR versions are supported.

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.

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

The simulcast receive lines needs to be inserted in the receive desc for all the m-line associated with local tracks, currently its being done only for the first video source.

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.

Fixes an issue where screenshare from Firefox appears blurry on other endpoints in the call.

In the older versions, the SSRCs in the SDP were in reversed order, i.e., the resolution order was 1:2:4 as opposed to Chromium and other browsers. Therefore, the client was confuguring the reverse order. However, the commit https://hg.mozilla.org/mozilla-central/rev/b0348f1f8d7197fb87158ba74542d28d46133997 which was seems to have landed in Firefox 117 has reverted the reversing of the RID order. Therefore, this hack is no longer needed on the newer Firefox versions.
This fixes an issue where the bridge suspends all video from Firefox to other endpoints in the call because of the reversed order of the encodings.

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.

This reverts commit 17ade96a70.

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

Fixes blurry screenshare in some cases. There is a higher probability of the higher layers getting suspended when all the stream encodings are enabled for low fps SS. Make sure only the highest spatial layer is sent for low fps SS after p2p->jvb switch. Make sure all the stream encodings are enabled for high fps SS after p2p->jvb switch.

* 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.

* 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

* 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.

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).

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).