Fixes https://github.com/jitsi/jitsi-meet/issues/10754.

This fixes a regression introduced by the previous commit.

sLD is called immediately after createOffer/createAnswer, therefore the desc provided by createOffer/createAnswer can be directly passed to sLD without the need for converting it to unified plan format. This also fixes a warning seen on the browser console that says 'The description does not look like plan-b'.

The issue currently affects browsers based on Chromium 96.0.4664.110
including Brave, Chrome, Edge & Opera and Google Chrome 97.0.4692.56

* fix(RTC): fixed typo in function name

* fix(xmpp): codec renegotiation task needs to be queued

* fixed typos

* fix(tpc) disable ulpfec on chrome 96

* ref(tpc) add comment

Broken with previous commit 579a49.

This fixes an issue where a user is not able to unmute their video if the MediaStreamTrack associated with the camera stream returns a null value for the track height.

* Save track source name to JitsiRemoteTrack

* add feature flag checks. code review adjustments.

* eslint cleanup1

* eslint cleanup2

* fix: move sourceName declaration outside of if block.

Co-authored-by: Dennis Dowhy <ddowhy@gmail.com>
Co-authored-by: William Liang <wliang67@bloomberg.net>

This ensures that the source signaling is sent before the mute state is sent in presence. Jicofo relies on mute state from presence to check if the sender limit has been reached.

Sends the source name to video type mapping to the jvb
in a new bridge channel message.

Add timestamp to all tracks, and display surface to screen sharing tracks

This fixes occasional failures of MuteTest.MuteAfterJoinCanShareAndUnmute torture test and also the case on Safari where user stopping the screenshare doesn't stop showing the screensharing indication on the thumbnail.

...to advertise track's muted state and the video type.

For now, if the source name signaling flag is enabled, both legacy and
new <SourceInfo> element will be used at the same time. This is to be
able to interoperate with legacy clients and start testing the new
format at the same time. Whenever possible <SourceInfo> will be used
as main source of truth with the fallback to legacy <audiomuted/>,
<videomuted/> and <videotype/> elements.

Fixes https://github.com/jitsi/jitsi-meet/issues/9988.

Instead rely on the 'negotiationneeded' event fired by the browser for JVB connection. This makes local source changes faster even if the modification queue is backed up.

* ref(TPC) Configure encodings in one go.
Configure the active state, maxBitrate and degradation preference in one go instead of configuring them separately.

* squash: Address review comments.

* squash: fix a comment.

* feat(CodecSelection) Disable VP9 on Safari.
Safari does not decode VP9 properly- https://bugs.webkit.org/show_bug.cgi?id=231071 and https://bugs.webkit.org/show_bug.cgi?id=231074.

* squash move VP9 support check to browser capabilities.

ChatRoom.setAudioMute and ChatRoom.setVideoMute are
not async, so no callbacks nor promises are necessary.

It makes more sense to call it pcConfig since it is the RTCConfiguration object passed to the WebRTC peerconnection.

Log only the important events that we care about on the HTMLMediaElement that the remote tracks are attached to.

This fixes an issue where the bridge doesn't forward the HD stream from Firefox to other users in the call. The order of the ssrcs produced by the browser is from Highest resolution to lowest whereas the bridge assumes it to be from lowest to highest as is the case in Chrome and Safari.

If Jicofo doesn't offer VP9 but the client expresses a preference for VP9, VP9 bitrates were being imposed before.

* fix: re-enable that ICE connection failures in Chrome are detected
* fixed linting errors

In unified plan we'll take the ssrcs off the first audio / video stream, but we
were assuming both would be present.

This commit fixes that, since the Spot wireless screen sharing feature uses a
dedicated TPC just with a video stream.

This lets the bridge adjust the bitrate allocation for this source so that layers with higher fps are prioritized over layers with higher resolution.
As a result, endpoints with restricted downlink will receive a high fps low resolution share as opposed to a high resolution low fps screenshare.

remove legacy APIs which are not even supported anymore

* fix: Missed SSRCs in Unified Plan with several "ssrc-group:FID" groups.

`TpcUtils.ensureCorrectOrderOfSsrcs()` takes into account only the first `ssrs-group`, and it causes missed `ssrs` attributes. It results in an error during `peerconnection.setRemoteDescription()`.

For fixing `ensureCorrectOrderOfSsrcs()`, all `ssrs-group` are traversed and all distinct `ssrs` are preserved with keeping their order.

* test: Added extra tests for `TpcUtils.ensureCorrectOrderOfSsrcs()`

Munge 3 ssrcs in the SDP for chrome in unified plan always for the simulcast case.

Do not enable unified plan for p2p chrome by default until StartMutedTest is fixed.
Fix media direction for case when there are no local and remote sources, should be set to 'inactive' in that case.

Munge only the m-line that corresponds to the source that the browser will be sending.
Do not select VP9 on Firefox.
Detect support for RTCRtpTransceiver#setCodecPreferences correctly.

In unified plan impl, it is not possible to enable/disable simulcast during the call since the same sender is re-used for all local video tracks. Therefore, disable the low resolution simulcast streams for low fps screensharing so that the bridge forwards only the highest resolution stream which is important for low fps screensharing.

In unified plan, re-use of m-line (i.e., adding an SSRC, removing it and then adding it back) causes the browser to not render the media on Chrome and Safari. The WebRTC spec is not clear as to how browsers have to behave, this doesn't cause any issues on Firefox. As a workaround, only change the media direction and leave the ssrc in the remote desc. This automatically triggers a 'removetrack' event on the associated MediaStream and the track can be removed from the UI.