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

Provide a method for changing the capture fps for desktop tracks during the call. These changes to the lib are needed for making it configurable from the UI.

Set the initial video type state to NONE to cover cases when the endpoint joins video muted.

This can be controlled through the config.js  option 'enableUnifiedOnChrome'.

When a msid attribute is missing in the 'a=ssrc' line, use the local endpoint id as an indentifier. Move this generation logic to LocalSdpMunger. Also suppress notifying Jicfo of a ssrc change on Firefox when the change is a result of the media being suspended on the jvb connection.

Do not send old format constraints if no constraints are set before the channel is initialized.

Use the new Colibri message "EndpointStats" for broadcasting the local stats. The bridge then will be able to filter the endpoint stats and send them only to the interested parties instead of broadcasting it to all the endpoints in the call.

Stop using the legacy pc constraints that are no longer wired up to WebRTC.

* 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.

* Added video mute participant

* Trigger mute event

* Optimized mute type checks

* Fixed event name

* Fixed some linter issues

* Fixed more linter issues

* And even more linter issues fixed

* And more linter fixes

* Added media type to analytics event

Translate the 'LastNChangedEvent', 'SelectedEndpointsChangedEvent' and 'ReceiverVideoConstraint' messages into the new 'ReceiverVideoConstraints' message that invokes the new bandwidth allocation algorithm in the bridge that is described here - https://github.com/jitsi/jitsi-videobridge/blob/master/doc/allocation.md. useNewBandwidthAllocationStrategy=true in config.js will invoke the translation in the client.

Fire PERMISSION_PROMPT_IS_SHOWN when none of the devices have a label
Fire a new SLOW_GET_USER_MEDIA event if the timeout for getUserMedia is exceeded

Update JitsiMeetJS.createLocalTracks to include the options for firing the above events
in the provided options argument. Deprecate the firePermissionPromptIsShownEvent flag in
method's signature

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.

"Pinned" endpoints are deprecated, use "selected" instead.

Remove redundant _channelOpen flag from RTC and rely on the channel's isOpen() getter instead.
This fixes an error throw when the _channelOpen flag was not updated while the channel's been
closed by the bridge.

Delays e2e encryption initialization until any encryption key is set.
Recreates peerconnections in order to enable insertable streams
only if the e2ee is used. This is to avoid bug around insertable streams
which may cause audio issue when the main JavaScript thread is loaded:

https://bugs.chromium.org/p/chromium/issues/detail?id=1103280

This is useful for testing insertable streams related issues.

Co-authored-by: Saúl Ibarra Corretgé <saghul@jitsi.org>

* feat(screenshare): support remote wireless screenshare

- ProxyConnectionService is exposed and meant to be the
  facade for creating and updating a peer connection with
  another peer, outside of the muc.
- ProxyConnectionPC wraps JingleSessionPC so the peer
  connection handling can be reused.

* attempt to make more configurable

It's supported in Chrome >= 65.

Use the W3C Permissions API where possible. At the time of this writing this is
supported in Chrome >= 64.

* feat(conference): expose method to select multiple participants

* squash: fix typo

* squash: reduce to one code flow

* squash: default to empty array to mimic previous null default

* squash: fix typos, change docs, default params

* squash: remove defaults

* squash: update comment about array arg

* ref(video-quality): cache max frame height and send on channel open

Currently it is possible to try to change the max receive video
frame height before the data channel is open. In that case an error
will be thrown. This change makes it so that the desired frame height
is saved and sent on channel open, avoiding the thrown error the
max receive video frame height logic is exercised through the
JitsiConference api.

* squash: do the second part of the actual fix

1. The example was using the Promise return value of JitsiMeetJS.init
which is no longer possible/correct after commit "core: refactor
initialization not to return a Promise".

2. We went back and forth with the value returned by JitsiMeetJS.init:
we initially didn't return a value, then we started returning a Promise,
and now we're not returning a value. Whether we'll go back to returning
a value is up in the air. Anyway, the return value is practically
determined by the last in a chain of function calls: JitsiMeetJS, RTC,
RTCUtil. Since the chain is not really documented, it will not hurt much
to make it easier to refactor the chain by "composing" the functions.

There is nothing asynchronous about the initialization process (anymore), thus
turn it into a synchronous method.

In addition, WebRTC support is absolute, it cannot change from not being
supported to being supported (as it plreviously could, thanks to Temasys) so get
rid of the ancillary logic to support that.

Last, introduce a way to check if WebRTC is supported in the current
environment: JitsiMeetJS.isWebRtcSupported().

This commit will append "-" + tpc.id to every local 'MSID', 'cname',
'label' and 'mslabel', before feeding the local SDP to the Jingle layer.
It will make stream IDs unique across TraceablePeerConnection instances
and prevent from conflicts in some corner cases.

For example this will fix a problem where if the client drops
the conference without leaving the XMPP MUC gracefully and will join
the conference again without recreating the local tracks it would lead
to the MSID conflict, because the stream is still advertised by
"the ghost" participant.

In order to include browser capabilities in RTCBrowserType I needed to
refactor RTCBrowserType.

When a mac's default audio output is changed, audio elements
are not being updated with the new default. A default audio
output change in the OS fires a DEVICE_LIST_CHANGED event,
so use that to always update the sinkId.

Also, use wrapAttachMediaStream for newGumFlow to be ensure
the proper sinkId is set on audio elements being coupled
a remote track.

* revert: Reverts be665cbff7.

* ref: Renames "peerjid".

* ref: Refactors the initialization of a peer connection.

* feat: Re-implements the A/B test for the "suspend video" feature.

* squash: Deep copy.

* ref: Renames forceSuspendVideo to abtestSuspendVideo.