Explicitly drop FFI handles on dispose

.changeset/ffi-handle-cleanup.md

@@ -0,0 +1,5 @@
+---
+livekit-ffi: patch
+---
+
+Clear remaining FFI handles during dispose so platform audio resources are released across repeated initialize/shutdown cycles.

libwebrtc/src/native/peer_connection_factory.rs

@@ -316,6 +316,14 @@ impl PeerConnectionFactory {
     pub fn is_platform_adm_active(&self) -> bool {
         self.sys_handle.audio_device().is_platform_adm_active()
     }
+
+    /// Stops platform/synthetic audio I/O and detaches WebRTC callbacks.
+    ///
+    /// Call before tearing down peer connections so capture worker threads
+    /// cannot deliver frames into transports that are being destroyed.
+    pub fn shutdown_audio_io(&self) {
+        self.sys_handle.shutdown_audio_io();
+    }
 }
 
 #[cfg(test)]

libwebrtc/src/peer_connection_factory.rs

@@ -156,6 +156,9 @@ pub mod native {
         fn release_platform_adm(&self);
         fn platform_adm_ref_count(&self) -> i32;
         fn is_platform_adm_active(&self) -> bool;
+
+        /// Stops platform/synthetic audio I/O before runtime teardown.
+        fn shutdown_audio_io(&self);
     }
 
     impl PeerConnectionFactoryExt for PeerConnectionFactory {
@@ -298,5 +301,9 @@ pub mod native {
         fn is_platform_adm_active(&self) -> bool {
             self.handle.is_platform_adm_active()
         }
+
+        fn shutdown_audio_io(&self) {
+            self.handle.shutdown_audio_io();
+        }
     }
 }

livekit-ffi/src/server/mod.rs

@@ -176,8 +176,9 @@ impl FfiServer {
 
         self.logger.set_capture_logs(false);
 
-        // Drop all handles
         *self.config.lock() = None; // Invalidate the config
+        self.ffi_handles.clear();
+        self.handle_dropped_txs.clear();
     }
 
     pub fn send_event(&self, message: proto::ffi_event::Message) -> FfiResult<()> {

livekit-ffi/src/server/room.rs

@@ -356,6 +356,10 @@ impl FfiRoom {
 
     /// Close the room and stop the tasks
     pub async fn close(&self, server: &'static FfiServer, reason: DisconnectReason) {
+        // Close the room first so local tracks are unpublished and WebRTC
+        // stops platform capture before FFI track handles are dropped.
+        let _ = self.inner.room.close_with_reason(reason.into()).await;
+
         // drop associated track handles
         for (_, &handle) in self.inner.track_handle_lookup.lock().iter() {
             if server.drop_handle(handle) {
@@ -364,8 +368,6 @@ impl FfiRoom {
             }
         }
 
-        let _ = self.inner.room.close_with_reason(reason.into()).await;
-
         let handle = self.handle.lock().await.take();
         if let Some(handle) = handle {
             let _ = handle.close_tx.send(());

livekit/src/platform_audio/mod.rs

@@ -304,8 +304,9 @@ struct PlatformAdmHandle {
 impl Drop for PlatformAdmHandle {
     fn drop(&mut self) {
         log::debug!("PlatformAdmHandle dropped - releasing Platform ADM");
-        // Release Platform ADM reference
-        // When ref_count reaches 0, the Platform ADM is terminated
+        // Stop platform capture before releasing the ADM reference so the
+        // capture worker cannot deliver frames during teardown.
+        let _ = self.runtime.stop_recording();
         self.runtime.release_platform_adm();
         log::info!(
             "PlatformAdmHandle: released Platform ADM (ref_count now: {})",

livekit/src/rtc_engine/lk_runtime.rs

@@ -275,10 +275,18 @@ impl LkRuntime {
     pub(crate) fn is_platform_adm_active(&self) -> bool {
         self.pc_factory.is_platform_adm_active()
     }
+
+    /// Stops platform/synthetic audio I/O before runtime teardown.
+    #[cfg(not(target_arch = "wasm32"))]
+    pub fn shutdown_audio_io(&self) {
+        self.pc_factory.shutdown_audio_io();
+    }
 }
 
 impl Drop for LkRuntime {
     fn drop(&mut self) {
         log::debug!("LkRuntime::drop()");
+        #[cfg(not(target_arch = "wasm32"))]
+        self.shutdown_audio_io();
     }
 }

webrtc-sys/include/livekit/adm_proxy.h

@@ -93,6 +93,13 @@ class AdmProxy : public webrtc::AudioDeviceModule {
   /// Returns true if Platform ADM is currently active (ref_count > 0).
   bool is_platform_adm_active() const;
 
+  /// Stops platform and synthetic audio I/O and detaches the WebRTC
+  /// AudioTransport callback.
+  ///
+  /// Call before tearing down the peer connection factory so capture worker
+  /// threads cannot deliver frames into transports that are being destroyed.
+  void ShutdownAudioIO();
+
   // ===========================================================================
   // Recording/Playout Control
   // ===========================================================================
@@ -227,6 +234,14 @@ class AdmProxy : public webrtc::AudioDeviceModule {
   // Must be called with mutex_ held.
   void SwitchRecordingAdmIfNeeded();
 
+  // Stops active capture/playout and detaches audio callbacks from both ADMs.
+  // Must be called with mutex_ held.
+  void StopAudioIOLocked();
+
+  // Stops platform capture/playout and detaches its callback without touching
+  // synthetic mode. Must be called with mutex_ held.
+  void StopPlatformAudioIOLocked();
+
 #if defined(__ANDROID__)
   // Lazily creates the Platform ADM on Android.
   // Must be called with mutex_ held.

webrtc-sys/include/livekit/peer_connection_factory.h

@@ -68,6 +68,10 @@ class PeerConnectionFactory {
 
   RtpCapabilities rtp_receiver_capabilities(MediaType type) const;
 
+  /// Stops platform/synthetic audio I/O and detaches callbacks on the worker
+  /// thread before peer connection factory teardown.
+  void shutdown_audio_io() const;
+
   std::shared_ptr<RtcRuntime> rtc_runtime() const { return rtc_runtime_; }
   std::shared_ptr<AudioDeviceController> audio_device() const;
 

webrtc-sys/src/adm_proxy.cpp

@@ -71,14 +71,18 @@ AdmProxy::AdmProxy(const webrtc::Environment& env, webrtc::Thread* worker_thread
 AdmProxy::~AdmProxy() {
   RTC_LOG(LS_VERBOSE) << "AdmProxy::~AdmProxy()";
 
-  if (synthetic_adm_) {
-    synthetic_adm_->Terminate();
-    synthetic_adm_ = nullptr;
-  }
+  {
+    webrtc::MutexLock lock(&mutex_);
+    StopAudioIOLocked();
+    if (synthetic_adm_) {
+      synthetic_adm_->Terminate();
+      synthetic_adm_ = nullptr;
+    }
 
-  if (platform_adm_) {
-    platform_adm_->Terminate();
-    platform_adm_ = nullptr;
+    if (platform_adm_) {
+      platform_adm_->Terminate();
+      platform_adm_ = nullptr;
+    }
   }
 }
 
@@ -180,6 +184,7 @@ void AdmProxy::ReleasePlatformAdm() {
   // Note: We do NOT terminate the Platform ADM - it stays alive until destructor.
   // This avoids iOS KVO race conditions from re-creating the ADM.
   if (platform_adm_ref_count_ == 0) {
+    StopPlatformAudioIOLocked();
     SwitchPlayoutModeIfNeeded();
     SwitchRecordingAdmIfNeeded();
   }
@@ -273,6 +278,42 @@ void AdmProxy::SwitchRecordingAdmIfNeeded() {
   }
 }
 
+void AdmProxy::StopPlatformAudioIOLocked() {
+  recording_ = false;
+
+  if (platform_adm_) {
+    platform_adm_->RegisterAudioCallback(nullptr);
+    platform_adm_->StopRecording();
+    platform_adm_->StopPlayout();
+  }
+}
+
+void AdmProxy::StopAudioIOLocked() {
+  recording_ = false;
+  playing_ = false;
+  recording_initialized_ = false;
+  playout_initialized_ = false;
+
+  if (platform_adm_) {
+    platform_adm_->RegisterAudioCallback(nullptr);
+    platform_adm_->StopRecording();
+    platform_adm_->StopPlayout();
+  }
+
+  if (synthetic_adm_) {
+    synthetic_adm_->RegisterAudioCallback(nullptr);
+    synthetic_adm_->StopRecording();
+    synthetic_adm_->StopPlayout();
+  }
+
+  audio_transport_ = nullptr;
+}
+
+void AdmProxy::ShutdownAudioIO() {
+  webrtc::MutexLock lock(&mutex_);
+  StopAudioIOLocked();
+}
+
 // =============================================================================
 // AudioDeviceModule Interface Implementation
 // =============================================================================
@@ -307,6 +348,7 @@ int32_t AdmProxy::Init() {
 
 int32_t AdmProxy::Terminate() {
   webrtc::MutexLock lock(&mutex_);
+  StopAudioIOLocked();
 
   int32_t result = 0;
   if (synthetic_adm_) {
@@ -554,11 +596,20 @@ int32_t AdmProxy::StopRecording() {
   webrtc::MutexLock lock(&mutex_);
   recording_ = false;
 
-  auto* adm = recording_adm();
-  if (adm) {
-    return adm->StopRecording();
+  int32_t result = 0;
+  if (platform_adm_) {
+    const int32_t platform_result = platform_adm_->StopRecording();
+    if (result == 0) {
+      result = platform_result;
+    }
   }
-  return 0;
+  if (synthetic_adm_) {
+    const int32_t synthetic_result = synthetic_adm_->StopRecording();
+    if (result == 0) {
+      result = synthetic_result;
+    }
+  }
+  return result;
 }
 
 bool AdmProxy::Recording() const {

webrtc-sys/src/peer_connection_factory.cpp

@@ -90,6 +90,8 @@ PeerConnectionFactory::PeerConnectionFactory(
 PeerConnectionFactory::~PeerConnectionFactory() {
   RTC_LOG(LS_VERBOSE) << "PeerConnectionFactory::~PeerConnectionFactory()";
 
+  shutdown_audio_io();
+
   peer_factory_ = nullptr;
   audio_device_ = nullptr;
   rtc_runtime_->worker_thread()->BlockingCall(
@@ -163,6 +165,14 @@ std::shared_ptr<AudioDeviceController> PeerConnectionFactory::audio_device() con
   return audio_device_;
 }
 
+void PeerConnectionFactory::shutdown_audio_io() const {
+  rtc_runtime_->worker_thread()->BlockingCall([this] {
+    if (adm_proxy_) {
+      adm_proxy_->ShutdownAudioIO();
+    }
+  });
+}
+
 std::shared_ptr<PeerConnectionFactory> create_peer_connection_factory() {
   return std::make_shared<PeerConnectionFactory>(RtcRuntime::create());
 }

webrtc-sys/src/peer_connection_factory.rs

@@ -89,6 +89,8 @@ pub mod ffi {
 
         fn create_peer_connection_factory() -> SharedPtr<PeerConnectionFactory>;
 
+        fn shutdown_audio_io(self: &PeerConnectionFactory);
+
         fn create_peer_connection(
             self: &PeerConnectionFactory,
             config: RtcConfiguration,