# 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] ## [0.3.26](https://github.com/livekit/rust-sdks/compare/rust-sdks/libwebrtc@0.3.25...rust-sdks/libwebrtc@0.3.26) - 2026-02-16 ### Other - add is_screencast to VideoSource ([#896](https://github.com/livekit/rust-sdks/pull/896)) ## [0.3.25](https://github.com/livekit/rust-sdks/compare/rust-sdks/libwebrtc@0.3.24...rust-sdks/libwebrtc@0.3.25) - 2026-02-09 ### Fixed - fix the 440->441 samples issue and pass a noop callback for release ([#848](https://github.com/livekit/rust-sdks/pull/848)) ### Other - Use workspace dependencies & settings ([#856](https://github.com/livekit/rust-sdks/pull/856)) - allow apm >=10ms frames ([#843](https://github.com/livekit/rust-sdks/pull/843)) ## [0.3.24](https://github.com/livekit/rust-sdks/compare/rust-sdks/libwebrtc@0.3.23...rust-sdks/libwebrtc@0.3.24) - 2026-01-15 ### Other - updated the following local packages: webrtc-sys ## [0.3.23](https://github.com/livekit/rust-sdks/compare/rust-sdks/libwebrtc@0.3.22...rust-sdks/libwebrtc@0.3.23) - 2025-12-19 ### Fixed - Exclude the desktop-capturer module link for mobile. ([#817](https://github.com/livekit/rust-sdks/pull/817)) ## [0.3.22](https://github.com/livekit/rust-sdks/compare/rust-sdks/libwebrtc@0.3.21...rust-sdks/libwebrtc@0.3.22) - 2025-12-17 ### Other - Expose WebRTC's audio_mixer ([#806](https://github.com/livekit/rust-sdks/pull/806)) ## [0.3.21](https://github.com/livekit/rust-sdks/compare/rust-sdks/libwebrtc@0.3.20...rust-sdks/libwebrtc@0.3.21) - 2025-12-04 ### Other - move starting/stopping GLib event loop into libwebrtc crate ([#798](https://github.com/livekit/rust-sdks/pull/798)) - Expose desktop capturer ([#725](https://github.com/livekit/rust-sdks/pull/725)) ## [0.3.20](https://github.com/livekit/rust-sdks/compare/rust-sdks/libwebrtc@0.3.19...rust-sdks/libwebrtc@0.3.20) - 2025-11-20 ### Other - Fix the fast path in capture_frame function, without buffering ([#778](https://github.com/livekit/rust-sdks/pull/778)) ## [0.3.19](https://github.com/livekit/rust-sdks/compare/rust-sdks/libwebrtc@0.3.18...rust-sdks/libwebrtc@0.3.19) - 2025-10-27 ### Other - updated the following local packages: webrtc-sys ## [0.3.18](https://github.com/livekit/rust-sdks/compare/rust-sdks/libwebrtc@0.3.17...rust-sdks/libwebrtc@0.3.18) - 2025-10-22 ### Other - License check ([#746](https://github.com/livekit/rust-sdks/pull/746)) ## [0.3.17](https://github.com/livekit/rust-sdks/compare/rust-sdks/libwebrtc@0.3.16...rust-sdks/libwebrtc@0.3.17) - 2025-10-13 ### Added - *(e2ee)* add data channel encryption ([#708](https://github.com/livekit/rust-sdks/pull/708)) ### Other - Enable buffer scaling ([#473](https://github.com/livekit/rust-sdks/pull/473)) ## [0.3.16](https://github.com/livekit/rust-sdks/compare/rust-sdks/libwebrtc@0.3.15...rust-sdks/libwebrtc@0.3.16) - 2025-10-03 ### Other - updated the following local packages: webrtc-sys ## [0.3.15](https://github.com/livekit/rust-sdks/compare/rust-sdks/libwebrtc@0.3.14...rust-sdks/libwebrtc@0.3.15) - 2025-09-29 ### Fixed - fix Builds/E2E Tests CI. ([#715](https://github.com/livekit/rust-sdks/pull/715)) ## [0.3.14](https://github.com/livekit/rust-sdks/compare/rust-sdks/libwebrtc@0.3.13...rust-sdks/libwebrtc@0.3.14) - 2025-09-09 ### Other - updated the following local packages: webrtc-sys ## [0.3.13](https://github.com/livekit/rust-sdks/compare/rust-sdks/libwebrtc@0.3.12...rust-sdks/libwebrtc@0.3.13) - 2025-09-03 ### Other - updated the following local packages: webrtc-sys # Changelog ## [0.3.12](https://github.com/livekit/rust-sdks/compare/rust-sdks/libwebrtc@0.3.11...rust-sdks/libwebrtc@0.3.12) - 2025-06-17 ### Other - updated the following local packages: livekit-protocol, webrtc-sys ## [0.3.11](https://github.com/livekit/rust-sdks/compare/rust-sdks/libwebrtc@0.3.10...rust-sdks/libwebrtc@0.3.11) - 2025-06-11 ### Fixed - fix uint32 overflow ([#615](https://github.com/livekit/rust-sdks/pull/615)) ### Other - remove ([#633](https://github.com/livekit/rust-sdks/pull/633)) - expose apm stream_delay ([#616](https://github.com/livekit/rust-sdks/pull/616)) - Add i420_to_nv12 ([#605](https://github.com/livekit/rust-sdks/pull/605)) - ffi-v0.13.0 ([#590](https://github.com/livekit/rust-sdks/pull/590)) - add AudioProcessingModule ([#580](https://github.com/livekit/rust-sdks/pull/580)) ## [0.3.10] - 2025-02-05 ### Fixed - Fix build issue ## [0.3.9] - 2025-01-17 ### Added - Expose DataChannel.bufferedAmount property ## [0.3.8] - 2024-12-14 ### Added - bump libwebrtc to m125 ## 0.3.35 (2026-05-29) ### Fixes - Add native video pipeline timing instrumentation for local video measurements, exposing local publish and subscribe timing through async streams and subscriber overlay GPU upload and receive-to-GPU latency metrics through explicit timing observers. ## 0.3.34 (2026-05-21) ### Fixes #### 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 ## 0.3.33 (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 | Mode | Recording | Playout | Use Case | |------|-----------|---------|----------| | 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()); ``` ## 0.3.32 (2026-05-11) ### Fixes - Upgrade protocol to v1.45.8 ## 0.3.31 (2026-05-10) ### Fixes - Fix missing `libwebrtc.jar` for Android builds, harden build scripts - 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) ## 0.3.30 (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(). ## 0.3.29 (2026-04-02) ### Features #### chore: upgrade libwebrtc to m144. ##965 by @cloudwebrtc ### 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. ## 0.3.28 (2026-03-31) ### Fixes - Upgrade to thiserror 2 #### fix: fix unavailable sem symbol for Linux aarch64. ##975 by @cloudwebrtc ## 0.3.27 (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 - Fix H.264 codec matching #### add bounded buffer to audio_stream, and use 10 frames as the default ##945 by @xianshijing-lk #### fix clang build issue from zed patches (#949) ##950 by @cloudwebrtc * webrtc-sys: Use clang instead of gcc * Debug CI output for aarch64-linux * ci: Install lld for aarch64-linux FFI builders * webrtc-sys: Disable CREL