v2.9.0

.version

@@ -1 +1 @@
-2.8.0
+2.9.0

CHANGELOG.md

@@ -1,5 +1,36 @@
 # CHANGELOG
 
+## 2.9.0
+
+* Added: Add native certificate pinning for SDK-owned connections
+* Added: Add RPC v2 (data-stream-based payloads) with v1 fallback
+* Added: Audio engine availability and externalCallSystem mode for CallKit
+* Added: AudioManager audio session options with engine-driven native lifecycle and platform routing controls
+* Added: Add ConnectionCheck utility for diagnosing connection issues (port of the client-sdk-js connection helper)
+* Added: Audio frame capture on Linux/Windows
+* Added: Microphone mute mode control on iOS/macOS
+* Added: Runtime audio processing controls for local audio tracks
+* Added: Widget Placeholder added for VideoTrackRenderer
+* Changed: Default VideoTrackRenderer renderMode to auto, which currently resolves to texture rendering
+* Fixed: Apply Android media audio attributes during WebRTC initialization
+* Fixed: Use initialization audio options as the default Android session policy
+* Fixed: Avoid sticky Android speaker routing when updating route preference
+* Fixed: Clean up local audio tracks when recording start fails
+* Fixed: Throw platformUnavailable when runtime audio processing is unsupported
+* Fixed: Apply create-time audio processing when local recording is prepared
+* Fixed: Remove non-public buttonPressed: selector from broadcast picker activation
+* Fixed: Allow selectAudioOutput on Android
+* Fixed: Support platform video rendering on macOS
+* Fixed: Emit a single disconnected event when connecting fails
+* Fixed: Handle switching video render modes without stale renderers
+* Docs: Add AGENTS.md with agent/contributor guidelines
+
+## 2.8.1
+
+* Added: Add agent deployment targeting to token source options
+* Fixed: Android plugin compatibility with AGP 9 built-in Kotlin
+* Fixed: Use maintain-resolution as the default video degradation preference for local video publishing
+
 ## 2.8.0
 
 * Added: Session API support for simpler E2EE setup

README.md

@@ -46,7 +46,7 @@ Include this package to your `pubspec.yaml`
 ```yaml
 ---
 dependencies:
-  livekit_client: ^2.8.0
+  livekit_client: ^2.9.0
 ```
 
 ### iOS
@@ -161,7 +161,7 @@ await AudioManager.instance.setAudioSessionOptions(
 );
 ```
 
-See the [audio session guide](https://github.com/livekit/client-sdk-flutter/blob/main/docs/audio.md) for more.
+See the [audio session guide](https://github.com/livekit/client-sdk-flutter/blob/main/doc/audio.md) for more.
 
 ### Desktop support
 
@@ -411,7 +411,7 @@ Widget build(BuildContext context) {
 
 Audio tracks are played automatically as long as you are subscribed to them.
 
-LiveKit owns the platform audio session through `AudioManager`. A call is managed automatically with no setup. Speaker routing and, when you need it, manual session control go through the same object. See the [audio session guide](https://github.com/livekit/client-sdk-flutter/blob/main/docs/audio.md) for examples covering the automatic and manual modes, speaker routing, per platform overrides, and migration from the older `Hardware` APIs.
+LiveKit owns the platform audio session through `AudioManager`. A call is managed automatically with no setup. Speaker routing and, when you need it, manual session control go through the same object. See the [audio session guide](https://github.com/livekit/client-sdk-flutter/blob/main/doc/audio.md) for examples covering the automatic and manual modes, speaker routing, per platform overrides, and migration from the older `Hardware` APIs.
 
 ```dart
 // A call is managed automatically. Route to the speaker when you want.

ios/livekit_client.podspec

@@ -1,6 +1,6 @@
 Pod::Spec.new do |s|
   s.name                = 'livekit_client'
-  s.version             = '2.8.0'
+  s.version             = '2.9.0'
   s.summary             = 'Open source platform for real-time audio and video.'
   s.description         = 'Open source platform for real-time audio and video.'
   s.homepage            = 'https://livekit.io/'

lib/src/livekit.dart

@@ -23,7 +23,7 @@ import 'support/webrtc_initialize_options.dart';
 /// Main entry point to connect to a room.
 /// {@category Room}
 class LiveKitClient {
-  static const version = '2.8.0';
+  static const version = '2.9.0';
 
   /// Initialize the WebRTC plugin.
   ///

macos/livekit_client.podspec

@@ -1,6 +1,6 @@
 Pod::Spec.new do |s|
   s.name                = 'livekit_client'
-  s.version             = '2.8.0'
+  s.version             = '2.9.0'
   s.summary             = 'Open source platform for real-time audio and video.'
   s.description         = 'Open source platform for real-time audio and video.'
   s.homepage            = 'https://livekit.io/'

pubspec.yaml

@@ -15,9 +15,13 @@
 name: livekit_client
 description: Flutter Client SDK for LiveKit. Build real-time video and audio
   into your apps. Supports iOS, Android, and Web.
-version: 2.8.0
+version: 2.9.0
 homepage: https://github.com/livekit/client-sdk-flutter
 
+# Test fixture keys for certificate pinning tests, not real secrets.
+false_secrets:
+  - /test/support/certificate_pinning_io_test.dart
+
 environment:
   sdk: ">=3.6.0 <4.0.0"
   flutter: ">=3.27.0"