Get rid of the browser specific keys and use the standard spec-compliant fields for stats.
Get the resolution/fps for remote streams from 'inbound-rtp' stats. Use the 'track' stats for the local resolution/fps since these take the active simulcast streams into account.

Capture the audio levels only for the top 5 speakers as RTCRtpReceiver#getSynchronizationSources can be expensive when we have too many audio receivers in the call.
Also, capture the audio levels for track that are unmuted if RTCRtpReceiver#getSynchronizationSources is not supported.
Switch Safari to using getStats since its reporting errorneous values, i.e., 0.000001 as audio level for all remote audio tracks.

Apple added getUserMedia support for WkWebview based browsers like chrome and Firefox on iOS 14.3. These browsers behave as Safari does on iOS. Therefore, extend the Safari checks to these webkit based browsers as well.

...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%.