livekit_client

What's Changed

• chore: introduce DevelopmentTokenSource and deprecate SandboxTokenSource by @hiroshihorie in #1149
• fix: Room.getSid() never resolves when the JoinResponse carries an empty room sid by @bkali in #1152
• Migrate enum conversions from map lookups to switch expressions by @hiroshihorie in #1156
• Default video degradation preference by track source, including the backup codec by @xianshijing-lk in #1155
• chore: update generated protocol definitions to v1.50.4 by @hiroshihorie in #1161
• Surface newer server disconnect reasons as distinct DisconnectReason values by @hiroshihorie in #1158
• Clear simulcast codec state on unpublish and full reconnect by @hiroshihorie in #1159
• Batch dependency bumps (supersedes open dependabot PRs) by @hiroshihorie in #1163
• Ensure roomOptions in Room.connect(...) are using the locally defined variable, NOT implicitly this.roomOptions by @1egoman in #1169
• Modernize the Flutter example app by @hiroshihorie in #1066

New Contributors

• @bkali made their first contribution in #1152

Full Changelog: v2.10.0...v2.11.0

• Added: New DisconnectReason members for newer server disconnect reasons, previously reported as unknown
• Added: Add DevelopmentTokenSource, the new name for the now-deprecated SandboxTokenSource
• Changed: Default video degradation preference is now based on the track source (camera maintains framerate, screen share maintains resolution, others balanced) and is applied to the backup codec's sender as well
• Changed: Update generated protocol definitions to v1.50.4
• Changed: Unrecognized protobuf enum values from newer servers now fall back to safe defaults, for example a new DisconnectReason maps to unknown
• Changed: Update dependencies: device_info_plus to 13.x, connectivity_plus 7.3.1, dart_webrtc 1.8.1, synchronized 3.4.1, uuid 4.6.0
• Changed: Modernize the example app with a redesigned connect page, connection history, room header, focus layout, and an in-room messages panel
• Fixed: Room.connect no longer ignores the roomOptions argument passed to it
• Fixed: Room.getSid() now resolves when the room sid is issued after the join response
• Fixed: Backup codec state is cleared on unpublish and full reconnect, so republishing no longer acts on senders from a torn down connection
• Fixed: DegradationPreference.disabled now maps to maintainFramerateAndResolution, as WebRTC defines it

What's Changed

• Add RPC tester interface to example app by @1egoman in #1085
• fix: don't set zero size if size is unknown by @holzgeist in #1141
• fix: compile errors on Dart 3.13 from stricter flow analysis by @hiroshihorie in #1146
• feat: Swift Package Manager support for iOS and macOS by @hiroshihorie in #1142
• feat: typed screen share helpers by @hiroshihorie in #1147
• Fix publish workflow failing on the flutter_webrtc pin warning by @hiroshihorie in #1140

Full Changelog: v2.9.0...v2.10.0

• Added: Swift Package Manager support for iOS and macOS. CocoaPods remains fully supported.
• Added: Add ScreenSelectDialog.show and Hardware.requestCapturePermission so apps can start screen share without importing flutter_webrtc
• Fixed: Omit data stream totalLength when size is unknown
• Fixed: Fix compile errors on Dart 3.13 where nullable publish options are no longer promoted across await
• Fixed: Harden the example RPC tester

Added: Add native certificate pinning for SDK-owned connections
Added: Add RPC v2 (data-stream-based payloads) with v1 fallback
Added: Audio engine availability and externalCallSystem mode for CallKit
Added: AudioManager audio session options with engine-driven native lifecycle and platform routing controls
Added: Add ConnectionCheck utility for diagnosing connection issues (port of the client-sdk-js connection helper)
Added: Audio frame capture on Linux/Windows
Added: Microphone mute mode control on iOS/macOS
Added: Runtime audio processing controls for local audio tracks
Added: Widget Placeholder added for VideoTrackRenderer
Changed: Default VideoTrackRenderer renderMode to auto, which currently resolves to texture rendering
Fixed: Apply Android media audio attributes during WebRTC initialization
Fixed: Use initialization audio options as the default Android session policy
Fixed: Avoid sticky Android speaker routing when updating route preference
Fixed: Clean up local audio tracks when recording start fails
Fixed: Throw platformUnavailable when runtime audio processing is unsupported
Fixed: Apply create-time audio processing when local recording is prepared
Fixed: Remove non-public buttonPressed: selector from broadcast picker activation
Fixed: Allow selectAudioOutput on Android
Fixed: Support platform video rendering on macOS
Fixed: Emit a single disconnected event when connecting fails
Fixed: Handle switching video render modes without stale renderers
Docs: Add AGENTS.md with agent/contributor guidelines

• Added: Add native certificate pinning for SDK-owned connections
• Added: Add RPC v2 (data-stream-based payloads) with v1 fallback
• Added: Audio engine availability and externalCallSystem mode for CallKit
• Added: AudioManager audio session options with engine-driven native lifecycle and platform routing controls
• Added: Add ConnectionCheck utility for diagnosing connection issues (port of the client-sdk-js connection helper)
• Added: Audio frame capture on Linux/Windows
• Added: Microphone mute mode control on iOS/macOS
• Added: Runtime audio processing controls for local audio tracks
• Added: Widget Placeholder added for VideoTrackRenderer
• Changed: Default VideoTrackRenderer renderMode to auto, which currently resolves to texture rendering
• Fixed: Apply Android media audio attributes during WebRTC initialization
• Fixed: Use initialization audio options as the default Android session policy
• Fixed: Avoid sticky Android speaker routing when updating route preference
• Fixed: Clean up local audio tracks when recording start fails
• Fixed: Throw platformUnavailable when runtime audio processing is unsupported
• Fixed: Apply create-time audio processing when local recording is prepared
• Fixed: Remove non-public buttonPressed: selector from broadcast picker activation
• Fixed: Allow selectAudioOutput on Android
• Fixed: Support platform video rendering on macOS
• Fixed: Emit a single disconnected event when connecting fails
• Fixed: Handle switching video render modes without stale renderers
• Docs: Add AGENTS.md with agent/contributor guidelines

chore: use pub doc directory layout

Added: Add agent deployment targeting to token source options
Fixed: Android plugin compatibility with AGP 9 built-in Kotlin
Fixed: Use maintain-resolution as the default video degradation preference for local video publishing

Full Changelog: v2.8.0...v2.8.1

• Added: Add agent deployment targeting to token source options
• Fixed: Android plugin compatibility with AGP 9 built-in Kotlin
• Fixed: Use maintain-resolution as the default video degradation preference for local video publishing

Added: Session API support for simpler E2EE setup
Changed: Manual video quality selection can be used with adaptive stream enabled
Changed: Generated protocol definitions for LiveKit protocol v1.45.8
Fixed: waitForBufferStatusLow busy-waiting after engine close
Fixed: Simulcast lower layers exceeding the top layer
Fixed: forceRelay log message interpolation
Fixed: sendSyncState error handling so sync-state preparation failures are not swallowed
Fixed: Screen share simulcast default low layer alignment
Fixed: Region failover null-provider dereference
Fixed: Android builds with dependencies that require compileSdk 36
Fixed: Deferred track listener leaks across reconnects
Fixed: Adaptive stream dimensions on high-density displays
Fixed: Session.start() reentrancy during concurrent calls
Fixed: Connected server address resolving from the wrong peer connection
Fixed: Reconnect counter null assertion on the first reconnect attempt
Fixed: Premature publication disposal during unpublish

• Added: Session API support for simpler E2EE setup
• Changed: Manual video quality selection can be used with adaptive stream enabled
• Changed: Generated protocol definitions for LiveKit protocol v1.45.8
• Fixed: waitForBufferStatusLow busy-waiting after engine close
• Fixed: Simulcast lower layers exceeding the top layer
• Fixed: forceRelay log message interpolation
• Fixed: sendSyncState error handling so sync-state preparation failures are not swallowed
• Fixed: Screen share simulcast default low layer alignment
• Fixed: Region failover null-provider dereference
• Fixed: Android builds with dependencies that require compileSdk 36
• Fixed: Deferred track listener leaks across reconnects
• Fixed: Adaptive stream dimensions on high-density displays
• Fixed: Session.start() reentrancy during concurrent calls
• Fixed: Connected server address resolving from the wrong peer connection
• Fixed: Reconnect counter null assertion on the first reconnect attempt
• Fixed: Premature publication disposal during unpublish

What's Changed

• Add audio renderer API by @hiroshihorie in #1004
• Fix missing offer in syncState by @hiroshihorie in #1028
• Fix signal resume events by @hiroshihorie in #1027
• chore: bump flutter-webrtc and upgrade to libwebrtc m144. by @cloudwebrtc in #1029
• Bump android m144 by @hiroshihorie in #1030
• Processor fixes 1 by @hiroshihorie in #1038
• Retry jitter by @hiroshihorie in #1036
• Ping RTT by @hiroshihorie in #1035
• Fix dynacast by @hiroshihorie in #1033
• Set video dimensions API by @hiroshihorie in #1037

Full Changelog: v2.6.5...v2.7.0

• Added: Add setVideoDimensions for remote track publications
• Added: Add audio renderer API for receiving raw audio frames
• Changed: Bump flutter-webrtc to 1.4.0 and upgrade to libwebrtc m144
• Fixed: Add signal RTT tracking via pingReq/pongResp
• Fixed: Add jitter to reconnect retry delays
• Fixed: Fix missing params in CameraCaptureOptions.copyWith
• Fixed: Fix missing RoomReconnectedEvent and incorrect SyncState timing during signal-only reconnection
• Fixed: Fix VP9/SVC dynacast layer handling
• Fixed: Stop processor on track end
• Fixed: Include last server offer in SyncState during reconnection

What's Changed

• Protocol v16 support by @hiroshihorie in #1003
• Fix pub score static analysis by @hiroshihorie in #1010
• chore: Update dependency constraints to latest compatible versions by @hiroshihorie in #1011
• Fix audio visualizer exception by @hiroshihorie in #1008
• Fix duplicate track publish by @hiroshihorie in #1009
• Fix: use correct ProcessorOptions type based on track kind by @hiroshihorie in #1014
• Fix: preserve audio category options when forcing speaker output by @hiroshihorie in #1015
• [🤖 readme-manager] Update README by @ladvoc in #1017
• chore: Add ReconnectReason for reconnect. by @cloudwebrtc in #1020
• Fix microphone stopping after pre-connect audio by @hiroshihorie in #1021

Full Changelog: v2.6.4...v2.6.5

• Added: Send reconnect reason to server during WebSocket reconnection
• Fixed: setProcessor() now uses VideoProcessorOptions for video tracks instead of AudioProcessorOptions
• Fixed: Fix wildcard variable lint warnings to improve pub.dev score
• Fixed: Preserve Bluetooth and AirPlay options when forcing speaker output on iOS
• Fixed: MissingPluginException when stopping audio visualizer on Android/iOS
• Fixed: Fix microphone stopping after pre-connect audio completes
• Fixed: Duplicate tracks published when setCameraEnabled called rapidly

What's Changed

• Fix setVideoInputDevice by @hiroshihorie in #973
• Fix duplicate leave by @hiroshihorie in #999
• Optimize audio renderer payloads by @hiroshihorie in #988
• Fix native audio renderer cleanup and simplify Android path by @hiroshihorie in #1001
• Add web support for pre-connect audio buffer by @hiroshihorie in #1000
• Fix web audio capture on Safari by @hiroshihorie in #1006
• Fix web pre-connect audio buffer sample rate by @hiroshihorie in #1007

Full Changelog: v2.6.3...v2.6.4

• Added: Add web support for pre-connect audio buffer
• Fixed: Remove duplicate sendLeave
• Fixed: Fix setVideoInputDevice not switching camera
• Fixed: Web pre-connect audio buffer using wrong sample rate
• Fixed: Fix iOS audio renderer resource leak and remove Android 32-bit dead code
• Fixed: Fix web audio capture not starting on Safari due to suspended AudioContext
• Performance: Send raw PCM bytes in audio renderer instead of boxed int arrays

What's Changed

• fix: debounce for ensurePublisherConnected. by @cloudwebrtc in #967
• chore: Bump protobuf to v1.44.0 by @hiroshihorie in #980
• Fix meta package import errors by @hiroshihorie in #981
• Fix region provider related bugs by @hiroshihorie in #983
• chore: fix docs and web example workflows by @hiroshihorie in #985
• Fix concurrent modification by @hiroshihorie in #986
• Prevent crashes from writing to closed data stream controllers by @hiroshihorie in #989
• Fix publisher pc causing redundant renegotiations by @hiroshihorie in #990

Full Changelog: v2.6.2...v2.6.3

• Fixed: Fix publisher connection causing redundant renegotiations on lower-end devices
• Fixed: Use explicit show clauses for foundation imports
• Fixed: Fix concurrent modification on collection iteration during async operations
• Fixed: Fix crash when writing to closed data stream controllers
• Fixed: Fix region URL provider cache using wrong time unit and wrong field check for regions in leave event
• Chore: Update protobuf to v1.44.0

• Added: Bitrate priority control APIs
• Fixed: Windows: stop camera after unpublishing video track
• Fixed: Keep mute state in sync with server
• Fixed: Fix example prejoin device dropdown handling
• Chore: Mark LocalVideoTrack constructor as internal for consistency with LocalAudioTrack
• Chore: Update protobuf dependency to v6.0.0
• Chore: Skip iOS platform download in CI

• Fixed: Enable E2EE for VP9 codec
• Fixed: Correct spelling in E2EE log and error messages
• Fixed: Ensure all dispose functions are executed even if one throws
• Fixed: Fix microphone label typo in example app prejoin audio settings
• Docs: Fix typos and inconsistencies in README and CHANGELOG

• Session API
• Allow customizing screen share dialog labels for localization

• Added pre-connect audio buffering
• Added token source API with caching and endpoint helpers
• Enabled software AEC/NS/AGC for Linux
• Added event logging with public/internal tagging
• Migrated manual JSON serialization to json_serializable code generation
• Re-enabled Flutter lint warnings and fixed existing issues
• Improved reliable data channel buffering, sequencing, and added integration tests
• Fixed race condition with pending tracks

• Feat: Data Packet Cryptor (#873)
• Feat: Add E2EE support for H265 (#864)
• Fix: fix async call for update participant info (#897)
• Move the accessToken to header (#891)
• Bumpup and clamp connection timeout, fix the comment (#893)
• Fix events emit order (#902)

• Fix missing properties for TextStreamInfo (#881)
• Enum for TextStreamOperationType (#882)
• Fix stream progress (#884)

• Fix Publication's track property initialization (#867)
• Fix Missing attributes in data stream (#865)
• Fix Internal WebRTC pod version conflict (#876)

• Bump flutter-webrtc to 1.0.0.
• Upgrade libwebrtc to m137.7151
• fix: E2EE bug for Chrome. (#832)
• fix(web, firefox): backup old values before overwriting them (#819)
• fix: memory leak visualizer stop android (#831)
• fix: ensure engine always has correct device id if switching camera fails (#820)
• fix: Always emit RoomDisconnectedEvent when the reason is clientInitiated. (#821)
• fix: sif frame enqueuing with e2ee (#822)
• fix: update camera deviceId when track is null (#814)
• feat: Audio Visualizer for Windows/Linux. (#739)

• fix: Switch to livekit fork of noise for 16KB page support (#839)
• hotfix: bump libwebrtc to m137.7151.01 for android (#837)
• fix: tighten VideoTrackRenderer Widget (#695)

• hotfix: bump libwebrtc to m137.7151.03 to support H.265 (#837)
• feat: Expose ParticipantState (#848)
• chore: Update protobuf to v1.39.2 (#847)

• fix: Fix @internal conflicting imports from package meta or flutter/foundation.

• Added: Attribute typings for agent and transcription (#811)
• Changed: Enum for VideoView.fit (#806)
• Chore: Update protobuf protocol to v1.39.2 (#812)

• fix: fix room.disconnect when pending reconnect. (#802)
• fix: fix bitrate display for remote video track. (#801)
• feat: add a flag to skip auto native audio config (#797)
• fix: sif detection for e2ee (#788)
• chore: Update protobuf dep (#790)

• fix: interop for encode and decode function in e2ee cryptor (#785)
• chore: Unorder the lossy data channel (#783)
• chore: bump flutter-webrtc to 0.14.1.
• fix: mitigate reconnect loop (#780)

• feat: Add smoothTransition option for AudioVisualizer. (#773)
• fix: Fix deadlock when creating a frame cryptor on iOS/macOS.
• fix: add task runner for linux to fix crashes.

• feat: noise filter for flutter web. (#762)

• feat: data stream
• fix: calling setSourceEnabled should not affect the current output settings (#754)
• fix: Calling connect on a disconnected room instance should reset _isClosed. (#752)
• fix: Fix for Chrome getUserMedia "ideal" Constraint Change.
• fix: fix duplicate fmtp for vp9 on some old Android devices. (#758)

• feat: Automatically configure audio mode for Android. (#746)

• fix: Fixed the echo issue on some specific models of Android devices.
• chore: Bbmp version for flutter-webrtc.
• fix: Improve browser version detection, fix issue #730 (#738)
• feat: Fast track publication. (#720)
• chore: Remove bypassVoiceProcessing = true settings for connect page. (#693)
• fix: [bug] Crash when join room on Windows.
• fix: Disable selectAudioInput for mobile platforms.

• fix: fixed bug for fast track publish.

• fix: Fix creation timing for local participant. (#749)
• fix: Filter out non-video codecs for fast video track publication. (#747)

• feat: VideoRenderer support cache renderer. (#723)
• feat: Visualizer for web. (#718)
• fix: Fix bar Visualizer overflow for Android. (#717)
• fix: disconnect reason inconsistent. (#715)
• fix: remove stats video data print from logs (#716)

• feat: RPC. (#682)
• fix: Properly handle broadcast capture state (#705)

• Emit a final empty AudioVisualizerEvent after track stops (#701)
• fix: Improve speaker switching logic for iOS. (#692)

• feat: add TrackProcessor support. (#657)
• fix: bug for mute/unmute and speaker switch. (#684)

• fix bypass voice processing not work. (#668)
• fix track.stop for remote track. (#669)
• fix Visualizer block UI. (#667)

• fix: bug for speakerPhone switch. (#672)

• fix: side effects for stop remote track.

• Support FocusMode/ExposureMode for camera capture options. (#658)
• Fix Swift compilation errors (#662)
• Improve reconnect logic. (#654)
• Fixed bug for Screen Share on iOS/Android.

• Add support for audio visualizer.

• Fix version mismatch for CocoaPods (#648)
• Android AGP upgrade (#649)

• fix: Fixed speaker cannot be switched on iOS. (#617)
• feat: Increase default audio bitrate. (#616)

• Fix TrackStreamStateUpdatedEvent not emitted. (#612)

• feat: support bypass voice processing. (#595)
• fix: Dispose LocalParticipant when room.disconnect. (#609)
• Fix spelling error in method name: _checkPremissions to_checkPermissions (#605)
• fix: fix speaker switching behavior for android. (#604)
• fix: Optimize stats display (#602)
• fix: Handle disconnect reasons correctly. (#594)

• fix: android build failure. (#587)
• Update README.md for Android Screenshare (#583)

• upgrade flutter-webrtc to 0.11.7
• upgrade web to 1.0.0
• feat: Add timestamps to TranscriptionSegment. (#577)
• feat: Prepare connection/region pinning. (#574)

• fix bug for PlatformView on iOS (#570)
• Deprecated connectOptions in Room constructor.
• Deprecated roomOptions int room.connect.
• Added screenShareEncoding for VideoPublishOptions.
• Change CameraCaptureOptions.params.encoding to optional. It is recommended to use VideoPublishOptions.videoEncoding/screenShareEncoding to control the video sending bitrate.

• add PlatformView for iOS (#509)
• update lints to 4.0.0. (#563)

• feat: expose DegradationPreference for publish options. (#556)
• fix room.disconnect issue. (#559)
• fix: use getSettings() to get actual dimensions for mobile (#560)
• Add support for participant attributes (#558)

• fix: fix crash for windows
• feat: bump version for flutter-webrtc.
• fix: ratchet on a single frame until ratchetWindowSize (#544)
• fix: fix mediaStreamTrack.getSettings() on Flutter Web.

• feat: add Transcription Event. (#531)
• feat: Expose Participant.Kind. (#532)
• fix: ignore unable to parse frames completely (#530)

• Framecryptor decrypting fixes (#520)
• feat: add voiceIsolation support (#523)
• fix: audio session control for ios.

fix: audio devcie ids consistency (#513) feat: provide option to skip stop/restartTrack for mute/unmute (#516)

• bump version of flutter-webrtc with privacy manifest files

• upgrade to connectivity plus 6.x.
• bump dart_webrtc to 1.4.3.

• fix: Expose keyRingSize/discardFrameWhenCryptorNotReady parameters for KeyProvider

• update to package:web (#484)
• feat: add keyRingSize/discardFrameWhenCryptorNotReady to KeyProviderOptions. (#493)

• Fix: bug for sync state (#491)

• Read capabilities from RtcRTPReceiver instead of from sender. (#488)
• Update screenshare logic for svc publishing (#487)
• Prevent screen-sharing on flutter web mobile. (#486)
• feat: add setKey variant, setRawKey (#482)