c8db7f4efa
Drives all four seams over a real WebRTC room against a live livekit-server: join under a minted token, gate each turn OUTSIDE the model (served vs refused for unheld-capability / over-ceiling), meter the model calls, seal one efficiency/1 that verifies under the steward key. examples/live_room.rs is the runnable proof (data-plane turns → gate/meter/seal, receipt verified). Clears two upstream blockers so `--features livekit` builds + runs on macOS: - vendor/libwebrtc: libwebrtc 0.3.38 omits user_data that webrtc-sys 0.3.36 requires (broken upstream release pair); patched additively + [patch.crates-io]. - build.rs: emit -ObjC so libwebrtc's ObjC categories force-load, else PeerConnectionFactory init crashes on an unrecognized selector. Default build + tests are unaffected (the feature stays off by default). Co-Authored-By: Claude Opus 4.8 <[email protected]>
329 lines
11 KiB
Markdown
329 lines
11 KiB
Markdown
# Changelog
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All notable changes to this project will be documented in this file.
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The format is based on [Keep a Changelog](https://keepachangelog.com/en/1.0.0/),
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and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0.html).
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## [Unreleased]
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## [0.3.26](https://github.com/livekit/rust-sdks/compare/rust-sdks/[email protected]/[email protected]) - 2026-02-16
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### Other
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- add is_screencast to VideoSource ([#896](https://github.com/livekit/rust-sdks/pull/896))
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## [0.3.25](https://github.com/livekit/rust-sdks/compare/rust-sdks/[email protected]/[email protected]) - 2026-02-09
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### Fixed
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- fix the 440->441 samples issue and pass a noop callback for release ([#848](https://github.com/livekit/rust-sdks/pull/848))
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### Other
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- Use workspace dependencies & settings ([#856](https://github.com/livekit/rust-sdks/pull/856))
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- allow apm >=10ms frames ([#843](https://github.com/livekit/rust-sdks/pull/843))
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## [0.3.24](https://github.com/livekit/rust-sdks/compare/rust-sdks/[email protected]/[email protected]) - 2026-01-15
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### Other
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- updated the following local packages: webrtc-sys
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## [0.3.23](https://github.com/livekit/rust-sdks/compare/rust-sdks/[email protected]/[email protected]) - 2025-12-19
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### Fixed
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- Exclude the desktop-capturer module link for mobile. ([#817](https://github.com/livekit/rust-sdks/pull/817))
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## [0.3.22](https://github.com/livekit/rust-sdks/compare/rust-sdks/[email protected]/[email protected]) - 2025-12-17
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### Other
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- Expose WebRTC's audio_mixer ([#806](https://github.com/livekit/rust-sdks/pull/806))
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## [0.3.21](https://github.com/livekit/rust-sdks/compare/rust-sdks/[email protected]/[email protected]) - 2025-12-04
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### Other
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- move starting/stopping GLib event loop into libwebrtc crate ([#798](https://github.com/livekit/rust-sdks/pull/798))
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- Expose desktop capturer ([#725](https://github.com/livekit/rust-sdks/pull/725))
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## [0.3.20](https://github.com/livekit/rust-sdks/compare/rust-sdks/[email protected]/[email protected]) - 2025-11-20
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### Other
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- Fix the fast path in capture_frame function, without buffering ([#778](https://github.com/livekit/rust-sdks/pull/778))
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## [0.3.19](https://github.com/livekit/rust-sdks/compare/rust-sdks/[email protected]/[email protected]) - 2025-10-27
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### Other
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- updated the following local packages: webrtc-sys
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## [0.3.18](https://github.com/livekit/rust-sdks/compare/rust-sdks/[email protected]/[email protected]) - 2025-10-22
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### Other
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- License check ([#746](https://github.com/livekit/rust-sdks/pull/746))
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## [0.3.17](https://github.com/livekit/rust-sdks/compare/rust-sdks/[email protected]/[email protected]) - 2025-10-13
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### Added
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- *(e2ee)* add data channel encryption ([#708](https://github.com/livekit/rust-sdks/pull/708))
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### Other
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- Enable buffer scaling ([#473](https://github.com/livekit/rust-sdks/pull/473))
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## [0.3.16](https://github.com/livekit/rust-sdks/compare/rust-sdks/[email protected]/[email protected]) - 2025-10-03
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### Other
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- updated the following local packages: webrtc-sys
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## [0.3.15](https://github.com/livekit/rust-sdks/compare/rust-sdks/[email protected]/[email protected]) - 2025-09-29
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### Fixed
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- fix Builds/E2E Tests CI. ([#715](https://github.com/livekit/rust-sdks/pull/715))
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## [0.3.14](https://github.com/livekit/rust-sdks/compare/rust-sdks/[email protected]/[email protected]) - 2025-09-09
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### Other
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- updated the following local packages: webrtc-sys
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## [0.3.13](https://github.com/livekit/rust-sdks/compare/rust-sdks/[email protected]/[email protected]) - 2025-09-03
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### Other
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- updated the following local packages: webrtc-sys
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# Changelog
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## [0.3.12](https://github.com/livekit/rust-sdks/compare/rust-sdks/[email protected]/[email protected]) - 2025-06-17
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### Other
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- updated the following local packages: livekit-protocol, webrtc-sys
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## [0.3.11](https://github.com/livekit/rust-sdks/compare/rust-sdks/[email protected]/[email protected]) - 2025-06-11
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### Fixed
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- fix uint32 overflow ([#615](https://github.com/livekit/rust-sdks/pull/615))
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### Other
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- remove ([#633](https://github.com/livekit/rust-sdks/pull/633))
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- expose apm stream_delay ([#616](https://github.com/livekit/rust-sdks/pull/616))
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- Add i420_to_nv12 ([#605](https://github.com/livekit/rust-sdks/pull/605))
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- ffi-v0.13.0 ([#590](https://github.com/livekit/rust-sdks/pull/590))
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- add AudioProcessingModule ([#580](https://github.com/livekit/rust-sdks/pull/580))
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## [0.3.10] - 2025-02-05
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### Fixed
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- Fix build issue
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## [0.3.9] - 2025-01-17
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### Added
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- Expose DataChannel.bufferedAmount property
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## [0.3.8] - 2024-12-14
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### Added
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- bump libwebrtc to m125
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## 0.3.38 (2026-06-23)
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### Fixes
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- Upgrade protocol to v1.48.0
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## 0.3.37 (2026-06-17)
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### Features
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#### Add dynacast support - #1003 (@chenosaurus, @stephen-derosa)
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This includes a minor breaking change for `libwebrtc`: `RtpParameters` now
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contains additional RTP sender state that must be preserved when round-tripping
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through `set_parameters()`.
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#### Make GLib an opt-in dependency
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`webrtc-sys` no longer links against `glib-2.0`/`gobject-2.0`/`gio-2.0` by default.
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Breaking: Wayland screen sharing now requires the `glib-main-loop` feature on `livekit` (or `libwebrtc`).
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### Fixes
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- Fix silent subscription failures in single-pc mode when the SFU reuses an existing empty transceiver for a new remote track. Also make `RtpTransceiver::mid()` safe to call on transceivers that haven't been negotiated yet — libwebrtc is built with `-fno-exceptions`, so `std::optional::value()` aborted the process instead of throwing.
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- Add `LK_DISABLE_NVDEC` to bypass NVIDIA NVDEC decoder registration when the environment variable is set.
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- Add Jetson DMA-buffer video publishing support for libargus MIPI capture and the Jetson hardware encoder, including AV1 hardware encoding on supported Jetson Orin devices.
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## 0.3.36 (2026-06-09)
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### Fixes
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- Fix NVIDIA encoder I420 uploads to copy each plane using its actual source stride, avoiding chroma corruption when source frames use padded YUV planes. Also fix the `local_video` publisher reusing mutable I420 frame storage after handing frames to WebRTC.
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- Keep one-frame native video streams as latest-frame queues. Move local video subscriber timing metrics into the example and use the WGPU paint callback as the render boundary.
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- Add per-publication video encoder backend selection. Add a video encoder backend availability query. Remove `LIVEKIT_PREFERRED_HW_ENCODER` in favor of per-publication backend selection.
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## 0.3.35 (2026-05-29)
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### Fixes
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- 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.
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## 0.3.34 (2026-05-21)
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### Fixes
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#### feat: add Android application context initialization for PlatformAudio support.
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Android requires `ContextUtils.initialize(applicationContext)` before WebRTC audio components can be created. This change:
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- Adds `livekit_ffi_initialize_android_context()` C FFI function for Unity and other FFI consumers
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- Uses `CreateAndroidAudioDeviceModule()` instead of generic `CreateAudioDeviceModule()` on Android
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- Handles empty device GUIDs on Android (falls back to index 0)
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- Documents Android-specific limitations: single default device, no app-level device selection
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Platform notes:
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- Android device enumeration returns only one "default" device with empty name/GUID
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- Audio routing (speaker/earpiece/Bluetooth) is controlled by Android's AudioManager, not WebRTC
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## 0.3.33 (2026-05-14)
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### Fixes
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- feat: add scalability mode for AV1/VP9. - #1076 (@cloudwebrtc)
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- Add `LIVEKIT_PREFERRED_HW_ENCODER` to prefer `nvenc` or `vaapi` hardware video encoding when both are available.
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- Relocate unrelated types out of `livekit-protocol`
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#### Get WebRTC ADM into Rust - #1037 (@xianshijing-lk)
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This PR introduces platform audio device management via WebRTC's Audio Device Module (ADM).
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#### Features
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- **ADM Proxy**: New `AdmProxy` class that switches between Dummy ADM (synthetic mode) and Platform ADM (real audio I/O)
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- **PlatformAudio API**: High-level Rust API for microphone capture and speaker playout with AEC/AGC/NS
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- **Device enumeration**: List and select recording/playout devices by index or GUID
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- **Mode switching**: Seamlessly switch between synthetic mode (FFI callbacks) and platform mode (native speakers) while audio is active
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- **FFI platform audio support**: Expose platform audio device enumeration and selection through `livekit-ffi`
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- **Audio processing**: Configure echo cancellation, noise suppression, and auto gain control with platform-specific defaults (hardware on iOS, software elsewhere)
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#### Audio Modes
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| Mode | Recording | Playout | Use Case |
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|------|-----------|---------|----------|
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| Synthetic | NativeAudioSource | Dummy ADM + FFI | Unity audio, agents |
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| Platform | Platform ADM mic | Platform ADM speakers | VoIP with AEC |
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#### API
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```rust
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// Create PlatformAudio for microphone/speaker access
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let audio = PlatformAudio::new()?;
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// Enumerate and select devices
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for i in 0..audio.recording_devices() as u16 {
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println!("Mic {}: {}", i, audio.recording_device_name(i));
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}
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audio.set_recording_device(0)?;
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// Create audio track for publishing
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let track = LocalAudioTrack::create_audio_track("mic", audio.rtc_source());
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```
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## 0.3.32 (2026-05-11)
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### Fixes
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- Upgrade protocol to v1.45.8
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## 0.3.31 (2026-05-10)
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### Fixes
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- Fix missing `libwebrtc.jar` for Android builds, harden build scripts
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- fix race in download_webrtc to reduce flaky build - #1047 (@hechen-eng)
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- Improve WebRTC build scripts and add external_audio_source patch - #1053 (@xianshijing-lk)
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## 0.3.30 (2026-04-23)
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### Features
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#### Add support for frame level packet trailer
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##890 by @chenosaurus
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- Add support to attach/parse frame level timestamps & frame ID to VideoTracks as a custom payload trailer.
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- Breaking change in VideoFrame API, must include `frame_metadata` or use VideoFrame::new().
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## 0.3.29 (2026-04-02)
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### Features
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#### chore: upgrade libwebrtc to m144.
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##965 by @cloudwebrtc
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### Fixes
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#### use the bounded buffer for video stream
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##956 by @xianshijing-lk
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Before this PR, it uses an unbounded buffer for video stream, that will cause multiple problems:
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1, video will be lagged behind if rendering is slow or just wake up from background
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2, it will be out of sync with audio
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This PRs provides options to set a bounded buffer for video stream, and use 1 buffer as the default option.
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## 0.3.28 (2026-03-31)
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### Fixes
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- Upgrade to thiserror 2
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#### fix: fix unavailable sem symbol for Linux aarch64.
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##975 by @cloudwebrtc
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## 0.3.27 (2026-03-22)
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### Features
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#### E2EE: allow setting key_ring_size and key_derivation_algorithm, update webrtc to m144
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##921 by @onestacked
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This PR uses [this webrtc-sdk PR](https://github.com/webrtc-sdk/webrtc/pull/224) to configure the KDF.
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I've tested this with https://codeberg.org/esoteric_programmer/matrix-jukebox and it is compatible with Element Call.
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Fixed: https://github.com/livekit/rust-sdks/issues/796
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### Fixes
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- Fix H.264 codec matching
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#### add bounded buffer to audio_stream, and use 10 frames as the default
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##945 by @xianshijing-lk
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#### fix clang build issue from zed patches (#949)
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##950 by @cloudwebrtc
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* webrtc-sys: Use clang instead of gcc
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* Debug CI output for aarch64-linux
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* ci: Install lld for aarch64-linux FFI builders
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* webrtc-sys: Disable CREL
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