Regression was introduced in 07f037e12e/. Also process encodeTime stats when H.264 is selected

It has been broken for over 3 years now, since ca325f5ef9 (diff-9e19da30f465ca5665ac3d7ca1aa03d0498aed1be0cb2d7eeb27684a2636da77)

Ever since that change, the "audioLevelReportHistory" property is not
populated, so it justs acts on nothing an generates bogus log lines such
as:

```
[modules/statistics/AudioOutputProblemDetector.js] A potential problem is detected with the audio output for participant b5fb30bc, local audio levels: [null,null], remote audio levels: undefined
```

Since nobody seems to have noticed in 3 years it's safe to assume we
don't need this at all, so it gets the axe treatment.

* feat(quality) Add a QualityController class for runtime adjustments.
Make run time adjustments to the client when adaptive mode is enabled.

* feat: Update lastN and receive resolution to improve quality.

* squash: Address review comments

* squash: Add more logging and address review comments.

* feat(codec-selection): Use the new codec selection API
https://github.com/w3ctag/design-reviews/issues/836. This allows the client to seamlessly switch between the codecs without having to trigger a renegotiation.
This feature is behind the flag testing.enableCodecSelectionAPI in config.js

* fix(stats): Fix local resolution stats.
The video codec for the local video sources needs to identified differently now, from the codecs field in the RTCRtpSendParameters returned by the browser. We no longer munge the remote SDP to switch to a different codec.

* feat(stats): Feed encodeTime stats for all local SSRCs to the codec selection mechanism.

* fix(codec-selection) Continue to mumge SDP for selecting H.264.

* feat(codec-selection) Make screenshare codec configurable.
If no 'screenshareCodec' is set under videoQuality or p2p settings, AV1 will be selected as default.

* squash: Address review comments

* Update modules/RTC/CodecSelection.js

Co-authored-by: Saúl Ibarra Corretgé <s@saghul.net>

* fix(codec-selection) Add codec to existing stats

---------

Co-authored-by: Saúl Ibarra Corretgé <s@saghul.net>

Creating the AudioContext at import time means a warning line is printed
in the JS console even before the user has a chance of initializing the
library, which is weird.

There is no harm in initializing it in init since it's the first thing
the user needs to do in order to use the library.

* 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