Rust SDKs

Fixes

• bump protocol to v1.46.4 - #1121 (@lukasIO)

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.

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.

Fixes

• bump protocol to v1.46.4 - #1121 (@lukasIO)

Fixes

• bump protocol to v1.46.4 - #1121 (@lukasIO)
• 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.

Features

• bump protocol to v1.46.4 - #1121 (@lukasIO)

Fixes

• bump protocol to v1.46.4 - #1121 (@lukasIO)

Features

• Introduce pipeline options for remote data tracks, support multiple in-flight frames.

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

Fixes

• Filter internal data streams out of livekit-ffi interface - #1112 (@1egoman)

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

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

Fixes

• Disable default features for zip dependency

Features

• Introduce pipeline options for remote data tracks, support multiple in-flight frames.

Fixes

• Filter internal data streams out of livekit-ffi interface - #1112 (@1egoman)

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

Features

• Introduce pipeline options for remote data tracks, support multiple in-flight frames.

Fixes

• Fix compilation error in depacketizer test by using correct variable name.

Fixes

• Bugfix: Always emit Disconnected on engine close - #1096 (@MaxHeimbrock)
• (WIP) FFI room event ready signal after initial connection - #1068 (@ladvoc, @stephen-derosa)
• Support for large RPC messages using data streams - #1013 (@1egoman)

Fixes

• Bugfix: Always emit Disconnected on engine close - #1096 (@MaxHeimbrock)
• Support for large RPC messages using data streams - #1013 (@1egoman)

Fixes

• Support for large RPC messages using data streams - #1013 (@1egoman)

Features

• FFI logging improvements

Make sample_rate and num_channels optional in NewAudioSourceRequest.

These fields are ignored for AudioSourcePlatform (ADM uses hardware native settings) and for AudioSourceNative fast path (queue_size_ms=0, frame values used directly). Defaults to 48000 Hz and 1 channel when not specified.

Fixes

• fix: don't fire local_track_subscribed during reconnect - #1099 (@davidzhao)
• Fix LocalTrackPublished handle leak - #1065 (@MaxHeimbrock)
• Return EOS event from data track stream read request

Fixes

• Expose room playout delay options in the server API and let the local video publisher recreate rooms with explicit min/max playout delay settings.

Fixes

• fix libwebrtc cache permissions and optimize CI test workflows - #1071 (@davidzhao)

Fixes

• Add AGENTS.md and minor doc revisions
• Add cargo-fuzz target for packet deserialization