This is performed by stripping all traces of H.264 from the local SDP before
passing it to `setLocalDescription`.

The default was still 360p if the option was missing from config.js. In
addition, move the logic outside of a browser check, so all browsers get
glorious HD by default.

ESLint 4.8.0 discovers a lot of error related to formatting. While I
tried to fix as many of them as possible, a portion of them actually go
against our coding style. In such a case, I've disabled the indent rule
which effectively leaves it as it was before ESLint 4.8.0.

detectChrome() detects the browser but then may throw an exception while
parsing its version. The exception could be reproduced by toggling the
device toolbar and instructing Chrome to identify as iPhone 6 Plus.

adapter.screenshare.js uses the variable requestUserMedia without
defining it. And that's a problem because it's packed in strict mode so
it will cause an error to be thrown at runtime which will fail
lib-jitsi-meet and, consequently, the whole app. As the variable
requestUserMedia appears to be global in certain browser, prevent the
use of the undefined variable without really changing the outcome of its
use.

@paweldomas said the typo should likely remain because it matches a typo
in Chrome.

For p2p, we have to take preferh264 into account when setting the local description, since
that prefer is what will be received by the far side.  if we only do it on the remote description,
like before this change, we'll never actually end up preferring h264.

The 'setMutedInProgress' flag is unreliable and it can lead to weird
situations when mute operation is in progress, but the removal from
JingleSessionPC is waiting on the queue. In such case LocalSdpMunger
will fake SDP even if the track is in the PeerConnection which may
confuse some parts of the SDP transformation chain.

For example RtxModifier will not attempt to inject Rtx group description
if it sees that the SSRC are already there. But it can happen that even
though the SSRCs are there they do not contain valid track ID, because
the track has changed.

The LocalSdpMunger is supposed to fake the SDP only when the video
MediaStream is not in the PeerConnection. So this commit implements just
that: a way to tell if JitsiLocalTrack's stream is currently in
the PeerConnection (added through TraceablePeerConnection).

Random coding style-related issues that bothered me while reading the
source code touched by "Fix local video disappearing from local
thumbnail" (https://github.com/jitsi/lib-jitsi-meet/pull/571). These
include naming, formatting, comments, typos, duplication.

Commit "fix(JitsiLocalTrack): restore 'inMuteOrUnmuteProgress'" removed
all read occurrences of dontFireRemoveEvent so the one and only write
left is now obsolete.

Flag 'dontFireRemoveEvent' was unreliable, because if
the JitsiLocalTrack is muted/unmuted rapidly it's state can be false
at the time when ended event is triggered. The event handlers must be
unregistered to handle this gracefully.

Restores 'inMuteOrUnmuteProgress' flag used by the LocalSdpMunger.

(cherry picked from commit 2e6c4dd)

* feat(quality-slider): expose video constraint method and data channel opening

* squash: rename to data channel opened

'echoCancellation' should be disabled by 'disableAEC'
'disableAP' should disable all options

Instead of failing with an error if another operation is in progress, which is
impossible to know as a user, chain them using the primises API.

As new use case for joining a conference without calling GUM has been
added to jitsi-meet enumerate devices should not be delayed. In fact not
only it was delayed, but never called in such case
(getUserMediaStatus.initialized stays false forever).

Fixes crash on undefined var in case get user media was not called,
before the conference started.

* make audio processing constraints configurable

AP: echoCancellation is a "master switch" for disabling all audio processing.  set to disableAP=true and stereo=true allows fullband, stereo sound.
AGC: misbehaves with certain 'pre-gained' mics.
HPF: can't hear any effect but it's the last constraint to make configurable.

* Update RTCUtils.js

* add support for signaling simulcast for unified plan

1) munge the incoming offer to advertise support for receiving simulcast in unified plan
2) add 3 simulcast encodings via rtp sender
3) inject an ssrc-group line, even when using rids for simulcast (jicofo will rely on this)

* add rid information to the session-accept jingle

* address lenny's feedback

* make firefox simulcast configurable

* fix missing comma

Will trigger external installation also for sites that are not the main
whitelisted domain. This allows to reuse one extension for many sites
(they still have to be on the whitelist, but only one can be "main").

Will not crash, but delay requests if arrive, before the initialization
process is done (there is no way to tell that really).

CHROME_EXTENSION_USER_GESTURE_REQUIRED will be triggered if
the extension installation fails, because the request to get desktop
stream did not originate in user gesture handler (a button click for
example).

* feat: add 'disableDesktopSharing' option

* ref(ScreenObtainer): simplify init

They are really verbose, and add no real value. In addition, log the channel
type once opened.

Some browsers may not support data channels (hi there, Edge!) and sometimes it
may not be desirable for other reasons.

* ref(JSPC): simplify SSRC owner in P2P

There's no need for any extra extensions for SSRCs owner signalling in
P2P, because it's always the remote peer who owns them.

This also fixes a problem where no SSRC owner was added for 'source-add'
in P2P (for JVB conference Jicofo adds that).

* fix(TPC): always advertise 'sendrecv'

Our media direction is only ever updated on the remote side with
the initial offer (or answer). Because of that we want to advertise
'sendrecv' even if we start with no video (or audio) track.

It is OK to adjust this direction in the localDescription getter,
because it's adjusted again in the setter to the correct value based on
local tracks, so the SDP transformation chain still works fine.

When the lastN is set to 0 the P2P session will notify the remote peer
to stop sending video.