Use 'preferredCodec' and 'disabledCodec' under videoQuality config.js settings for setting codec preferences.
Do not prefer VP9 on Firefox because of https://bugzilla.mozilla.org/show_bug.cgi?id=1633876.

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.

* fix: Takes disabled encodings into account when calculating local resolution.
* feat: Adds a new event that's triggered when the max enabled resolution changes.
* feat: Broadcasts the max enabled resolution value along with other stats.

* Initial impl of lobby rooms.

* Fixes tests to check the new fulljid added to MUC_MEMBER_JOINED.

* Updates few of the comments, renaming some functions.

* Renames disableLobby ChatRoom option to enableLobby.

* Fixes a comment.

* Moves setMembersOnly method to ChatRoom.

* Fixes counting members, to exclude jicofo.

* Moves setLobbyRoomJid earlier and renames a method.

Rename _maybeEnableDisable to maybeJoinLeaveLobbyRoom.

* Drops using custom roomconfig lobbypassword field and reuse room lock.

* Handles destroying the lobby room.

* Handles clear lobby room on destroy for moderators.

We do not try to leave the lobby room as it is server-side destroyed and we handle that. The only case of leaving a lobby room is when request to join room is being approved.

* Join main room if lobby is disabled while waiting.

* Adds MEMBERS_ONLY_CHANGED conference event.

* fix: Make sure we leave lobby if main room is joined.

* fix: Setting password when joining locked room.

* fix: Fixes case where we enable lobby for already locked room.

* fix: Fixes case where we enable lobby and then lock room.

* fix: Fixes lint.

* ref: Removes shared password for lobby.

This functionality is handled by the lock room password and handled there.
Removes duplication and unnecessary complicated API for lobby room.

* fix: Fixes comments.

Send conference.join event only after the peer connection is created.
This way we won't send events for lonely calls.

The client will reject P2P session-initiate if the conditions which
allow for P2P session to start are not met(there more or less than 2
human participants in the call).

* Updates kicked event to inform local and remote kicks and who it affects

* Parses actor from MuteIQ and adds participant info to mute track event.

* Fixes emitting both events.

* Fixes comments.

Changes the behavior to actively open new WebRTC Data Channel instead
of waiting for the JVB to do it.

Adds ICE_RESTART_SUCCESS event used to re-initialize the data channel in
case of ICE restart where the conference could have been moved to
another bridge.

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.

Use the W3C Permissions API where possible. At the time of this writing this is
supported in Chrome >= 64.

The code for handling device availability has been disabled for a long time,
plus it's ill named since it represents 2 abstractions: lack of permissions and
lack of devices.

Time for it to rest in the git graveyard.

* feat: Read the server region from Jingle and broadcast it with statistics.

* feat: Adds the bridge count to local "statistics", refactors conference-properties.

* fix: Emits the conference properties with the event, small fixes.

* ref: Orders the imports alphabetically.

As described by @virtuacoplenny:
[T]he ordering is based on import path, not import name, with different
file depths being grouped together, node modules being grouped together
at the top.

* fix: Keeps JitsiConference#properties always defined.

* fix: Does not fire an event when the argument is undefined.

Emit ACTION_JINGLE_SI_TIMEOUT analytics event if it takes too long for
Jicofo to send session-initiate (it usually arrives within 1 second).

* feat: Adds the user's region to presence. Fixes typos.

* feat: Measures end-to-end RTT with ping over the data channel.

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().

* ref(recording): change implementation to match VideoSIPGW

VideoSIPGW takes in a chat room and uses instance variables
on the chat room. RecordingManager has been changed to
mirror this approach because of the case where jitsi is
deployed on a domain requiring authentication. In that
case, the initial chat room is created and fails, a
new chat room (connection) is created for authentication,
authentication is put onto the failed chat room, and the
failed chat room is used. As such, recordingManager
should not be tied to chat room creation itself but
rather tied to the first chat room.

* squash: use git mv to detect capitalization change

* Support layer suspension

Add support for a message which notifies the endpoint whether or not it
is selected (meaning its HD stream is in use).  If it is not
selected and enableLayerSuspension is set to true then it will impose a
bandwidth limit in the SDP to suspend sending the higher layers.

* only add the IS_SELECTED_CHANGED listener in JingleSessionPC if layer
suspension is enabled

this prevents doing a local o/a when we don't need it

* feat(recording): frontend logic can support live streaming and recording

Instead of either live streaming or recording, now both can live together.
The changes to facilitate such include the following:
- Remove the old recording.js module which allowed one recording at a time.
- Remove events for live stream url changes as the url is now part of a
  sesssion and not fired independently.
- Availability of sipgw and recording are gone. Instead sessions have a
  failure reason. For sipgw sessions, store that failure and emit it to
  listeners.
- Create a new recordingManager singleton that can start/stop sessions
  and handle updating known state of those sessions. Known state is
  emitted through one event.
- Create a JibriSession model to encapsulate state of a session.

* update comments, use map to store sessions

* always pass in focusmucjid

* try to fix jibrisession docs and remove default null

* feat(recording): show the YouTube live stream URL

- Pass you_tube_broadcast_id to Jibri so it can create a
  YouTube link for the live stream.
- Emit the liveStreamViewURL (the YouTube link) when it
  is received from the participant doing the recording.

* set member.liveStreamViewURL only when truthy

* Adds new parameter details for the connection failed event.

Adds suspend and shard_changed details.

* Updates the used strophejs version in dependencies.

* Updates the used strophejs version in dependencies (package-lock file).

* Changes to always report suspend time, reports 0 on no suspend detected.

* Updates the used strophejs version in dependencies (1.2.14-1).

* Changes details field name.

* wip: initial version of the new AnalyticsAdapter.

* ref: Restructures the ICE duration and state change events.

* ref: Restructures the JitsiLocalTrack events.

* ref: Restructures the TTFM events.

* ref: Updates the user feedback event.

* ref: Restructures the _CONNECTION_TIMES_ and TTFM events.

* ref: Restructures the BRIDGE_DOWN and NO_DATA_FROM_SOURCE events.

* ref: Restructures the FOCUS_LEFT event.

* ref: Restructures the DATA_CHANNEL_OPEN event.

* ref: Removes the ICE_FAILED event (it is a duplicate of a state change event).

* ref: Restructures the device list events.

Uses one event per device, since the new format does not allow non-atomic attributes.

* fix: Does not obey "unmute" commands from the focus.

* ref: Restructures the "remotely muted" event.

* ref: Restructures the CONFERENCE_ERROR events.

* ref: Removes the CONNECTION_INTERRUPTED event

We can use ICE_STATE_CHANGED instead.

* ref: Renames isreconnect to isReconnect.

* ref: Removes the CONNECTION_RESTORED event. Use ICE state changes instead.

* ref: Restructures the p2p events.

* ref: Restructures the jingle events.

* ref: Restructures the RTP statistics event.

* ref: Restructures the CONNECTION_FAILED and DISCONNECTED events.

* ref: Restructures the getUserMedia analytics events.

* ref: Cleans up AnalyticsEvents and restructures some of the events.

* fix: Adds error logs to the analytics adapter.

* ref: Refactor Statistics.sendEventToAll

Renames to sendEventAndLog, supports the object-based API, uses the
function where appropriate.

* fix: Addresses PR feedback.

* fix: Addresses Lyubomir's feedback.

* ref: Remove unused functions, adds documentation.

* feat: Adds a Statistics.sendAnalytics shortcut.

* ref: Uses the conference name as the default containerId.

* fix: Adrdesses Lenny's feedback.

* fix: Addresses more feedback.

* fix: Uses 'operational' as the default event type.

* doc: Updates the documentation.

* fix: Fixes adding of permanent properties.

* ref: Uses consistent naming for events' attributes.

Uses "_" as a separator instead of camel case or ".".

* feat: Adds the conference name as a permanent property automatically.

* ref: Don't expose Setting.machineId.

* fix: Adds a "p2p" attribute to jingle events.

* ref: Uses "action" instead of "name".

* ref: Uses underscore in events' attribute names.

* ref: Logs a message to the logger/console

instead of callstats in sendAnalyticsAndLog().

Adds rtc.LOCAL_TRACK_SSRC_UPDATED event to be emitted when ssrc is updated for a local track.
Fixes adding rtc listeners for CREATE_ANSWER_FAILED, CREATE_OFFER_FAILED, SET_LOCAL_DESCRIPTION_FAILED, SET_REMOTE_DESCRIPTION_FAILED.

* Changes initialization of videoSIPGW.

* Adds some errors returned on creating videoSIPGW session.

* Fixes sending videoSIPGW session STATE_CHANGED event.

* Fixes sending jibriIQ, no result status is received.

* Adds VIDEO_SIP_GW_SESSION_STATE_CHANGED to JitsiConferenceEvents.

* Fixing comments.

* ref: Simplifies the logic for handling an incoming jingle session-initiate.

* fix: Don't redundantly log cross region

information under a field name called "label".

* cleanup: Simplifies code. Adds the userAgent as a permanent property

for statistics (so that the client doesn't have to).

* ref: Names the parameter which specifies the name of the event "eventName".

* ref: Extracts event names to AnalyticsEvents.

* ref: Exports and imports constants individually.

* fix: Fixes CONNECTION_TIMES event names.

* ref: Arranges constants alphabetically.

* ref: Adds line breaks.

via JSON messages received via the Chatroom

* doc(JitsiConference): deprecate 'isInLastN'

The 'isInLastN' method should not be used for the UI purposes, but
ParticipantConnectionStatus value should be used instead.

* fix(ParticipantConn..Status): speed up INACTIVE transition

Before this change when user's video stops playing, after user is
removed from last N we were waiting 2 seconds, before going to INACTIVE
state. This commit reduces the time to 500ms for such case.

* fix(ParticipantConn...): reduce logging

Reduce logging verbosity.

* fix(ParticipantConn...): handle LastN == 0

When LastN is set to 0 we should not rely on video playback and last N
set for figuring out participant connection status.

* fix(JitsiConference): undefined participants

Fixes a crash when this.participants field is accessed from _init.

* ref: store SSRCs as a number

Converts all the places where SSRCs where stored as string to use
numbers.

* doc(RTPStatsCollector): getNonNegativeStat

Fixes invalid description about returning NaN.

* ref(JitsiConf...EventManager): simplify for..of

* ref(JitsiRemoteTrack): throw TypeError

Will throw a TypeError when 'ssrc' is not a number.

* fix(RTPStatsCollector): invalid reference

* doc(RTPStatsCollector): getNonNegativeStat private

* ref(RTPStatsCollector): simplify for..of

* fix(SSRCs): check for negative value

Will not accept negative SSRCs, since those are supposed to be unsigned.

* Adds video sip gw handling.

Parsing presence and availability of the service. Adding possibility to start/stop a video sip gw session.

* Fix comments.

* Adds more jsdoc about create session and renames import to use the filename is imports.