# Changelog
All notable changes to this project will be documented in this file.
The format is based on [Keep a Changelog](https://keepachangelog.com/en/1.0.0/),
and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0.html).
## [Unreleased]
## [0.7.32](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.31...rust-sdks/livekit@0.7.32) - 2026-02-16
### Fixed
- fix full_reconnect downgrade & don't ignore Leave messages ([#893](https://github.com/livekit/rust-sdks/pull/893))
### Other
- turn single peerconnection off by default ([#897](https://github.com/livekit/rust-sdks/pull/897))
- ensure signal connections times out properly and retries ([#895](https://github.com/livekit/rust-sdks/pull/895))
- added Single Peer Connection support to Rust ([#888](https://github.com/livekit/rust-sdks/pull/888))
- set the simulcast codec & layers ([#891](https://github.com/livekit/rust-sdks/pull/891))
## [0.7.31](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.30...rust-sdks/livekit@0.7.31) - 2026-02-10
### Other
- don't use clamp as the ultimate_kbps can be lower than 300 ([#886](https://github.com/livekit/rust-sdks/pull/886))
- pre-connect the publisher PC when an RPC handler is registered ([#880](https://github.com/livekit/rust-sdks/pull/880))
## [0.7.30](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.29...rust-sdks/livekit@0.7.30) - 2026-02-09
### Fixed
- fix the RPC race condition ([#865](https://github.com/livekit/rust-sdks/pull/865))
### Other
- update proto & fix CI ([#871](https://github.com/livekit/rust-sdks/pull/871))
- Use workspace dependencies & settings ([#856](https://github.com/livekit/rust-sdks/pull/856))
- Upgrade protocol to v1.44.0 ([#857](https://github.com/livekit/rust-sdks/pull/857))
- Expose participant's permission to ffi layer ([#824](https://github.com/livekit/rust-sdks/pull/824))
## [0.7.29](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.28...rust-sdks/livekit@0.7.29) - 2026-01-15
### Fixed
- ensure Room.creation_time is ms ([#822](https://github.com/livekit/rust-sdks/pull/822))
### Other
- try setting x-google-start-bitrate for vp9 ([#820](https://github.com/livekit/rust-sdks/pull/820))
## [0.7.28](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.27...rust-sdks/livekit@0.7.28) - 2025-12-19
### Added
- *(ParticipantInfo)* export kind details ([#813](https://github.com/livekit/rust-sdks/pull/813))
## [0.7.27](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.26...rust-sdks/livekit@0.7.27) - 2025-12-17
### Other
- Handle server initiated mute request ([#812](https://github.com/livekit/rust-sdks/pull/812))
## [0.7.26](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.25...rust-sdks/livekit@0.7.26) - 2025-12-04
### Added
- *(connector)* initial service impl ([#790](https://github.com/livekit/rust-sdks/pull/790))
### Fixed
- fix mute/unmute events for LocalTrack. ([#799](https://github.com/livekit/rust-sdks/pull/799))
### Other
- Add RoomEvent::TokenRefreshed ([#803](https://github.com/livekit/rust-sdks/pull/803))
## [0.7.25](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.24...rust-sdks/livekit@0.7.25) - 2025-11-20
### Other
- perform full reconnect if resume fails ([#792](https://github.com/livekit/rust-sdks/pull/792))
- E2E RPC tests ([#769](https://github.com/livekit/rust-sdks/pull/769))
- Remove unused dependencies ([#761](https://github.com/livekit/rust-sdks/pull/761))
## [0.7.24](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.23...rust-sdks/livekit@0.7.24) - 2025-10-27
### Other
- Expose set video quality ([#759](https://github.com/livekit/rust-sdks/pull/759))
## [0.7.23](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.22...rust-sdks/livekit@0.7.23) - 2025-10-23
### Other
- add h265 codec support ([#762](https://github.com/livekit/rust-sdks/pull/762))
## [0.7.22](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.21...rust-sdks/livekit@0.7.22) - 2025-10-22
### Other
- License check ([#746](https://github.com/livekit/rust-sdks/pull/746))
- Remove participant check for data packets ([#757](https://github.com/livekit/rust-sdks/pull/757))
- clamp connection timeout and fixed the comment ([#748](https://github.com/livekit/rust-sdks/pull/748))
- put examples in root Cargo workspace ([#731](https://github.com/livekit/rust-sdks/pull/731))
## [0.7.21](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.20...rust-sdks/livekit@0.7.21) - 2025-10-13
### Added
- *(e2ee)* add data channel encryption ([#708](https://github.com/livekit/rust-sdks/pull/708))
### Fixed
- fix some potential audio issues, clean up the code a bit, and suppress some warnings ([#737](https://github.com/livekit/rust-sdks/pull/737))
- do not log 'signal client closed: "stream closed"' on disconnect ([#727](https://github.com/livekit/rust-sdks/pull/727))
### Other
- Test participant disconnect ([#732](https://github.com/livekit/rust-sdks/pull/732))
- Increase RPC max RT time to 7s ([#729](https://github.com/livekit/rust-sdks/pull/729))
- E2E audio test ([#724](https://github.com/livekit/rust-sdks/pull/724))
## [0.7.20](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.19...rust-sdks/livekit@0.7.20) - 2025-10-03
### Other
- updated the following local packages: libwebrtc
## [0.7.19](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.18...rust-sdks/livekit@0.7.19) - 2025-09-29
### Fixed
- apply original participant fields in data messages ([#709](https://github.com/livekit/rust-sdks/pull/709))
### Other
- Implement Display and Error for RpcError ([#719](https://github.com/livekit/rust-sdks/pull/719))
- Fix intermittently failing E2E reliability test ([#718](https://github.com/livekit/rust-sdks/pull/718))
- Do not modify raw packets ([#714](https://github.com/livekit/rust-sdks/pull/714))
- Add send_bytes method ([#691](https://github.com/livekit/rust-sdks/pull/691))
- Disable opus red for e2ee enabled clients ([#706](https://github.com/livekit/rust-sdks/pull/706))
- Upgrade protocol to v1.41.0 ([#703](https://github.com/livekit/rust-sdks/pull/703))
## [0.7.18](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.17...rust-sdks/livekit@0.7.18) - 2025-09-09
### Other
- Add option to enable audio preconnect buffer in SDK & FFI ([#700](https://github.com/livekit/rust-sdks/pull/700))
- Data channel reliability ([#688](https://github.com/livekit/rust-sdks/pull/688))
## [0.7.17](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.16...rust-sdks/livekit@0.7.17) - 2025-09-03
### Other
- updated the following local packages: livekit-api, libwebrtc
## [0.7.16](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.15...rust-sdks/livekit@0.7.16) - 2025-07-31
### Other
- updated the following local packages: livekit-api
# Changelog
## [0.7.15](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.14...rust-sdks/livekit@0.7.15) - 2025-07-16
### Other
- remove published tracks when the room is closed ([#677](https://github.com/livekit/rust-sdks/pull/677))
## [0.7.14](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.13...rust-sdks/livekit@0.7.14) - 2025-06-23
### Fixed
- `audio_frame_ms` didn't work expectedly ([#671](https://github.com/livekit/rust-sdks/pull/671))
## [0.7.13](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.12...rust-sdks/livekit@0.7.13) - 2025-06-17
### Other
- Expose room updates, support MoveParticipant (protocol 15) ([#662](https://github.com/livekit/rust-sdks/pull/662))
## [0.7.12](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.11...rust-sdks/livekit@0.7.12) - 2025-06-11
### Fixed
- fix duration overflow ([#654](https://github.com/livekit/rust-sdks/pull/654))
### Other
- Remove debouncer when fast_publish is enabled ([#649](https://github.com/livekit/rust-sdks/pull/649))
## [0.7.9] - 2025-04-08
### Added
- High-level data streams API
### Deprecated
- Low-level data stream packet events
## [0.7.8] - 2025-03-21
### Fixed
- Revert the RPC change, need more robust way
## [0.7.7] - 2025-03-18
### Fixed
- Move RPC handlers to room
## [0.7.6] - 2025-02-28
### Added
- Add audio filter support
## [0.7.5] - 2025-02-06
### Fixed
- Fix a dependency issue with an older version of the libwebrtc crate
## [0.7.4] - 2025-02-03
### Added
- Support for track subscription permissions
## [0.7.3] - 2025-01-17
### Added
- Add an API to set buffer_amount_low_threshold for DataChannel
- Update RoomInfo to contain buffer_amount_low_threshold for DataChannel
### Fixed
- Wait for the buffered amount to become low before sending data during publish_data for Reliable Data Channel
## [0.7.2] - 2025-01-04
### Added
- bump
### Fixed
- Fixed deadlock with nested RPC calls
## [0.7.1] - 2024-12-14
### Added
- bump libwebrtc to m125
## 0.7.42 (2026-05-21)
### Features
- Introduce pipeline options for remote data tracks, support multiple in-flight frames.
### Fixes
- Filter internal data streams out of livekit-ffi interface - #1112 (@1egoman)
#### feat: add Android application context initialization for PlatformAudio support.
Android requires `ContextUtils.initialize(applicationContext)` before WebRTC audio components can be created. This change:
- Adds `livekit_ffi_initialize_android_context()` C FFI function for Unity and other FFI consumers
- Uses `CreateAndroidAudioDeviceModule()` instead of generic `CreateAudioDeviceModule()` on Android
- Handles empty device GUIDs on Android (falls back to index 0)
- Documents Android-specific limitations: single default device, no app-level device selection
Platform notes:
- Android device enumeration returns only one "default" device with empty name/GUID
- Audio routing (speaker/earpiece/Bluetooth) is controlled by Android's AudioManager, not WebRTC
## 0.7.41 (2026-05-20)
### Fixes
- Bugfix: Always emit Disconnected on engine close - #1096 (@MaxHeimbrock)
- Support for large RPC messages using data streams - #1013 (@1egoman)
## 0.7.40 (2026-05-14)
### Fixes
- feat: add scalability mode for AV1/VP9. - #1076 (@cloudwebrtc)
- Add `LIVEKIT_PREFERRED_HW_ENCODER` to prefer `nvenc` or `vaapi` hardware video encoding when both are available.
- Relocate unrelated types out of `livekit-protocol`
#### Get WebRTC ADM into Rust - #1037 (@xianshijing-lk)
This PR introduces platform audio device management via WebRTC's Audio Device Module (ADM).
#### Features
- **ADM Proxy**: New `AdmProxy` class that switches between Dummy ADM (synthetic mode) and Platform ADM (real audio I/O)
- **PlatformAudio API**: High-level Rust API for microphone capture and speaker playout with AEC/AGC/NS
- **Device enumeration**: List and select recording/playout devices by index or GUID
- **Mode switching**: Seamlessly switch between synthetic mode (FFI callbacks) and platform mode (native speakers) while audio is active
- **FFI platform audio support**: Expose platform audio device enumeration and selection through `livekit-ffi`
- **Audio processing**: Configure echo cancellation, noise suppression, and auto gain control with platform-specific defaults (hardware on iOS, software elsewhere)
#### Audio Modes
| Synthetic | NativeAudioSource | Dummy ADM + FFI | Unity audio, agents |
| Platform | Platform ADM mic | Platform ADM speakers | VoIP with AEC |
#### API
```rust
// Create PlatformAudio for microphone/speaker access
let audio = PlatformAudio::new()?;
// Enumerate and select devices
for i in 0..audio.recording_devices() as u16 {
println!("Mic {}: {}", i, audio.recording_device_name(i));
}
audio.set_recording_device(0)?;
// Create audio track for publishing
let track = LocalAudioTrack::create_audio_track("mic", audio.rtc_source());
```
## 0.7.39 (2026-05-11)
### Fixes
- fix: Sync inner.enabled state for E2EE manager. - #1073 (@cloudwebrtc)
- Upgrade protocol to v1.45.8
## 0.7.38 (2026-05-10)
### Fixes
- Bump `rustls-webpki` to 0.103.13, addressing [GHSA-82j2-j2ch-gfr8](https://github.com/advisories/GHSA-82j2-j2ch-gfr8)
- Fix missing `libwebrtc.jar` for Android builds, harden build scripts
- fix: derive `simulcasted` from non-deprecated TrackInfo fields - #1052 (@cloudwebrtc)
- fix race in download_webrtc to reduce flaky build - #1047 (@hechen-eng)
- Improve WebRTC build scripts and add external_audio_source patch - #1053 (@xianshijing-lk)
- support SimulateScenario through FFI to improve testing - #1069 (@davidzhao)
- TEL-464: reduce redundant resampling in audio filter - #1019 (@hechen-eng)
## 0.7.37 (2026-04-23)
### Features
#### Add support for frame level packet trailer
##890 by @chenosaurus
- Add support to attach/parse frame level timestamps & frame ID to VideoTracks as a custom payload trailer.
- Breaking change in VideoFrame API, must include `frame_metadata` or use VideoFrame::new().
### Fixes
- Add device-info crate and send device_info to telemetry - #982 (@maxheimbrock)
- Fix data track packet format issue breaking E2EE
- Fix unbound send queue that can cause latency in data track messages - #1032 (@chenosaurus)
- Fix for raw stream drop called from non tokio thread like Unity .NET GC - #1016 (@MaxHeimbrock)
## 0.7.36 (2026-04-02)
### Features
- Initial support for data tracks
### Fixes
#### use the bounded buffer for video stream
##956 by @xianshijing-lk
Before this PR, it uses an unbounded buffer for video stream, that will cause multiple problems:
1, video will be lagged behind if rendering is slow or just wake up from background
2, it will be out of sync with audio
This PRs provides options to set a bounded buffer for video stream, and use 1 buffer as the default option.
## 0.7.35 (2026-03-31)
### Features
- Expose participant active event, state, and joined at
### Fixes
- Upgrade to thiserror 2
## 0.7.34 (2026-03-22)
### Features
#### E2EE: allow setting key_ring_size and key_derivation_algorithm, update webrtc to m144
##921 by @onestacked
This PR uses [this webrtc-sdk PR](https://github.com/webrtc-sdk/webrtc/pull/224) to configure the KDF.
I've tested this with https://codeberg.org/esoteric_programmer/matrix-jukebox and it is compatible with Element Call.
Fixed: https://github.com/livekit/rust-sdks/issues/796
### Fixes
- Add disconnectReason to Room::close
- End-to-end testing for video streams
- Fix H.264 codec matching
#### add bounded buffer to audio_stream, and use 10 frames as the default
##945 by @xianshijing-lk
#### fix PC timeout when connecting with can_subscribe=false
##955 by @s-hamdananwar
When a participant connects with `canSubscribe=false` in their token, the server sends `subscriber_primary=false` in the JoinResponse and does not send a subscriber offer. This results in `wait_pc_connection` timing out as it is expecting a subscriber PC even when the publisher PC is primary. This PR will skip waiting for subscriber PC when `subscriber_primary=false`.
#### Send client os and os_version from rust
##952 by @MaxHeimbrock
Adds [os_info](https://crates.io/crates/os_info) crate as dependency and sends the data for client connections.
## 0.7.33 (2026-03-13)
### Fixes
#### enhanced build configuration to support macOS and iOS platforms with proper system library linking
##847 by @SchmErik
#### fix video track subscription in single peer connection mode
##914 by @xianshijing-lk