Until M69 Chrome used to consistently emit "onstreamadded" when
audio/video stream is added for the first time and then emit track
events only if the stream is modified afterwards (video track replaced).

However it looks like now it can first emit "onstreamadded" and then
additional "ontrackadded" for the MediaStreamTrack that was already
included in the MediaStream signalled by the "onstremadded" event. I
have not managed to figure out what is causing this and the only
difference in the SDP is the fact that the remote peer (JVB) includes
IPv6 candidates. Anyway it doesn't hurt to have such a safeguard.

Just skip it in such environments, such as React Native.

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

The code for handling device availability has been disabled for a long time,
plus it's ill named since it represents 2 abstractions: lack of permissions and
lack of devices.

Time for it to rest in the git graveyard.

We now use PeerConnection promise-based APIs, which were introduced in M51.

This function helps simplify the code when handling browsers that are pretty
much Chrome in disguise.

Chrome fires devicechange twice. This causes the DEVICE_LIST_CHANGED
event to fire twice. The listener in jitsi-meet has async logic
to handle the event, including creation of new tracks. So it
can happen that two devicechange events fire, the jitsi-meet
listener fires twice, creates duplicate tracks, and uses both
tracks. Diffing the device list for an actual change is a workaround
against extra devicechange events.

Note: this does not solve the issue in jitsi-meet of it not
handling quick successive DEVICE_LIST_CHANGED events.

Keeps it consistent with the previous implementation.

Just use navigator.mediaDevices.enumerateDevices().

It's 2018 after all...

Get rid of all wrappers and use navigator.mediaDevices.getUserMedia, since all
supported platforms have it by now.

Also use the unprefixed versions of WebRTC APIs.

- Don't use it on supported browsers
- Make sure no stray timers are left

* Enables adapter for edge.

We were not filtering correctly all unsupported iceServers and an error is thrown and no connection is established.

* Enable desktop sharing for Edge.

Currently when replacing video track with desktop sharing one, doesn't work because of: https://developer.microsoft.com/en-us/microsoft-edge/platform/issues/17528460/
If Edge user joins first and enables desktop sharing it will work when others join. Tried also to use replaceTrack/setTrack as a workaround, but again we hit an error, this time InvalidAccessError.

* Adds the helper function usesAdapter in BrowserCaps.

* 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

* Change layer suspension to use parameters in RTPSender

We no longer suspend unused simulcast layers via a bandwidth cap in the
SDP, instead we'll use the new parameters in RTPSender to enable and
disable streams explicitly.  The main advantage here is the RTPSender
method ramps up immediately when we re-enable the layers (as opposed to
the SDP bandwidth cap which took 30+ seconds).

* Fix linter issues

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().

RTCUtils.attachMediaStream was changed not to return elements;
instead it returns undefined by default. When mapping over
containers and call RTCUtils.attachMediaStream, containers
would be changed to undefined.

* change connection quality to take layer suspension into account

* Support layer suspension

Add support for a message which notifies the endpoint whether or not it
is selected (meaning its HD stream is in use).  If it is not
selected and enableLayerSuspension is set to true then it will impose a
bandwidth limit in the SDP to suspend sending the higher layers.

* only add the IS_SELECTED_CHANGED listener in JingleSessionPC if layer
suspension is enabled

this prevents doing a local o/a when we don't need it

When cancelling the prompt to install the extension, instead of
treating it as a generic error treat is as the cancel extension
error. This allows existing logic to trigger that will not show
an error message on cancel.

* Switches camera id to mandatory when using old gum flow.

When it fails we retry with different resolutions, and if that doesn't work we remove device id and let gum to decide which device to use.

* Fixes comments.

* fix(edge): get p2p working up to ice failure

Edge does not support trickle ice. In an attempt to work around
such, and get p2p working, some other work was needed to even
get to the point of ice failure. This is that some other work.

* squash: fix typos, no intermediary map

* squash: remove extra line break

Adds stats and audio levels for Safari. Enables the new getStats API for Firefox, that will get rid of the following warning:
'non-maplike pc.getStats access is deprecated, and will be removed in the near future! See http://w3c.github.io/webrtc-pc/#getstats-example for usage.'

This PR exposes the functionality that's already built-in the
TraceablePeerConnection that has it gather stats from the wrapped
PeerConnection.

* fix(SS): Set min and max frame rate to 5

* fix(SS): Set min and max frame rate for the new GUM flow.

* feat(ss_framerate): Add config option for min/max frame rate.

* doc(RTCUtils): Fix params format.

* fix(SS_framerate): Don't pass undefined constraints.

* fix(SS_constraints): Handle  chromeMediaSourceId === undefined.

* ref(gum): try to reduce complexity of obtainAudioAndVideoPermissions

Even though I walk through the valley of the shadow of death
I will fear no evil... Reduce indenting and repeated calls to
getting desktop streams by creating a promise chain.

* squash: use arrow func

* squash: move constructor support to helper

* squash: put ff into mediastream constructor check

* squash: ff support media stream constructor

MDN states its been supported since 44. ESR is currently
52.

* squash: rename dsoptions, constant defaults

* squash: move comment about missing tracks error handling

* squash: wrap getUserMediaWithConstraints in promise

* squash: split up av funcs

* squash: hey, some tests...are better than no tests?

* log: Logs when analytics are disposed, drops events early.

* fix: Adds a missing parameter to a function call.

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.

In Rect Native it will stall the app as it's huge.