Add native self-hosted instance connection to fluxer_desktop

Trimmed monorepo checkout (fluxer_desktop + packages/voice_engine_v2 + tools/ci) with a "Connect to a Different Server" menu item and popout that lets the desktop app switch to any self-hosted Fluxer instance, plus fixes for well-known discovery on single-domain self-hosted deployments and a false-positive ERR_ABORTED on same-origin client redirects during the switch. Defaults to chat.fluxr.chat and uses an isolated userData directory from the official build.
2026-07-01 18:22:43 -04:00
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+# 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