fix(conference) Fix different bugs with startMuted policy. (#2777)
* fix(conference) Fix different bugs with startMuted policy.
If the user changes mute state after joining the call and before the other participats join the call, make sure those state changes are honored.
* feat(startMuted) Ignore startMuted in Jingle from focus.
Apply startMuted policy based on startMuted setting in room meta data.
Moderator settings and AV moderation settings will now be propagated to new endpoints joining the call through room meta data.
Advertise a new capability 'start-muted-room-metadata' to Jicofo for feature detection.
feat(RTC) drop PERMISSION_PROMPT_IS_SHOWN event (#2609)
* feat(RTC) drop PERMISSION_PROMPT_IS_SHOWN event
It was never accurate since browsers have changed their behavior over
time and can be implemented in the application more reliably.
* squash: Update JitsiMeetJS.ts
* squash: Update hand-crafted JitsiMeetJS.d.ts
---------
Co-authored-by: Дамян Минков <damencho@jitsi.org>
* ref(SDPDiffer) Convert to ES6 class.
Make it work directly with unified plan SDP that has multiple m-lines and add more unit tests.
* ref(xmpp) translate unified-plan SDP->Jingle directly.
Without having to run it through the SDPInterop.toPlanB cycle.
* ref(SDP) Always generate the MSID for signaling it to Jicofo.
* fix(SDPDiffer) Check explicitly for ssrc changes
* fix(SDP): Fix comments and cleanup.
Remove LOCAL_TRACK_SSRC_UPDATED event as the application ignores the event and no additional action needs to be taken when that event is fired.
* ref(SDP) Add a helper function for parsing the 'a=ssrc-group' line.
* squash: Address review comments
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.
Local and remote Jitsi tracks expose a reference to the
`JitsiConference` through `track.conference`. This was not available in
the TypeScript definitions.
Local tracks are initialized with `conference = null` but will be set
later, assuming the caller adds the track to a conference. Remote tracks
are always associated with a conference.
fix(video-quality): Adjust encodings bitrates/scalefactor for high capture resolutions (#2436)
* fix(video-quality): Adjust encodings bitrates/scalefactor for high capture resolutions.
When the client is configured to capture camera sources at 1080p or higher, adjust the bitrates and the encodings scale factor accordingly.
* squash: Address review comments
Fix the bitrates set on the remote description and also bump up the bitrates for 1080p, 4k and screenshare.
* squash: Address some more review comments.
* fix(RTC): Remove support for plan-b SDP.
Since unified plan mode becomes the default mode, so does multi-stream mode.
* squash: Remove plan-b unit tests and fix comments
* squash: address review comments
Unify events and output single TypeScript declaration (#2407)
* 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
All modern browsers implement window.addEventListener for the 'error'
and 'unhandledrejection' listeners. Thus there is zero benefit from
using the legacy `onerror` handler here and rolling multiple handlers
ourselves.
There is an old config option `enableWindowOnErrorHandler` which was
never whitelisted in Jitsi Meet, so I'm 100% sure nobody is using it.
Even if someone was, migrarting to the above is simply better in every
possible way.
Add SVC and Simulcast support for AV1 and VP9. (#2350)
Add SVC and Simulcast support for AV1 and VP9.
The default mode is the full SVC mode for VP9. It works as expected with the current version of JVB.
AV1 in the client is only supported when the bridge negotiates the Dependency Descriptor extension headers which are needed for reading the layer information for AV1 and H.264 video frames.
* feat: Enable H.264 simulcast support.
Multi-encoding simulcast for H.264 is supported now because of the DD header ext support added on the bridge side.
* fix(video-quality): Add default bitrates for all codecs.
Expect the videoQuality settings in the new format, deprecated configs are still supported.
videoQuality: {
AV1: {
maxBitratesVideo: {
low: 100000,
standard: 300000,
high: 1000000,
ssHigh: 1500000
},
useScalabilityModeAPI: {
enabled: true,
useSimulcast: true, //(defaults to SVC),
useKSVC: true //(defaults to L3T3_KEY)
},
},
H264: {
maxBitratesVideo: {
low: 200000,
standard: 500000,
high: 1500000,
ssHigh: 2500000
}
},
VP8: {
maxBitratesVideo: {
low: 200000,
standard: 500000,
high: 1500000,
ssHigh: 2500000
}
},
VP9: {
maxBitratesVideo: {
low: 100000,
standard: 300000,
high: 1200000,
ssHigh: 2500000
},
useScalabilityModeAPI: {
enabled: true,
useSimulcast: true, //(defaults to SVC),
useKSVC: true //(defaults to L3T3_KEY)
},
}
}
* Disable encodings when requested height for screenshare is 0.
* feat: add unit tests for scalability modes.
* ref: Move all encoding configuration calculation to TPCUtils.
* Add more unit tests for VP9 K-SVC mode.
* Make codec name in settings case insensitive.
Also read the deprecated max bitrates correctly, add unit tests to test it.
* ref(TPC): isSimulcastOn -> isSpatialScalabilityOn.
It makes more sense to call it spatial scalability than simulcast now that full SVC support is available.
* fix: Negotiate AV1 DD header exts only for AV1 and H.264.
* fix: Rename setting useL3T3Mode->useKSVC and adjust the defaults.
* fix(codec-selection) Use a configurable preferred codecs list to select the codecs.
Allow asymmetric codecs to be configured on the endpoints. This means that Firefox and Safari which have bugs with VP9 encode will now encode VP8 but decode VP9 coming in from Chromium endpoints.
* fix(codec-selection) Add unit tests for the codec selection logic.
* feat(codec-selection): Introduce mobileCodecPreferenceOrder setting in config.js.
* fix(codec-selection) Avoid unnecessary renegotiations.
On participant join/leave, check if the new intersection of codecs are already configured to be the top n codecs.
* Address review comments
* fix: Strip the codecs that are not in the codec list from the video m-lines.
Also, ignore remote codecs published in presence for p2p connections.
* fix: Define default codec order for mobile and desktop.
Fix an issue where p2p between mobile and desktop was broken.
* fix: Add default codecs for both p2p and jvb on mobile devices.
Munge the initial offer sent out by RN clients since RN doesn't support RTCRtpTransceiver#setCodecPreferences.
fix(TPC) Disable media instead of changing dir for p2p->jvb switch. (#2226)
* fix(TPC) Disable media instead of changing dir for p2p->jvb switch.
Resume or suspend the media on the jvb peerconnection by changing the RTCRtpEncodingParamters.active state instead of changing the direction on the transceiver. This avoids the needs to start a O/A renegotiation cycle for these operations. The media direction will be changed only for p2p lastn=0 case since video needs to be disabled on both the sender and the peer for p2p lastn=0 case.
* Address review comments
* Disable media after adding source while media is suspended on the jvb connection. Default 'active' state for stream encodings after the source is added is 'true'.
* Wait for all the promises to be settled before returning
fix(SignalingLayer) Update SSRC owners on leave. (#2184)
* fix(SignalingLayer) Update SSRC owners on leave.
Update the SSRC owners in the following cases:
1. When a remote endpoint leaves the call.
2. When a source-remove is received.
3. When a source is remapped (with ssrc-rewriting enabled).
Create the remote track even if presence is not yet received. The ssrc owner check prevents the client from creating a dummy track when the call switches over from p2p to jvb when the last remote endpoint leaves the call.
* ref(SignalingLayer) alpha sort methods.
Clean up unused methods, _findEndpointSourceInfoForMediaType is not used anymore.
* squash: Address review comments.
fix(quality-control) Fix receiver constraints for default case.
When the application doesn't set receiver constraints, generate default constraints in the source-name mode.
Cleanup code related to the endpoint based video constraint handling for p2p connection.
fix(qualitycontrol): Cleanup old receiver constraints.
Endpoint based receiver constraints and other endpoint based bridge signaling messages are no longer supported by latest JVB after the switch to source-name signaling.
Rename method names 'sendNewReceiverVideoConstraintsMessage'->'sendReceiverVideoConstraintsMessage', 'setNewReceiverVideoConstraints'->'setReceiverVideoConstraints'
Remove support for legacy endpoint based signaling. (#2147)
* Remove support for legacy endpoint based signaling and make source-name signaling as the only signaling mode. The legacy screensharing mode will no longer be supported. The client will still be able to process remote presence sent in the old format to support interop with very old mobile clients and jigasi that do not support source-name signaling.
* remove code related to presenter mode.
Jaya Allamsetty
Jaya Allamsetty
Saúl Ibarra Corretgé
Pawel Domas
damencho
Jesse Gibson
Roman Melnyk
Daniel McAssey
Auxane Thouary
Jesse Gibson
Saúl Ibarra Corretgé
Andrei Gavrilescu
Jorge Oliveira
dependabot[bot]