* fix(events): unify events to a single EventManager type, add support for single typescript declaration

* fix(lint): fix lint

* fix(events): fix incorrect instatiation

* fix(events): clean up redundant methods

* fix(events): keep EventEmitter name, alias NodeEventEmitter

* fix(events): fix loose reference

* fix(EventEmitter): remove on/off alias as redundant

* fix(RTCUtils): bring event handlers under class to use same event emitter

* fix(RTCUtils): fix lint

* ref: Remove PrecallTest.

* ref: Remove callstats support.

* Added rtcstats

* increased webpack performance size warning to 830K

* Addressed most of Saghuls and Andrei's review comments.

* moved rtcstatsUseLibJitsi from analytics to testing

* refactor ljm rtcstats, use ljm conference events

* remove debug logs

* remove wrongly named files

* rename rtcstats module

* fix test

* address code review

* fix pack error

* address code review

---------

Co-authored-by: Nils Ohlmeier <github@ohlmeier.org>

* fix(stats) Obtain resolution/fps from 'outbound-rtp' stats.
'Track' based stats were dropped in Chrome 112, therefore send resolution and fps for the simulcast case needs to be calculated based on the 'outbound-rtp' streams that are currently active.

* squash: remove an unwanted log

* Squash: Address review comments.
Use outbound-rtp stats for both Firefox and Safari.
Firefox - Ignore active field if not present in the stats and calc fps using 'framesSent'.

Track based stats have been removed recently and RTCRtpReceiver:getSynchronizationSources has been available in the browsers for a while now.

* feat(face-landmarks): add timeline

* add duration to xmpp payload

Use the participant ID as the key to codecs, etc, instead of the source
name. The object is further indexed by SSRC.

Don't wait for both codecs to be set before propagating codec
information.

Sample data model for codecs:

codec: {
    participant1: {
        ssrc1: {
            audio: 'opus',
            video: undefined
        },
        ssrc2: {
            audio: undefined,
            video: 'VP8'
        },
        ...
    }
    ,...
}

Some values, like codecs, are available right away, we don't need to
compute differences.

This fixes an error that gets logged when no media is being sent/received by the endpoint.
Logger.js:154 2022-09-12T14:40:53.476Z [modules/statistics/AvgRTPStatsReporter.js] <xl.addNext>:  bandwidth_upload - invalid value for idx: 0

* fix(multi-stream): Fix local SSRC cache to include multiple video streams.
If multiple local video streams are found in the SDP, cache all of them instead of the first video SSRC. This fixes an issue where the resolution/fps stats for the screenshare track are not available.

* squash: new track inherits the source name of old track if it exists.

This allows us to display resolution and framerate stats on separate tiles when multi-stream mode is enabled.

* feat: facial expression in speaker stats

* feat: send xmpp message with facial expression to server

* fix: rebase conflicts

* feat: facial expression in speaker stats

* feat: facial expression in speaker stats

* fchore(facial-expressions): remove the send facial expression call from update facial expression function

* feat(facial-expressions): store expresions as a timeline

* refactor(facial-expressions): store expressions by counting them in a map

* feat(facial-expressions): camera time tracker

* add(facial-expression): increase facial expression with duration parameter from payload

* add(facial-expressions): the disgusted expression

* refactor(facial-expressions): remove camera time tracker

* refactor(facial-expressions): change data channel message handel position for facial expressions and renamed some types

* fix(facial-expressions): move facial expression endpoint message handler from statistics.js to ConnectionQuality.js

* fix(facial-expressions): remove unused type

* enable config callstats configParams

* add comments

* wrap long comments

* fix format

* fix format

* fix format

* add jsDoc

Switch to returning a Promise for getStats. Reset frame rate stat to 0 when video is suspended as a result of endpoint falling out of last-n.

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.