* 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/...ae70962bfaa1e6c91dc7a8ecdff983740b99b874
feat(lang) add helper script for manual translations
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 (#10043)
* 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/...0646bc3403807dbf1370c88f028d9e0a16bcab1a
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/...3b8baa9d3be2839510abaa954357d0b0ab023649
* 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/...609e3d5a1aac600cb3ea112c0ec83ebedd43166d
* 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/...28a53553569dbb4adcba39cd997ff121b4d05c62
* 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/...b0d27fa8daef615d45fe566a0385f66facfcf025
* 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/...6eaffc4b11b42d851a17ed941a19a9513265d3bb
fix(av-moderation): Fixes approving and dismissing the notification.
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/...6a3df11ffa7a2204b579326e23cdaa85a79521d1
* 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/...9e632a77c504fa993717017cfc949e8883d07a56
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/...b815157a22cec219a26457143b6d6cb2f430e01b
* 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/...053a26604da568188a1f99a45a275d1d04e7b8d9
* 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/...3a313a244daf3d54b8ea95e9012b2ae01598b113
* 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/...89a7e2d9cdd24f48f95f6668ae4d8db1b635cf36
* 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/...f974007ca6e27592625b83e38d22c6756ac286cb
* 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/...b43a9fa0eea73ba48d805fba97d0cd3fb9d8d07a
* 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/...229015a6f307b17f939c8db8549d8543cdc3f561
Дамян Минков
Saúl Ibarra Corretgé
Hristo Terezov
Jaya Allamsetty
Andrei Gavrilescu
robertpin
Jaya Allamsetty
hmuresan
Christoph Settgast
Avram Tudor
Calin Chitu