All notable changes to this project will be documented in this file.
The format is based on Keep a Changelog, and this project adheres to Semantic Versioning.
0.3.23 - 2026-02-16
- add is_screencast to VideoSource (#896)
0.3.22 - 2026-02-09
- fix the 440->441 samples issue and pass a noop callback for release (#848)
- Use workspace dependencies & settings (#856)
0.3.21 - 2026-01-15
- Rename namespace for ffi bridge code. (#836)
0.3.20 - 2025-12-19
- Exclude the desktop-capturer module link for mobile. (#817)
0.3.19 - 2025-12-17
- lazy loading for additional dependencies. (#814)
- Expose WebRTC's audio_mixer (#806)
0.3.18 - 2025-12-04
0.3.17 - 2025-11-20
- fix Android libwebrtc builds (#791)
- Enable H265 encoding & decoding on Nvidia GPU (#776)
- copy desktop_capture.ninja into C++ library archive (#787)
- enable Pipewire (Wayland) and X11 desktop capture (#784)
0.3.16 - 2025-10-27
- fix unable to locate __arm_tpidr2_save for android ffi. (#765)
- Linux hardware acceleration build fixes (#753)
0.3.15 - 2025-10-22
0.3.14 - 2025-10-13
- (e2ee) add data channel encryption (#708)
- fix some potential audio issues, clean up the code a bit, and suppress some warnings (#737)
- fix linux so link issue. (#733)
- change search_dirs to use cc --print-search-dirs instead of clang --print-search-dirs (#697)
0.3.13 - 2025-10-03
- Fix empty audio frames after resample (#722)
0.3.12 - 2025-09-29
- fix Builds/E2E Tests CI. (#715)
0.3.11 - 2025-09-09
- Optional flags for video hw codec. (#701)
0.3.10 - 2025-09-03
- VA-API support for linux. (#638)
- hardware rendering (#695)
0.3.9 - 2025-06-17
- updated the following local packages: webrtc-sys-build
0.3.8 - 2025-06-11
- fix libwebrtc.jar build issue (#586)
- bump version for webrtc (fix win CI) (#650)
- try to fix webrtc build for iOS/macOS. (#646)
- remove (#633)
- expose apm stream_delay (#616)
- Add i420_to_nv12 (#605)
- ffi-v0.13.0 (#590)
- add AudioProcessingModule (#580)
- Expose DataChannel.bufferedAmount property
- bump libwebrtc to m125
- Fix malformed RTC error handling
MAINTAIN_FRAMERATE_AND_RESOLUTIONis now the recommended value (replaces deprecatedDISABLED)DISABLEDis deprecated but still supported for backwards compatibility- Both values map to the same behavior: maintain framerate and resolution, dropping frames if needed
- Fix AV1 subscriber decode when packet trailers are enabled.
- 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, sostd::optional::value()aborted the process instead of throwing. - Add
LK_DISABLE_NVDECto bypass NVIDIA NVDEC decoder registration when the environment variable is set. - 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.
webrtc-sys no longer links against glib-2.0/gobject-2.0/gio-2.0 by default.
Breaking: Wayland screen sharing now requires the glib-main-loop feature on livekit (or libwebrtc).
- 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_videopublisher reusing mutable I420 frame storage after handing frames to WebRTC. - Add per-publication video encoder backend selection. Add a video encoder backend availability query. Remove
LIVEKIT_PREFERRED_HW_ENCODERin favor of per-publication backend selection.
- 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.
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 genericCreateAudioDeviceModule()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
- chore: bump libwebrtc version to webrtc-51ef663
- Add
LIVEKIT_PREFERRED_HW_ENCODERto prefernvencorvaapihardware video encoding when both are available.
- Add fix_license_json_parsing.patch to handle GN warnings in JSON output
- Enable add_licenses.patch for iOS and Android builds (was commented out)
- Restore LICENSE.md copy in iOS build script (regression from #1053)
The license generation script was failing because gn desc --format=json
outputs warnings before the JSON when certain build args trigger deprecation
notices. The new patch strips non-JSON content before parsing.
This PR introduces platform audio device management via WebRTC's Audio Device Module (ADM).
- ADM Proxy: New
AdmProxyclass 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)
| Mode | Recording | Playout | Use Case |
|---|---|---|---|
| Synthetic | NativeAudioSource | Dummy ADM + FFI | Unity audio, agents |
| Platform | Platform ADM mic | Platform ADM speakers | VoIP with AEC |
// 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());Wrap bare ObjC class references in RTC_OBJC_TYPE() in objc_video_factory.mm and objc_video_frame_buffer.mm to support builds with rtc_objc_prefix set.
- Add
set -eto all build scripts so CI properly reports build failures instead of silently creating empty/broken artifacts - Re-add
use_clang_modules=falseto macOS, iOS, and Linux build scripts to fix C++ module compilation errors
Without use_clang_modules=false, builds fail due to libc++ header incompatibilities (on macOS/iOS with Xcode 26.0) or other C++ module issues, resulting in:
- macOS/iOS: Empty
libwebrtc.a(~13KB instead of ~700MB) - Android: Missing
libwebrtc.jar - Linux: Incomplete artifacts
The builds appeared successful because the scripts continued after ninja failures, but now with set -e, failures will be properly reported.
- Fix missing
libwebrtc.jarfor 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)
##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_metadataor use VideoFrame::new().
- Add NVENC check when Nvidia GPU detected - #1015 (@chenosaurus)
##965 by @cloudwebrtc
##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.
- fix unity android build with "livekit" prefixed jni - #983 (@xianshijing-lk)
##975 by @cloudwebrtc
- fix: enable AGC2 adaptive digital controller
- Fix H.264 codec matching
##921 by @onestacked
This PR uses this webrtc-sdk PR to configure the KDF.
I've tested this with https://codeberg.org/esoteric_programmer/matrix-jukebox and it is compatible with Element Call.
Fixed: #796
##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
Before this change, the Rust implementation would only start sending silence frames after missing 10 consecutive audio frames. This could cause WebRTC's audio pipeline to enter an underrun state when audio stopped temporarily.
Once WebRTC enters underrun, resuming audio can significantly increase latency until the pipeline stabilizes again. In testing, this could add hundreds of milliseconds of additional latency when audio resumed shortly after the underrun.
This change ensures silence frames are sent earlier to prevent the audio pipeline from entering underrun. By maintaining a continuous stream of audio (including silence), WebRTC can avoid unnecessary buffering and latency spikes when audio resumes.
In testing, this reduces the additional latency observed after underrun recovery and results in more stable end-to-end audio latency.
##924 by @kubkon