* feat(tracks) Clean up the track if a source addition is rejected.
When jicofo rejects a source-add because of sendder limits, dispose and remove the local track from the conference.
* chore(deps) update LJM to latest.
* fix(ChatRoom) make sure we wait for all promises on leave()
* fix(ChatRoom) make sure EMUC is destroyed
* fix(JitsiConference) make sure RTC is always destroyed
* fix(log) don't log full stanzas
* fix(avmoderation,breakout-rooms) dispose handlers when leaving
6b3d3d2783...61aef90835
* feat: Handle disableBeforeUnloadHandlers option.
* feat(conference) Implement audio/video mute disable when sender limit is reached. Jicofo sends a presence when the audio/video sender limit is reaced in the conference. The client can then proceed to disable the audio and video mute buttons when this occurs.
f4f7db2e5f...6b3d3d2783
* fix(TPC): Select 1 as the default scale factor for p2p. This fixes an issue where a user is not able to unmute their video if the MediaStreamTrack associated with the camera stream returns a null value for the track height.
* Save track source name to JitsiRemoteTrack
bdfbb82087...a1c966058c
* TPC: make the comments more descriptive.
* fix(SignalingLayerImpl): Log an error when only the ssrc owner gets overwritten with a diff ep id.
* fix(TPC): Force reneg when user unmutes the first time. This ensures that the source signaling is sent before the mute state is sent in presence. Jicofo relies on mute state from presence to check if the sender limit has been reached.
6d981ebb6c...131b9458fe
* Initial implementation; Happy flow
* Maybe revert this
* Functional prototype
* feat(facial-expressions): get stream when changing background effect and use presenter effect with camera
* add(facial-expressions): array that stores the expressions durin the meeting
* refactor(facial-expressions): capture imagebitmap from stream with imagecapture api
* add(speaker-stats): expression label
* fix(facial-expression): expression store
* revert: expression leabel on speaker stats
* add(facial-expressions): broadcast of expression when it changes
* feat: facial expression handling on prosody
* fix(facial-expressions): get the right track when opening and closing camera
* add(speaker-stats): facial expression column
* fix(facial-expressions): allow to start facial recognition only after joining conference
* fix(mod_speakerstats_component): storing last emotion in speaker stats component and sending it
* chore(facial-expressions): change detection from 2000ms to 1000ms
* add(facial-expressions): send expression to server when there is only one participant
* feat(facial-expressions): store expresions as a timeline
* feat(mod_speakerstats_component): store facial expresions as a timeline
* fix(facial-expressions): stop facial recognition only when muting video track
* fix(facial-expressions): presenter mode get right track to detect face
* add: polyfils for image capture for firefox and safari
* refactor(facial-expressions): store expressions by counting them in a map
* chore(facial-expressions): remove manually assigning the backend for tenserflowjs
* feat(facial-expressions): move face-api from main thread to web worker
* fix(facial-expressions): make feature work on firefox and safari
* feat(facial-expressions): camera time tracker
* feat(facial-expressions): camera time tracker in prosody
* add(facial-expressions): expressions time as TimeElapsed object in speaker stats
* fix(facial-expresions): lower the frequency of detection when tf uses cpu backend
* add(facial-expressions): duration to the expression and send it with durantion when it is done
* fix(facial-expressions): prosody speaker stats covert fro string to number and bool values set by xmpp
* refactor(facial-expressions): change expressions labels from text to emoji
* refactor(facial-expressions): remove camera time tracker
* add(facial-expressions): detection time interval
* chore(facial-expressions): add docs and minor refactor of the code
* refactor(facial-expressions): put timeout in worker and remove set interval in main thread
* feat(facial-expressions): disable feature in the config
* add(facial-expressions): tooltips of labels in speaker stats
* refactor(facial-expressions): send facial expressions function and remove some unused functions and console logs
* refactor(facial-expressions): rename action type when a change is done to the track by the virtual backgrounds to be used in facial expressions middleware
* chore(facial-expressions): order imports and format some code
* fix(facial-expressions): rebase issues with newer master
* fix(facial-expressions): package-lock.json
* fix(facial-expression): add commented default value of disableFacialRecognition flag and short description
* fix(facial-expressions): change disableFacialRecognition to enableFacialRecognition flag in config
* fix: resources load-test package-lock.json
* fix(facial-expressions): set and get facial expressions only if facial recognition enabled
* add: facial recognition resources folder in .eslintignore
* chore: package-lock update
* fix: package-lock.json
* fix(facial-expressions): gpu memory leak in the web worker
* fix(facial-expressions): set cpu time interval for detection to 6000ms
* chore(speaker-stats): fix indentation
* chore(facial-expressions): remove empty lines between comments and type declarations
* fix(facial-expressions): remove camera timetracker
* fix(facial-expressions): remove facialRecognitionAllowed flag
* fix(facial-expressions): remove sending interval time to worker
* refactor(facial-expression): middleware
* fix(facial-expression): end tensor scope after setting backend
* fix(facial-expressions): sending info back to worker only on facial expression message
* fix: lint errors
* refactor(facial-expressions): bundle web worker using webpack
* fix: deploy-facial-expressions command in makefile
* chore: fix load test package-lock.json and package.json
* chore: sync package-lock.json
Co-authored-by: Mihai-Andrei Uscat <mihai.uscat@8x8.com>
* fix(e2ee) disable p2p when e2ee is enabled
* fix(e2ee) fix race condition when restarting media sessions
* fix(p2p) fix error if p2p session is stopped while accepting it
* fix(e2ee) removed no longer needed code
* feat: SourceVideoTypeMessage message
* chore(lint) tame the new linter
* chore(deps) update Babel and ESLint to the latest versions
* chore(deps) adapt to logger package rename
* fix(e2ee): fix loading web worker when using a relative path inside a blob for the E2EE context
* * fix(sdp): provide SCTP streams, because the XMPP parser expects them
c193f0d433...51f77cbd51
* feat(recording): Add unexpected-request error
* fix(xmpp): use RTX with Firefox from 96 on only
* fix(sdp): update data channel to RFC format
* sdp: switch port to 9 and rtp protocol to UDP/TLS/RTP/SAVPF (#1697)
03bc5278da...c193f0d433
* fix(presence) Send presence on mute state change.
* fix(TPC) change the tranceiver dir to recvonly when track is removed. This fixes occasional failures of MuteTest.MuteAfterJoinCanShareAndUnmute torture test and also the case on Safari where user stopping the screenshare doesn't stop showing the screensharing indication on the thumbnail.
* fix: Avoid sending two presences if start muted and then screen share. (#1771)
* feat: use source names in presence
* ref: move SignalingLayer to the conference
* feat: Delays deployment info stats till we get update from backend. (#1770)
e566291864...e6b330186f
* ref(JitsiConference) Remove remote tracks from conf before reneg is done. We do not have to wait for the removal of the ssrcs from the remote description for removing the remote tracks associated with a participant that left the call. This speeds up removal of the participant from call even if the JingleSession modification queue is backed up.
* faet(SDP): Add test for jingle JSON format.
42c675249a...111e50c38a
* ref(JingleSessionPC) Do not renegotiate on every local source change. Instead rely on the 'negotiationneeded' event fired by the browser for JVB connection. This makes local source changes faster even if the modification queue is backed up.
75d3106544...012c38769d
* feat: Hides prejoin screen on conference in progress event.
We enter the conference view as early as possible on conference in progress as the joined event can be late in a big conference.
Also, we show conference view only when joining is in progress, for example, the with the lobby enabled where we try to join but fail, we do not want to show the conference view for a fraction of a second before showing lobby screen.
* squash: Drops CONFERENCE_JOIN_IN_PROGRESS.
* squash: Updates ljm with the new JitsiConference event.
* squash: Adds some debugs to the github action.
Easier to catch problems with package-lock.json file.
* feat(identity): add region identity parsing
* fix(IceFailedHandling) force client reload when ICE fails locally.
* fix(iOS15) fix not being able to unmute if "everyone starts muted" is set
* fix: logger calling wrong function typo
* feat: generates source names (#1725)
b5288c2989...febd9087b9
Changed screen capture to non effect. Effects are used to alter the stream, this feature does not need to alter the stream, it just needs access to it
Changed image diff library. Previous library diff’ed the whole image, the new one has en early return threshold
Use ImageCaptureAPI to take the screenshot. Added polyfill for it and polyfill for createImageBitmap
Added analytics
* fix(xmpp): disable RTX for Firefox < 93, because it results in random SSRC order
* fix(Jingle): stop reverting the SSRCs from Firefox
c15dda1537...7a56f7b341
* fix(browser-support) fix detecting iOS browsers correctly
* fix(JitsiConference):2 instances for the same room
* ref: sendMuteStatus is not async
ae70962bfa...c15dda1537
* fix(Jingle) Log the extracted info from Jingle IQ.
* ref(Jingle) Alpha sort and prefix the local functions with '_'
* ref(Jingle) Log formatted source information. Instead of logging the full IQs for Jingle messages like session-initiate, source-add and source-remove which can be very long, log just the formatted source information.
* ref(RTC) rename iceConfig to pcConfig. It makes more sense to call it pcConfig since it is the RTCConfiguration object passed to the WebRTC peerconnection.
* fix(logging) Log only the imp events on remote tracks. Log only the important events that we care about on the HTMLMediaElement that the remote tracks are attached to.
0646bc3403...ae70962bfa
It updates the main language file for a given locale from the canonical one and
sets the empty string on the missing keys. No longer used keys are discarded.
* feat(av-moderation) Ask to Unmute and remove from Whitelist
Make Ask to Unmute work without moderation
Add remove from moderation whitelist functionality
* chore(deps) lib-jitsi-meet@latest
* feat(av-moderation) Remove from moderation whitelist functionality (#1729)
* fix(chore) corrected typo in log message
* fix(e2ee) replace nullish coalescing with or
* fix(e2ee) restore initial key when RATCHET_WINDOW_SIZE reached
3b8baa9d3b...0646bc3403
Co-authored-by: Дамян Минков <damencho@jitsi.org>
* fix(JitsiConference) Check for room before calling isFocus method on the room object.
* fix(Jingle) Reverse the order of ssrcs signaled for Firefox. This fixes an issue where the bridge doesn't forward the HD stream from Firefox to other users in the call. The order of the ssrcs produced by the browser is from Highest resolution to lowest whereas the bridge assumes it to be from lowest to highest as is the case in Chrome and Safari.
* fix(codec-selection): Impose VP9 bitrates only when VP9 is the negotiated codec. If Jicofo doesn't offer VP9 but the client expresses a preference for VP9, VP9 bitrates were being imposed before.
609e3d5a1a...3b8baa9d3b
* fix: Reads shard name and from disco-info if available.
* chore(deps): bump sdp-interop to get another fix for ICE restart
* update sdp-interop to include Unified ICE restart fix
fbf85bdcec...609e3d5a1a
* fix(LocalSdpMunger): do not fake video sdp when screen sharing
* fix(JitsiConference) avoid extra processing if the room was left
* fix(moderator) remove unneeded log
b0d27fa8da...28a5355356
* fix(browser-support): Add audio track to pc always on mobile Safari. On mobile Safari, if a user joins audio and video muted, the browser doesn't decode the incoming audio. Workaround is to always add the audio track to pc and mute it if needed.
* feat: JSON encoded sources. (#1695)
2820d649ea...b0d27fa8da
* fix(ProxyConnection) add new required stubs
* fix(tpc) fix extracting ssrc map when using single stream
* fix(transcribing): send transcripton_language only when necessary (#1677)
97ff597425...6eaffc4b11
When participants panel is open and we approve a participant to unmute, the notification was not hidden as we were not correctly updating the state. We were expecting a participant object, but an id of the participant was used.
* fix(ConnectionQuality): Do not show red/yellow GSM bars on join. When the user first unmutes their video, the connection quality is shown as poor until the local stats are available. Calculate the connection quality only after the stats are available, i.e., assume 100% until pcStatsInterval has elapsed.
* feat(non-participant-messages) Add a new JitiConferenceEvent for messages ignored by ENDPOINT_MESSAGE_RECEIVED
* fix(precall) respect custom callstats script url for precall test
9e632a77c5...6a3df11ffa
* feat(BridgeChannel): Signal a new videoType for high fps screenshare. This lets the bridge adjust the bitrate allocation for this source so that layers with higher fps are prioritized over layers with higher resolution. As a result, endpoints with restricted downlink will receive a high fps low resolution share as opposed to a high resolution low fps screenshare.
* fix(log) lower severity of overly verbose logs (2)
fa834c2923...9e632a77c5
In version 1.15 the storage backend was rewritten, which hopefully allows us to
fix this crash on Android:
Caused by java.lang.IllegalStateException: attempt to re-open an already-closed object: SQLiteDatabase: /data/user/0/org.jitsi.meet/databases/RKStorage
at android.database.sqlite.SQLiteClosable.acquireReference(SQLiteClosable.java:55)
at android.database.sqlite.SQLiteDatabase.queryWithFactory(SQLiteDatabase.java:1160)
at android.database.sqlite.SQLiteDatabase.query(SQLiteDatabase.java:1036)
at android.database.sqlite.SQLiteDatabase.query(SQLiteDatabase.java:1204)
at com.reactnativecommunity.asyncstorage.AsyncStorageModule$1.doInBackgroundGuarded(AsyncStorageModule.java:159)
at com.reactnativecommunity.asyncstorage.AsyncStorageModule$1.doInBackgroundGuarded(AsyncStorageModule.java:146)
at com.facebook.react.bridge.GuardedAsyncTask.doInBackground(GuardedAsyncTask.java:35)
at com.facebook.react.bridge.GuardedAsyncTask.doInBackground(GuardedAsyncTask.java:19)
at android.os.AsyncTask$2.call(AsyncTask.java:305)
at java.util.concurrent.FutureTask.run(FutureTask.java:237)
at com.reactnativecommunity.asyncstorage.AsyncStorageModule$SerialExecutor$1.run(AsyncStorageModule.java:63)
at java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:1133)
at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:607)
at java.lang.Thread.run(Thread.java:760)
THe new version fixed a longstanding problem with RN not updating the JS side
SDP representation properly. This will allow us to remove a hack we currently
have to sidestep this.
* feat(JingleSessionPC): Enable unfied plan by default for chrome p2p.
* fix(JingleSessionPC): Fix startMuted cases for p2p unified plan. Chrome doesn't create a decoder for ssrc in the remote description when there is no local source and the endpoint is offerer. Initiating a renegotiation with the endpoint as a responder fixes this issue. Add a workaround until Chrome fixes this bug.
* fix: Missed SSRCs in Unified Plan with several "ssrc-group:FID" groups. (#1658)
e6648fac96...b815157a22
* fix(TPC): Fix the screenshare issue when user starts video muted on chrome. Munge 3 ssrcs in the SDP for chrome in unified plan always for the simulcast case.
053a26604d...e6648fac96
* fix(JingleSessionPC): Disable unified-plan for p2p chrome. Do not enable unified plan for p2p chrome by default until StartMutedTest is fixed. Fix media direction for case when there are no local and remote sources, should be set to 'inactive' in that case.
3a313a244d...053a26604d
* fix(codec-selection): Fix VP9 codec switching issue in Chrome unified-plan. Munge only the m-line that corresponds to the source that the browser will be sending. Do not select VP9 on Firefox. Detect support for RTCRtpTransceiver#setCodecPreferences correctly.
89a7e2d9cd...3a313a244d
* fix(RTC): In unified-plan mode, disable the low resolution streams for low fps SS. In unified plan impl, it is not possible to enable/disable simulcast during the call since the same sender is re-used for all local video tracks. Therefore, disable the low resolution simulcast streams for low fps screensharing so that the bridge forwards only the highest resolution stream which is important for low fps screensharing.
f974007ca6...89a7e2d9cd
* fix(JingleSessionPC): Do not check if the ssrc already exists in the RD when adding a ssrc-group from source-add.
* feat: Switch to unified plan on chrome by default unless explicitly disabled.
* fix(VADAudioAnalyser): NPE error evaluating this._vadEmitter.on (#1652)
* Small fix in tokens doc.
b43a9fa0ee...f974007ca6
* fix(TPC): Do not remove ssrcs from remote desc for p2p. In unified plan, re-use of m-line (i.e., adding an SSRC, removing it and then adding it back) causes the browser to not render the media on Chrome and Safari. The WebRTC spec is not clear as to how browsers have to behave, this doesn't cause any issues on Firefox. As a workaround, only change the media direction and leave the ssrc in the remote desc. This automatically triggers a 'removetrack' event on the associated MediaStream and the track can be removed from the UI.
* Drops old prosody versions from the tokens instructions
0cdfb79c2e...b43a9fa0ee
* squash: Set capScreenShareBitrate flag every time a new pc is created.
* feat(RTC): Add the ability to change desktop share fps. Provide a method for changing the capture fps for desktop tracks during the call. These changes to the lib are needed for making it configurable from the UI.
46ec23fcdc...229015a6f3
* fix(RTC): Do not overwrite other constraints when resolution option is used. When the resolution option was being used, all the other constraints like frameRate and facing mode were being overwritten.
24627e1b95...46ec23fcdc
* fix(TPC): Filter ssrcs differently while extracting the SSRC map from SDP. Use 'msid' for plan-b clients and 'cname' for unified-plan clients.
fad985e95a...d5e60583b8
* fix(TPC): fix local resolution/fps stats. Browsers do not generate a 'msid' attribute for ssrcs in unified plan mode, use mediaType as a key for the TrackSSRCInfo map.
* fix(recording): Send participant id when recording starts/stops (#1632)
8057f12a39...2259d44185
* fix(RTC): Adjust the media direction for p2p conn. For p2p connections, the media direction needs to be adjusted after every source-add/source-remove is processed based on the availability of local sources.
* fix(RTC): Use a enum for media direction.
5738c80baf...d9d9b7fc31
* fix(JingleSessionPC): Disable unified-plan for p2p. Disable cross browser p2p using unified plan until all the issues are fixed.
0993c8e93d...5738c80baf
* fix(LocalSdpMunger): Fix unit test.
* fix(CodecSelection): Call RTCRtpTransceiver#setCodecPreferences before renegotiation. Call RTCRtpTransceiver#setCodecPreferences with the preferrred codec order before every createOffer/createAnswer. This ensures that the codec preference is enforced even when there is no local description available yet while the preferred codec is being set immediately after media session creation.
* fix(JingleSessionPC): Add a workaround for chrome issue. The 'signalingstatechange' event for 'stable' is fired after the 'iceconnectionstatechange' event for 'completed' is fired on chrome in Unified plan. This prevents the client from switching the media connection to the p2p connection once the ice connection for p2p gets established.
* fix(Logging): Log enhancements. Add a preifx to logs for idenitifying the type of TPC/jingleSessionPC.
* feat(TPC): Enable unified-plan support for Chromium based browsers. This can be controlled through the config.js option 'enableUnifiedOnChrome'.
* fix(TPC): Do not configure encodings on Safari until reneg. Avoid configuring the encodings on Chromium/Safari until simulcast is configured for the newly added track using SDP munging which happens during the renegotiation.
* fix(TPC): Do not configure encodings on chromium immediately after replace track. Avoid configuring the encodings on chromium immediately after replace track since the encoding params are read-only until the renegotation is done.
* fix: send json message (#1180)
be3e2a69f2...3fb44f7695
* fix(SDP): Add missing msid for p2p sources.
* fix(TPC): Don't convert plan-b<->unified-plan SDPs for p2p.
* squash: Implement review comments.
* fix(JingleSessionPC): Do not try to re-use inactive mid for new remote ssrcs. The direction was marked as 'inactive' only on Firefox as Safari had audio issues when an inactive mid is re-used. Chrome (in unified-plan) needs the direction of the mid in remote desc to be set to 'inactive' for a 'removetrack' to be fired on the associated media stream whenever a remote source is removed.
* fix(SDP): Drop SSRCs whenever the transceiver direction is 'inactive' or 'recvonly'. This is needed only for JVB connections. Add unit tests for LocalSdpMunger.
* fix: Ignore startAudioMuted/startVideoMuted for p2p. The tracks will not be added when the call switches from jvb to p2p for an endpoint that joins muted by focus.
* fix(RTC): Do not suppress the source updates on Firefox. If the msid attribute is missing, then remove the ssrc from the transformed description so that a source-remove is signaled to Jicofo. This happens when the direction of the transceiver (or m-line) is set to 'inactive' or 'recvonly' on Firefox. Not signaling these source updates creates issues with remote track handling on the other endpoints in the call.
* fix(RTC): Set transceiver direction after RTCRtpSender#replaceTrack. This fixes the issue where TRACK_REMOVED event is not fired when a remote track is removed from the peerconnection. Fixes https://github.com/jitsi/lib-jitsi-meet/issues/1612 and https://github.com/jitsi/jitsi-meet/issues/8482.
60c5667957...be3e2a69f2
* fix(caps): Disable TCC on Firefox. There is a known issue with Firefox where the BWE gets halved on every renegotiation causing the low upload bitrates from the Firefox clients.
* fix: Drops unused config, fixesjitsi/lib-jitsi-meet#1620.
* fix(e2ee): destroys olm session on disabling e2ee
f95a455c08...60c5667957
* fix(TPC): Return default codec if the local sdp is not available. Get the correct media type when generating the source identifier.
88560a8a5e...9eb4af1e80
* feat(prosody-modules): Moves a function for getting room to util.
* feat: Audio/Video moderation.
* squash: Fix docs.
* squash: Changes a field name in the message for adding jid to whitelist.
* squash: Moves to boolean from boolean string.
* squash: Only moderators get whitelist on join.
* squash: Check whether in room and moderator.
* squash: Send to participants only message about approval.
Skips sending the whole list.
* feat: Separates enable/disable by media type.
Adds actor to the messages to inform who enabled it.
* squash: Fixes reporting disable of the feature.
* squash: Fixes init of av_moderation_actors.
* squash: Fixes av_moderation_actor jid to be room jid.
* squash: Fixes comments.
* squash: Fixes warning about shadowing definition.
* squash: Updates ljm.
* fix: Fixes auto-granting from jicofo.
* squash: Further simplify...
* fix(JingleSession): Move the ssrc identifier generation to LocalSdpMunger.
* fix(logger): Logging enhancements. Get rid of noisy logs related to SDP transformations which are redundant. Fix formatting and add missing information.
7cbd9c8f2a...923aa449c4
* fix(quality-control): Propagate the height constraints to p2p session. If the application is using the new receiver constraints, propagate the height constraint to the p2p session as well.
* build(deps): bump lodash from 4.17.19 to 4.17.21
* chore(deps): bump hosted-git-info from 2.8.8 to 2.8.9
74a90f7035...7cbd9c8f2a
* fix(quality-control): fix constraints sent on channel initialization. Do not send old format constraints if no constraints are set before the channel is initialized.
* chore(deps) run npm audit fix
* chore(deps) update webrtc-adater@8.0.0
86c7a35817...74a90f7035
* Add dependency for promise.allSettled. Older chrome versions like M72 do not support Promise.allSettled.
* fix(conference): Enable p2p for unified plan clients.
* fix(TPC): Use addTrack instead of addStream in Unified-plan impl.
* Add missing spaces in debug logs.
ad5692d6aa...e362c89eb6
* fix(SDP): Move all SDP related files to a different dir. SDP utility classes are spread across RTC and XMPP directories now, moving these class files to a 'sdp' directory.
* fix(stats): Return promise for getStats. 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.
* Fix: sysMessageHandler not deleted (#1590)
* task(e2ee): switch back to GCM
463e213b3f...7667117117
* fix(quality-control): Send the new constraint on join. Fixes the case where the old format height constraint is sent on join for a jvb media session.
7dedb59b9c...463e213b3f
* fix(quality-control): Switch to new receiver constraints by default. Use the new receiver constraints unless it is explicitly disabled through config.js.
3c9913ed61...7dedb59b9c
Jaya Allamsetty
Avram Tudor
Дамян Минков
tmoldovan8x8
Saúl Ibarra Corretgé
Hristo Terezov
George Politis
Jaya Allamsetty
Vlad Piersec
Andrei Gavrilescu
Gabriel Borlea
Calin Chitu
Pavol Cvengros
Robert Pintilii
dependabot[bot]
hmuresan
Christoph Settgast
Steffen Kolmer
Tudor-Ovidiu Avram