It doesn't get translated in the TS build, for one.

Script I used:

```python
import os

for (dirpath, dirnames, filenames) in os.walk('.'):
    if '.git' in dirpath:
        continue
    if 'node_modules' in dirpath:
        continue
    if 'dist' in dirpath:
        continue
    if 'types' in dirpath:
        continue
    for filename in filenames:
        path = os.path.join(dirpath, filename)
        if not path.endswith('.js') and not path.endswith('.ts'):
            continue
        #print(path)
        with open(path, 'r+') as f:
            #print(f)
            data = f.read()
            if '__filename' in data:
                p, ext = os.path.splitext(path)
                txt = f"'{p[2:]}'"
                print(txt)
                data = data.replace('__filename', txt)  # Assign the result back to data
                f.seek(0)
                f.write(data)
                f.truncate()
```

Avoid duplicates.

* Change jid to participantId

* Define participantId

* feat(silent): track if participant joined without audio

* Fix failing tests, use jitsi.org namespace

---------

Co-authored-by: Дамян Минков <damencho@jitsi.org>

* 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>

Use an exponential backoff time for initiating ICE restart to prevent loading the prosody.

When a large number of endpoints loose connection to the signaling and media servers because of an ISP failure, these endpoints will try to reconnect at the same time causing a storm of source-remove/source-add events to the signaling servers that can lead to prosody going down otherwise.

When the bridge WS is re-established, jvb sends the full map of audio and video sources. Without the check, the library will end up firing TRACK_REMOVED and TRACK_ADDED for all the exiting tracks.
Also, process audio and video source maps only on JVB sessions.

When an existing SSRC for a remote track gets re-mapped from one source to another, fire a TRACK_REMOVED event followed by TRACK_ADDED event instead of TRACK_OWNER_CHANGED event. This should simplify the application logic for track handling.

* feat(visitors): Fires events for received visitors message promotion request/response.

* squash: Updates logs.

* ref: Remove PrecallTest.

* ref: Remove callstats support.

* feat: Handles join error about required display name.

* squash: Fixes tests.

* squash: Adds param to DISPLAY_NAME_REQUIRED event indicating is it coming because of lobby.

* squash: Drops all clear up methods for unrecoverable errors.

The jitsi-meet part is taking care of it.

* feat(visitors): Visitors chat.

For messages coming from visitors, which are not in the current room we need to supply a display name.
Mark messages with custom display name as coming from guest.

* feat: Adds a visitors version to conferenceIQ.

<iq id='some-id' from='focus.exmaple.com' type='error' xml:lang='en-US' to='some-jid' xmlns='jabber:client'>
    <error type='modify'>
        <redirect xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'/>
        <conference xmlns='http://jitsi.org/protocol/focus' vnode='v1' main-room='roomtest@conference.focus.exmaple.com' focusjid='focus@auth.meet.jitsi'/>
    </error>
</iq>

It's emitted when remotely closed / failed.

Instead of Jicofo signaling all the remote sources available in the call, the bridge now signals only a limited set of SSRCs and then rewrites the SSRC on the outgoing media streams. The SSRC mapping is done based on the sources requested by the clients through the receiver constraints. This limits the number of m-lines in the remote/local SDPs on the client and therefore results in better performance in large calls.

* Handle source remapping messages from bridge
* Added track_owner_changed events
* don't process an invalid rtx ssrc.
* keep track of remote ssrcs, only renegotiate on new ones.
* Change source name on remote track on ssrc remapping.
* Don't remove tracks on member leave.
* Remove (orphaned) tracks on session terminated.
* Use serial number (per media type) to create msid attribute.
* Update videoType on remapping.

Co-authored-by: James A <jqdrqgnq@users.noreply.github.com>

When the participant has joined after A/V moderation has been enabled
they are auto muted with the startmuted node in the session-initiate
packet. We were not marking those as muted by the focus and therefore on
unmute we were not sending the <muted>false</muted> IQ to jicofo.
The result was that the bridge wasn't forwarding the media to the
remote participants.

Discovered it during investigation of
https://github.com/jitsi/jicofo/issues/905

* Revert "fix: Cleans up rooms after failure to join."

This reverts commit f23372dd88.

* fix: Cleanups JitsiConference on reservation and max user failures.

Subsequent attempts to join will show produce an error "[JitsiConference.js] : A coference with the same name has already been created!".

They are companion rooms created in a separate MUC. The room relationship is
maintained by a Prosody plugin.

All signalling happens through the breakout rooms MUC component.
Co-authored-by: Saúl Ibarra Corretgé <saghul@jitsi.org>

* feat: Adds to JitsiConferenceEvents CONFERENCE_JOIN_IN_PROGRESS.

* squash: Fixes tests.

* feat(av-moderation) Remove from moderation whitelist functionality

* Fix comment

* fix(recording): Send resource if local participant started the recording

* Updates js docs.

Co-authored-by: Дамян Минков <damencho@jitsi.org>

* feat: Audio/Video moderation.

* squash: Fix docs.

* squash: Adds some warning logs when execution is rejected.

* squash: Changes a field name in the message for adding jid to whitelist.

* squash: Send to participants only message about approval.

Skips sending the whole list.

* squash: Fixes tests.

* squash: Adds more logs.

* feat: Separates enable/disable by media type.

Adds actor to the messages to inform who enabled it.

* squash: Fixes log line.

* squash: Fixes comments.

* squash: Fixes log messages.

* squash: Fixes comments.

Use the new Colibri message "EndpointStats" for broadcasting the local stats. The bridge then will be able to filter the endpoint stats and send them only to the interested parties instead of broadcasting it to all the endpoints in the call.

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.