This refactors all handling of audio-only and last N to 2 features in preparation
for "low bandwidth mode".

The main motivation to do this is that lastN is a "global" setting so it helps
to have all processing for it in a single place.

The conference disconnection process is asynchronous which means there's
no guarantee that there will be CONFERENCE_LEFT event for the old
conference, before the next conference is joined. Because of that we can
end up with two simultaneous calls on the native side which is not
always supported. End the call on CONFERENCE_WILL_LEAVE to fix this
corner case.

* feat(Android): implement ConnectionService

Adds basic integration with Android's ConnectionService by implementing
the outgoing call scenario.

* ref(callkit): rename _SET_CALLKIT_SUBSCRIPTIONS

* ref(callkit): move feature to call-integration directory

* feat(ConnectionService): synchronize video state

* ref(AudioMode): use ConnectionService on API >= 26

Not ready yet - few details left mentioned in the FIXMEs

* feat(ConnectionService): add debug logs

Adds logs to trace the calls.

* fix(ConnectionService): leaking ConnectionImpl instances

Turns out there is no callback fired back from the JavaScript side after
the disconnect or abort event is sent from the native. The connection
must be marked as disconnected and removed immediately.

* feat(ConnectionService): handle onCreateOutgoingConnectionFailed

* ref(ConnectionService): merge classes and move to the sdk package

* feat(CallIntegration): show Alert if outgoing call fails

* fix(ConnectionService): alternatively get call UUID from the account

Some Android flavours (or versions ?) do copy over extras to
the onCreateOutgoingConnectionFailed callback. But the call UUID is also
set as the PhoneAccount's label, so eventually it should be available
there.

* ref(ConnectionService): use call UUID as PhoneAccount ID.

The extra is not reliable on some custom Android flavours. It also makes
sense to use unique id for the account instead of the URL given that
it's created on the per call basis.

* fix(ConnectionService): abort the call when hold is requested

Turns out Android P can sometimes request HOLD even though there's no
HOLD capability added to the connection (what!?), so just abort the call
in that case.

* fix(ConnectionService): unregister account on call failure

Unregister the PhoneAccount onCreateOutgoingConnectionFailed. That's
before the ConnectionImpl instance is created which is normally
responsible for doing that.

* fix(AudioModeModule): make package private and run on the audio thread

* address other review comments

BaseApp does all the heavy-lifting related to creating the redux store,
navigation, and so on.

App currently handles URL props and actually triggering navigation based on
them.

Fix the "mute ping pong" for once and for all. This patch takes a new approach
to the problem: it keeps track of the user generated CallKit transaction ations
and avoids calling the delegate method in those cases.

This results in a much cleaner and easier to understand handling of the flow: if
the delegate method is called it means the user tapped on the mute button. When
we sync the muted state in JS with CallKit the delegate method won't be called
at all, thus avoiding the ping-pong altogether.

In addition, make sure all CallKit methods run in the UI thread. CallKit will
call our delegate methods in the UI thread too, thsu there is no need to
synchronize access to the listener / pending action sets.

Not all TRACK_ actions include the `local` attribute, so use the underlying
`jitsiTrack` to check it.

Your truly introduced this regression in
8c7a3f16b1, alas.

The audio only mode is used to set the CallKit call type. This affects the
behavior on the recent calls entries (calls are marked either as audio or video
calls).

Sync both at the start and for transitions. The previous code was working by
chance (in a way): when the CallKit UI is presented the local video is muted,
which triggers a SET_VIDEO_MUTED action, at which point the audio-only mode was
checked for. Now we are more explicit and act on SET_AUDIO_MUTED.

Rely solely on actual track state, rather than the desired state, (what
base/media represents).

Read the muted state from the track itself instead of from base/media. This
avoid expressing the incorrect desire when the call starts muted because
permission was never granted.

Audio muting does not have an authority.

Add ability to provide a display name in the configOverwrite object that
when available it will be used to customize the name of the meeting in
callkit screen and recent call list.

Co-authored-by: Daniel Ornelas <daniel.ob64@gmail.com>
Co-authored-by: Lyubo Marinov <lmarinov@atlassian.com>

With this the RN component and the consumer app can share same CallKit
provider, configuration, and enable to be part of multiple listeners of
the CallKit flow events. The main driver of this is to enable the
consumer app to be able to report an incoming call to the OS before
loading the JitsiMeetView. Once the user answers the call, the app can
instantiate a JitsiMeetView, pass the CallKit call UUIID, and the Jitsi
Meet components will handle the connection and report back to CallKit
that the  call has been established.

Turns out this was a bit more involved than I originally thought due to an
interesting (corner) case: IFF the user was never asked about microphone
permissions and the call starts with audio muted, unmuting from the CallKit
interface won't work (iOS won't show the prompt, it fails immediately) and we
need to sync the mute state back.

* ref: Restructures the pinned/unpinned events.

* ref: Refactors the "audio only disabled" event.

* ref: Refactors the "stream switch delay" event.

* ref: Refactors the "select participant failed" event.

* ref: Refactors the "initially muted" events.

* ref: Refactors the screen sharing started/stopped events.

* ref: Restructures the "device list changed" events.

* ref: Restructures the "shared video" events.

* ref: Restructures the "start muted" events.

* ref: Restructures the "start audio only" event.

* ref: Restructures the "sync track state" event.

* ref: Restructures the "callkit" events.

* ref: Restructures the "replace track".

* ref: Restructures keyboard shortcuts events.

* ref: Restructures most of the toolbar events.

* ref: Refactors the API events.

* ref: Restructures the video quality, profile button and invite dialog events.

* ref: Refactors the "device changed" events.

* ref: Refactors the page reload event.

* ref: Removes an unused function.

* ref: Removes a method which is needlessly exposed under a different name.

* ref: Refactors the events from the remote video menu.

* ref: Refactors the events from the profile pane.

* ref: Restructures the recording-related events.

Removes events fired when recording with something other than jibri
(which isn't currently supported anyway).

* ref: Cleans up AnalyticsEvents.js.

* ref: Removes an unused function and adds documentation.

* feat: Adds events for all API calls.

* fix: Addresses feedback.

* fix: Brings back mistakenly removed code.

* fix: Simplifies code and fixes a bug in toggleFilmstrip

when the 'visible' parameter is defined.

* feat: Removes the resolution change application log.

* ref: Uses consistent naming for events' attributes.

Uses "_" as a separator instead of camel case or ".".

* ref: Don't add the user agent and conference name

as permanent properties. The library does this on its own now.

* ref: Adapts the GA handler to changes in lib-jitsi-meet.

* ref: Removes unused fields from the analytics handler initializaiton.

* ref: Renames the google analytics file and add docs.

* fix: Fixes the push-to-talk events and logs.

* npm: Updates lib-jitsi-meet to 515374c8d3.

* fix: Fixes a recently introduced bug in the google analytics handler.

* ref: Uses "value" instead of "delay" since this is friendlier to GA.

* ref(analytics): centralize all event names

* squash: fix typo and alpha ordering

* squash: rename file to AnalyticsEvents to parallel lib-jitsi-meet

Fallback to the room name otherwise. This allows us to have better entries in
the phone history.

* Javadoc introduced @code as a replacement of <code> and <tt> which is
  better aligned with other javadoc tags such as @link. Use it in the
  Java source code. If we switch to Kotlin, then we'll definitely use
  Markdown.

* There are more uses of @code in the JavaScript source code than <tt>
  so use @code for the sake of consistency. Eventually, I'd rather we
  switch to Markdown because it's easier on my eyes.

* Xcode is plain confused by @code and @link. The Internet says that
  Xcode supports the backquote character to denote the beginning and end
  of a string of characters which should be formatted for display as
  code but it doesn't work for me. <tt> is not rendered at all. So use
  the backquote which is rendered itself. Hopefully, if we switch to
  Markdown, then it'll be common between JavaScript and Objective-C
  source code.

This commit adds initial support for CallKit on supported platforms: iOS >= 10.

Since the call flow in Jitsi Meet is basically making outgoing calls, only
outgoing call support is currently handled via CallKit.

Features:
 - "Green bar" when in a call.
 - Native CallKit view when tapping on the call label on the lock screen.
 - Support for audio muting from the native CallKit view.
 - Support for recent calls (audio-only calls logged as Audio calls, others show
   as Video calls).
 - Call display name is room name.
 - Graceful downgrade on systems without CallKit support.

Limitations:
 - Native CallKit view cannot be shown for audio-only calls (this is a CallKit
   limitaion).
 - The video button in the CallKit view will start a new video call to the same
   room, and terminate the previous one.
 - No support for call hold.