Calculate the target bps based on the video quality settings and the codec configured on the peerconnection.
Hardcode target video bitrates for RN since it doesn't support setting max bitrates.

Make sure the codec order is munged on all the m-lines for unified plan clients. Implement the logic for setting the preferences through RTCRtpTransceiver#setCodecPreferences.

Capture the audio levels only for the top 5 speakers as RTCRtpReceiver#getSynchronizationSources can be expensive when we have too many audio receivers in the call.
Also, capture the audio levels for track that are unmuted if RTCRtpReceiver#getSynchronizationSources is not supported.
Switch Safari to using getStats since its reporting errorneous values, i.e., 0.000001 as audio level for all remote audio tracks.

* fix(TPC): Configure degradation preference in RTCRtpSendParameters.
Properly configure degradation preference on RTCRtpSendParameters instead of RTCRtpEncodingParameters. Fixes https://github.com/jitsi/lib-jitsi-meet/issues/1510.

* feat(JingleSessionPC): Remove ssrcs from remote desc when a user leaves.
Remove the ssrcs (associated with remote sources) from the remote desc along with the removal of the remote tracks when an endpoint leaves the call. The source-remove signaling message from Jicofo will no longer be needed in this case and can be dropped.

Apple added getUserMedia support for WkWebview based browsers like chrome and Firefox on iOS 14.3. These browsers behave as Safari does on iOS. Therefore, extend the Safari checks to these webkit based browsers as well.

When a second remote track of the same mediatype is received for an endpoint, remove the existing track before creating the new remote track.

All the send video constraints for the client, i.e., what simulcast streams will be enabled based on constraints received on the bridge channel, will be handled by the SendVideoController class.
The receive video constraints like lastN, selectEndpoints and receive video resolution will be handled by the ReceiveVideoController class.

Make sure screenshare w/ presenter gets the full 2500 Kbps bitrate.

The max bitrate for VP9 is enforced by adding the b=AS:<limit> line in the SDP since there is no way to configure the max. bitrates for the individual SVC streams using RTCRtpSender.setParameters.

Fixes https://github.com/jitsi/lib-jitsi-meet/issues/1466

Determine the preferred codec for a given endpoint based on the config.js settings and the codecs supported by the endpoint.
The preferred codec is published in presence and then used by the other endpoints in the call during join/leave to determine
if the codec needs to be changed on the fly. Different codecs can be configuered for p2p/jvb connections.
The preferredCodec/disabledCodec settings under videoQuality will have precedence over the older settins like preferH264/disableH264.

in order to not match a=msid-somethingelse, a=ridiculous and similar

On every call to RTCRtpSender.setParameters(), ensure that the resolution configured for the encoding matches that of the value configured on the RTCRtpSender when the source was added to the peerconnection. This should prevent us from overriding the default values if the browser returns erroneous values when RTCRtpSender.getParameters is used for getting the encodings info. This fixes the issue on recent versions of Safari where the 'scaleResolutionDownBy' value comes back as 1 for all encodings even though the encoding resolution is different from the stream capture resolution.

In p2p mode, 'scaleResolutionDownBy' is used for downscaling a stream when needed, i.e. when the user receives a receive constraint of 360p because the other participant is in tile view. When desktop share is started, the encoding config has to be scaled back up so that the other participant starts receiving HD resolution for the share as desktop shares are autopinned. Therefore, encodings have to enabled/disabled for desktop shares as well. Earlier it was done only for camera tracks.

Fixes https://github.com/jitsi/lib-jitsi-meet/issues/1333 in applications that use lib-jitsi-meet and do not have layer suspension enabled.

Add the conference flag back since the bridge is able to handle the case where more than 1 temporal layers are received even when only 1 ssrc is signaled.

seen in #1364 but it is older

In Chrome, if 'maxBitratesVideo' options are specified
and 'capScreenshareBitrate' is disabled, then simulcast is enabled
for screenshare. Chrome in plan B, does not send screen stream if
'maxBitrate' are set and simulcast is enabled.

On Safari, MediaStream.onremovetrack is not fired when a remote desciption with a removed MSID is applied. As a result, new remote tracks for the same endpoint are not created causing issues where the video is not updated on Safari. Also, make sure local tracks on pc are updated when devices are changed while the user is in muted state.

* add p2p meta info to pc

* fix typos

This should fix the case when camera is started with 1080p and LD simulcast stream's resolution is 270p but the requested resolution is 180p.

We're pretty sure it's no longer necessary with JVB2; and it costs
half an RTT in setup, and stops us being able to choose which
DTLS/SRTP protection profile to use.

The constraints are applied again after renegotiation is done. This should mitigate a fail case on Safari when the RTCRtpSendParameters are not available until the simulcast streams are added via SDP munging before sLD is called.

 Only add/remove the tracks from the peerconnection so that the camera led turns off on mute.

Configure the encodings on FF after the track is added.

Calculate the bitrate based on the sender video constraint applied on the track and the bitrates specified for different resolutions.

Make 'preferH264' and 'disableH264' as deprecated.

Use 'preferredCodec' and 'disabledCodec' under videoQuality config.js settings for setting codec preferences.
Do not prefer VP9 on Firefox because of https://bugzilla.mozilla.org/show_bug.cgi?id=1633876.

Make sdp munging logic for changing codec preferences more generic
Do not negotiate High profile H264 codecs on when RN clients are in the call

...in setMediaTransferActive. The sender video constraints
are controlling the 'encoding.active' flag and this code
would interfere. It's right now not used anyway as P2P is
disabled in the unified mode. If this modification is
necessary we'll add it back when working on unified P2P
in such a way that will play nicely with the sender constraints code.

...and remove duplication.

...when starting without video.

The app would crash in setSenderVideoConstraint,
because the RTP encoding are not available on Safari
until sLD/sRD cycle is performed.

Handle the promise returned by RTCRtpSender.setParameters.

(cherry picked from commit a14b3c80ae)

When the video sender constraint changes in the p2p case when the remote user switches between stage/tile view,
get a downsampled video of the original 720p stream instead of requesting the camera for a new stream with the lower resolution.
Fixes https://github.com/jitsi/jitsi-meet/issues/7267

* feat: Apply max bitrates on video sender
Add the ability to control the max bitrates on the video sender through a config.js videoQuality setting.

* ref: Use '_' prefix only for internal methods