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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, and this project adheres to Semantic Versioning.

[Unreleased]

0.3.26 - 2026-02-16

Other

  • add is_screencast to VideoSource (#896)

0.3.25 - 2026-02-09

Fixed

  • fix the 440->441 samples issue and pass a noop callback for release (#848)

Other

  • Use workspace dependencies & settings (#856)
  • allow apm >=10ms frames (#843)

0.3.24 - 2026-01-15

Other

  • updated the following local packages: webrtc-sys

0.3.23 - 2025-12-19

Fixed

  • Exclude the desktop-capturer module link for mobile. (#817)

0.3.22 - 2025-12-17

Other

  • Expose WebRTC's audio_mixer (#806)

0.3.21 - 2025-12-04

Other

  • move starting/stopping GLib event loop into libwebrtc crate (#798)
  • Expose desktop capturer (#725)

0.3.20 - 2025-11-20

Other

  • Fix the fast path in capture_frame function, without buffering (#778)

0.3.19 - 2025-10-27

Other

  • updated the following local packages: webrtc-sys

0.3.18 - 2025-10-22

Other

  • License check (#746)

0.3.17 - 2025-10-13

Added

  • (e2ee) add data channel encryption (#708)

Other

  • Enable buffer scaling (#473)

0.3.16 - 2025-10-03

Other

  • updated the following local packages: webrtc-sys

0.3.15 - 2025-09-29

Fixed

  • fix Builds/E2E Tests CI. (#715)

0.3.14 - 2025-09-09

Other

  • updated the following local packages: webrtc-sys

0.3.13 - 2025-09-03

Other

  • updated the following local packages: webrtc-sys

Changelog

0.3.12 - 2025-06-17

Other

  • updated the following local packages: livekit-protocol, webrtc-sys

0.3.11 - 2025-06-11

Fixed

  • fix uint32 overflow (#615)

Other

  • remove (#633)
  • expose apm stream_delay (#616)
  • Add i420_to_nv12 (#605)
  • ffi-v0.13.0 (#590)
  • add AudioProcessingModule (#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

0.3.40 (2026-07-09)

Fixes

  • Fix malformed RTC error handling
  • introduce LiveKitAPI construct, added smoke tests - #1220 (@davidzhao)

0.3.39 (2026-06-30)

Features

Improve initial video quality by setting x-google-start-bitrate SDP hint for all video codecs (VP8, VP9, AV1, H264, H265) and defaulting to MaintainResolution degradation preference.

This addresses the issue where video starts blurry for several seconds before improving, by:

  1. Telling WebRTC's bandwidth estimator to start at 70% of target bitrate instead of ramping up from ~300kbps
  2. Preferring frame drops over resolution reduction when bandwidth is constrained

The DegradationPreference option is now exposed via FFI for Python, C++, Unity, and Node SDKs.

Add MaintainFramerateAndResolution to DegradationPreference enum to align with WebRTC M144.

  • MAINTAIN_FRAMERATE_AND_RESOLUTION is now the recommended value (replaces deprecated DISABLED)
  • DISABLED is deprecated but still supported for backwards compatibility
  • Both values map to the same behavior: maintain framerate and resolution, dropping frames if needed

Fixes

  • Fix AV1 subscriber decode when packet trailers are enabled.

0.3.38 (2026-06-23)

Fixes

  • Upgrade protocol to v1.48.0

0.3.37 (2026-06-17)

Features

Add dynacast support - #1003 (@chenosaurus, @stephen-derosa)

This includes a minor breaking change for libwebrtc: RtpParameters now contains additional RTP sender state that must be preserved when round-tripping through set_parameters().

Make GLib an opt-in dependency

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).

Fixes

  • 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.
  • Add LK_DISABLE_NVDEC to 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.

0.3.36 (2026-06-09)

Fixes

  • 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.
  • 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.
  • 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.

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

// 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 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

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