...which contains conference duration in seconds and
performance stats (currently long tasks).

Add a performance stat around long tasks. Chrome supports PerformanceObserver API that lets us
register for long tasks event. Any task that takes longer than 50ms is considered a long task.

The audio levels reported on the audio receivers are lower when compared to the value reported by getStats.
Values reported by getStats on chrome do not follow the the spec and since we have combination of clients using both getStats and getSynchronizationSources,
lets stick to one scale to make them look uniform.
Also, the receivers seem to be reporting audio level for a little bit after the remote user has muted. Make sure the track is unmuted
before setting the audio level on the track.

* feat: use getSynchronizationSources on the receiver for remote audio levels
Use getSynchronizationSources if it is supported, fallback to using getStats otherwise.
* feat/ref: Use the local audio levels from LocalStatsCollector
When using getSynchronizationSources, use the audio levels from LocalStatsCollector for NoAudioSignalDetection.js
Remove obsolete code - TalkMutedDetection feature using audio levels is not used anymore

This has become obsolete thanks to ES6 arrow functions

This adds the 'codec' object to the RTP stats passed out of lib-jitsi-meet.

Enabled desktop sharing and discontinue support for Safari versions prior to 12.1, i.e., the ones with no VP8 support

To be sure we always report room name in small case and as mobile and jigasi report this way it will take time for them to adopt which leads to wrong stats.

* Adds confID to JitsiConference option and enables callstats siteID.

* Simplifies siteID logic and use '/' as default when there is no tenant.

In TraceablePeerConnection: we're no longer injecting a recvonly SSRC
when the local video track is muted, so it's normal that there is no
SSRC found in the local SDP when it's muted.

About RTPStatsCollector: at the time of adding this log statement a case
was missed when local audio track could be replaced in the P2P
connection when a new audio device is selected.

It should be provided by the environment, just like on web. This avoids bundling
callstats on web, where it won't be used (we dynamically download it).

This reduces the bundle size from about 1MB to 675KB, almost 40%.

Fixes tests.

Updates speakers stats with values received by the spealerstats component, all the logic is activate only in case we discover speakerstats component address from disco-info and the incoming message is coming from that component.

When the main domain advertises component with type 'speakerstats' every participant will send message to the component with the name of the room where this happen.

The 'self.errorCallback' was passed incorrectly - it should have been
bound to 'self' eventually, but now the arrow function is the preferred
way.

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().