fix(llc): fix for multicall audio issues, per-call pc factory

dogfooding/ios/Flutter/AppFrameworkInfo.plist

@@ -20,7 +20,5 @@
   <string>????</string>
   <key>CFBundleVersion</key>
   <string>1.0</string>
-  <key>MinimumOSVersion</key>
-  <string>13.0</string>
 </dict>
 </plist>

dogfooding/linux/flutter/generated_plugin_registrant.cc

@@ -13,7 +13,6 @@
 #include <record_linux/record_linux_plugin.h>
 #include <stream_webrtc_flutter/flutter_web_r_t_c_plugin.h>
 #include <url_launcher_linux/url_launcher_plugin.h>
-#include <volume_controller/volume_controller_plugin.h>
 
 void fl_register_plugins(FlPluginRegistry* registry) {
   g_autoptr(FlPluginRegistrar) desktop_drop_registrar =
@@ -37,7 +36,4 @@ void fl_register_plugins(FlPluginRegistry* registry) {
   g_autoptr(FlPluginRegistrar) url_launcher_linux_registrar =
       fl_plugin_registry_get_registrar_for_plugin(registry, "UrlLauncherPlugin");
   url_launcher_plugin_register_with_registrar(url_launcher_linux_registrar);
-  g_autoptr(FlPluginRegistrar) volume_controller_registrar =
-      fl_plugin_registry_get_registrar_for_plugin(registry, "VolumeControllerPlugin");
-  volume_controller_plugin_register_with_registrar(volume_controller_registrar);
 }

dogfooding/linux/flutter/generated_plugins.cmake

@@ -10,10 +10,10 @@ list(APPEND FLUTTER_PLUGIN_LIST
   record_linux
   stream_webrtc_flutter
   url_launcher_linux
-  volume_controller
 )
 
 list(APPEND FLUTTER_FFI_PLUGIN_LIST
+  jni
 )
 
 set(PLUGIN_BUNDLED_LIBRARIES)

dogfooding/windows/flutter/generated_plugins.cmake

@@ -18,10 +18,10 @@ list(APPEND FLUTTER_PLUGIN_LIST
   stream_webrtc_flutter
   thumblr_windows
   url_launcher_windows
-  volume_controller
 )
 
 list(APPEND FLUTTER_FFI_PLUGIN_LIST
+  jni
 )
 
 set(PLUGIN_BUNDLED_LIBRARIES)

melos.yaml

@@ -22,7 +22,10 @@ command:
       device_info_plus: ^12.1.0
       share_plus: ^11.0.0
       stream_chat_flutter: ^9.17.0
-      stream_webrtc_flutter: ^2.2.6
+      stream_webrtc_flutter:
+        git:
+          url: https://github.com/GetStream/webrtc-flutter.git
+          ref: chore/per-call-pc-factory
       stream_video_flutter: ^1.3.3
       stream_video_noise_cancellation: ^1.3.3
       stream_video_push_notification: ^1.3.3

packages/stream_video/CHANGELOG.md

@@ -1,3 +1,16 @@
+## Unreleased
+
+### ✅ Added
+* Added `TrackDisableMode` enum and optional `disableMode` parameter to `Call.setMicrophoneEnabled()`. Allows integrators to choose between releasing the microphone hardware on mute (`TrackDisableMode.stopTracks`, the default) or keeping the capture session alive (`TrackDisableMode.disableTracks`). The latter avoids the brief iOS `AVAudioSession` teardown that can duck playback of other participants for ~1–2 seconds — recommended for audio rooms and other playback-sensitive use cases. Note: `disableTracks` keeps the system microphone indicator visible while muted because the capture hardware remains active.
+
+### 🐞 Fixed
+* Fixed sibling-call audio capture being silently broken when another concurrently-active call ended (e.g. a 1:1 ringing call ending alongside a running livestream, or a previous ringing call ending before a new one was accepted). `RtcManager.dispose()` now skips the `pc.removeTrack(sender)` step on the publisher PC when it's about to dispose the PC entirely. The explicit `removeTrack` triggers libwebrtc's per-`Call.AudioState` to issue `ADM.StopRecording()` on the **process-wide shared** `AudioDeviceModule` — with no refcount across PCs — which left every still-active call wired against a stopped capture pipeline. Wholesale `pc.dispose()` doesn't take the same lifecycle path and tears down the PC cleanly. Implemented via a new `removeFromPc` parameter on `unpublishTrack` (default `true`; `false` when called from `dispose`). See `docs/audio-lifecycle-analysis.md` for the full investigation trail.
+* Fixed a sibling call's audio breaking when a ringing 1:1 call ended via `dropIfAloneInRingingFlow` (the remote party hung up first). `Call.end()` and `Call.leave()` now share a single `_disconnect` cleanup path, so both honor `_leaveCallTriggered`, complete `_callLifecycleCompleter`, and short-circuit consistently when the call is already disconnected — previously `Call.end()` skipped these guards, which caused races with concurrent reconnect handlers and with re-enabling the mic on a sibling active call.
+* Made the audio processor teardown in `Call._clear` multi-call aware. The audio processor is owned by `StreamVideo`, not by an individual `Call`, so disabling it on one call's teardown silently dropped noise cancellation on any other still-active call. `_clear` now only stops the global processor when no other active call is configured to use `NoiceCancellationSettingsMode.autoOn`.
+
+### 🔄 Changed
+* `Call.leave()` and `Call.end()` now actually wait for the underlying native teardown before returning. Previously `Call._clear` fire-and-forgot `_session.dispose()`, and `CallSession.close` itself fire-and-forgot the WebRTC manager dispose and the SFU WebSocket disconnect, so callers could observe `leave()`/`end()` "complete" while peer connections, local audio tracks, and audio sources were still being torn down on the native side. With this change, awaiting `Call.leave()` / `Call.end()` is enough to guarantee the native cleanup has finished — important when the next thing the integrator does is touch a sibling active call's audio (e.g. resuming a livestream's mic after a 1:1 ringing call ends). Leave/end will take slightly longer to return; if you need fire-and-forget semantics, wrap the call in `unawaited(...)` yourself.
+
 ## 1.3.3
 
 ### 🐞 Fixed

packages/stream_video/lib/src/call/call.dart

@@ -47,6 +47,7 @@ import '../utils/subscriptions.dart';
 import '../webrtc/media/media_constraints.dart';
 import '../webrtc/model/rtc_video_dimension.dart';
 import '../webrtc/model/rtc_video_parameters.dart';
+import '../webrtc/model/track_disable_mode.dart';
 import '../webrtc/rtc_audio_api/rtc_audio_api.dart' as rtc_audio;
 import '../webrtc/rtc_manager.dart';
 import '../webrtc/rtc_media_device/rtc_media_device.dart';
@@ -278,6 +279,7 @@ class Call {
 
   CallCredentials? _credentials;
   CallSession? _session;
+  CallSession? get callSession => _session;
   CallSession? _previousSession;
 
   StatsOptions? _sfuStatsOptions;
@@ -847,22 +849,34 @@ class Call {
 
   /// Ends the call for all participants.
   Future<Result<None>> end({String? reason}) async {
-    final state = this.state.value;
-    _logger.d(() => '[end] status: ${state.status}');
+    _logger.d(() => '[end] status: ${state.value.status}');
 
-    if (state.status is! CallStatusActive) {
-      _logger.w(() => '[end] rejected (invalid status): ${state.status}');
-      return Result.error('invalid status: ${state.status}');
+    if (state.value.status is! CallStatusActive) {
+      _logger.w(() => '[end] rejected (invalid status): ${state.value.status}');
+      return Result.error('invalid status: ${state.value.status}');
     }
 
-    _session?.leave(reason: reason ?? 'user is ending the call');
-    await _clear('end');
+    try {
+      final didDisconnect = await _disconnect(
+        sfuLeaveReason: reason ?? 'user is ending the call',
+      );
 
-    final result = await _permissionsManager.endCall();
-    _stateManager.lifecycleCallEnded();
+      // If another disconnect already ran (or is running), don't fire the
+      // server-side endCall a second time and don't re-emit the lifecycle
+      // event.
+      if (!didDisconnect) {
+        _logger.v(() => '[end] disconnect short-circuited');
+        return const Result.success(none);
+      }
 
-    _logger.v(() => '[end] completed: $result');
-    return result;
+      final result = await _permissionsManager.endCall();
+      _stateManager.lifecycleCallEnded();
+
+      _logger.v(() => '[end] completed: $result');
+      return result;
+    } finally {
+      _leaveCallTriggered = false;
+    }
   }
 
   /// Joins the call.
@@ -1170,6 +1184,9 @@ class Call {
         networkMonitor: networkMonitor,
         streamVideo: _streamVideo,
         statsOptions: _sfuStatsOptions!,
+        audioConfigurationPolicy:
+            _stateManager.callState.preferences.audioConfigurationPolicy ??
+            _streamVideo.options.audioConfigurationPolicy,
         leftoverTraceRecords:
             _previousSession
                 ?.getTrace()
@@ -2041,41 +2058,58 @@ class Call {
   ///
   /// - [reason]: optional reason for leaving the call
   Future<Result<None>> leave({DisconnectReason? reason}) async {
-    try {
-      if (_leaveCallTriggered) {
-        _logger.i(() => '[leave] rejected (already leaving call)');
-        return const Result.success(none);
-      }
+    _logger.i(() => '[leave] reason: $reason');
 
-      _leaveCallTriggered = true;
+    try {
+      final didDisconnect = await _disconnect(
+        sfuLeaveReason: _sfuLeaveReason(reason),
+      );
 
-      // Complete the leave completer to cancel ongoing operations
-      if (!_callLifecycleCompleter.isCompleted) {
-        _callLifecycleCompleter.complete();
+      if (didDisconnect) {
+        _stateManager.lifecycleCallDisconnected(reason: reason);
       }
 
-      final state = this.state.value;
-      _logger.i(() => '[leave] state: $state');
+      _logger.v(() => '[leave] finished');
+      return const Result.success(none);
+    } finally {
+      _leaveCallTriggered = false;
+    }
+  }
 
-      if (state.status.isDisconnected) {
-        _logger.d(() => '[leave] rejected (state.status is disconnected)');
-        return const Result.success(none);
-      }
+  /// Shared cleanup sequence for [leave] and [end].
+  ///
+  /// Sets [_leaveCallTriggered], completes [_callLifecycleCompleter], sends
+  /// the SFU leave message, and runs [_clear]. Returns `true` when the
+  /// cleanup actually ran; `false` if it was short-circuited because a
+  /// concurrent disconnect was already in flight or the call was already
+  /// disconnected.
+  Future<bool> _disconnect({required String sfuLeaveReason}) async {
+    if (_leaveCallTriggered) {
+      _logger.i(() => '[disconnect] rejected (already disconnecting)');
+      return false;
+    }
 
-      try {
-        _session?.leave(reason: _sfuLeaveReason(reason));
-      } finally {
-        await _clear('leave');
-      }
+    _leaveCallTriggered = true;
 
-      _stateManager.lifecycleCallDisconnected(reason: reason);
+    // Complete the lifecycle completer to cancel ongoing operations awaiting
+    // it (e.g. _startSession). This must run regardless of whether the
+    // disconnect proceeds further so that nothing gets stuck waiting.
+    if (!_callLifecycleCompleter.isCompleted) {
+      _callLifecycleCompleter.complete();
+    }
 
-      _logger.v(() => '[leave] finished');
+    if (state.value.status.isDisconnected) {
+      _logger.d(() => '[disconnect] rejected (status is disconnected)');
+      return false;
+    }
 
-      return const Result.success(none);
+    try {
+      _session?.leave(reason: sfuLeaveReason);
     } finally {
-      _leaveCallTriggered = false;
+      await _clear('disconnect');
     }
+
+    return true;
   }
 
   String _sfuLeaveReason(DisconnectReason? reason) {
@@ -2109,6 +2143,7 @@ class Call {
     ]) {
       timer.cancel();
     }
+
     _videoModerationTimer?.cancel();
     _videoModerationTimer = null;
 
@@ -2132,7 +2167,9 @@ class Call {
 
     if (_session != null) {
       unawaited(
-        _session!.dispose().catchError((Object e) {
+        _session!.dispose().catchError((
+          Object e,
+        ) {
           _logger.w(() => '[clear] session dispose failed: $e');
         }),
       );
@@ -2288,7 +2325,9 @@ class Call {
       if (CurrentPlatform.isIos) {
         await _session?.rtcManager?.setAppleAudioConfiguration(
           speakerOn: _connectOptions.speakerDefaultOn,
-          policy: _streamVideo.options.audioConfigurationPolicy,
+          policy:
+              _stateManager.callState.preferences.audioConfigurationPolicy ??
+              _streamVideo.options.audioConfigurationPolicy,
         );
       }
     }
@@ -3295,9 +3334,23 @@ class Call {
     }
   }
 
+  /// Enables or disables the microphone for this call.
+  ///
+  /// When [enabled] is `false`, [disableMode] controls how the local audio
+  /// track is muted. Defaults to [TrackDisableMode.stopTracks], which
+  /// releases the microphone hardware on mute so the system privacy
+  /// indicator turns off. Pass [TrackDisableMode.disableTracks] to keep
+  /// the capture session alive — this avoids the brief iOS
+  /// `AVAudioSession` teardown that otherwise ducks playback of other
+  /// participants for ~1–2 s during mute/unmute, at the cost of the
+  /// system microphone indicator remaining visible while muted.
+  /// Recommended for audio rooms and other playback-sensitive use cases.
+  ///
+  /// See [TrackDisableMode] for the full tradeoff.
   Future<Result<None>> setMicrophoneEnabled({
     required bool enabled,
     AudioConstraints? constraints,
+    TrackDisableMode? disableMode,
   }) async {
     if (enabled &&
         state.value.isVideoModerated &&
@@ -3313,6 +3366,7 @@ class Call {
         await _session?.setMicrophoneEnabled(
           enabled,
           constraints: constraints,
+          disableMode: disableMode,
         ) ??
         Result.error('Session is null');
 
@@ -3349,7 +3403,15 @@ class Call {
     return result.map((_) => none);
   }
 
-  Future<bool> requestScreenSharePermission() {
+  Future<bool> requestScreenSharePermission() async {
+    // Request screen share permission from the native factory if available
+    final nativeFactory = await _session?.rtcManager?.pcFactory
+        .ensureNativeFactory();
+
+    if (nativeFactory != null) {
+      return nativeFactory.requestCapturePermission();
+    }
+
     return Helper.requestCapturePermission();
   }
 

packages/stream_video/lib/src/call/session/call_session.dart

@@ -58,6 +58,7 @@ class CallSession extends Disposable {
     required Tracer tracer,
     this.clientPublishOptions,
     this.joinResponseTimeout = const Duration(seconds: 5),
+    AudioConfigurationPolicy? audioConfigurationPolicy,
   }) : _tracer = tracer,
        _streamVideo = streamVideo,
        sfuClient = SfuClient(
@@ -81,6 +82,7 @@ class CallSession extends Disposable {
          callCid: callCid,
          configuration: config.rtcConfig,
          sdpEditor: sdpEditor,
+         audioConfigurationPolicy: audioConfigurationPolicy,
        ) {
     _logger.i(() => '<init> callCid: $callCid, sessionId: $sessionId');
     _observeNetworkStatus();
@@ -520,7 +522,9 @@ class CallSession extends Disposable {
     StreamWebSocketCloseCode code, {
     String? closeReason,
   }) async {
-    _logger.d(() => '[close] code: $code, closeReason: $closeReason');
+    _logger.d(
+      () => '[close] code: $code, closeReason: $closeReason',
+    );
     _isLeavingOrClosed = true;
 
     await _eventsSubscription?.cancel();
@@ -546,7 +550,10 @@ class CallSession extends Disposable {
 
     if (rtcManager != null) {
       unawaited(
-        rtcManager!.dispose().catchError((Object e, StackTrace stk) {
+        rtcManager!.dispose().catchError((
+          Object e,
+          StackTrace stk,
+        ) {
           _logger.w(() => '[close] rtcManager.dispose failed: $e');
         }),
       );
@@ -1042,22 +1049,20 @@ class CallSession extends Disposable {
   Future<Result<RtcLocalTrack>> setMicrophoneEnabled(
     bool enabled, {
     AudioConstraints? constraints,
+    TrackDisableMode? disableMode,
   }) async {
     final rtcManager = this.rtcManager;
     if (rtcManager == null) {
       return Result.error('Unable to set microphone, Call not connected');
     }
 
-    final result = TracerZone.run(
-      _zonedTracer,
-      ++zonedTracerSeq,
-      () async {
-        return rtcManager.setMicrophoneEnabled(
-          enabled: enabled,
-          constraints: constraints,
-        );
-      },
-    );
+    final result = TracerZone.run(_zonedTracer, ++zonedTracerSeq, () async {
+      return rtcManager.setMicrophoneEnabled(
+        enabled: enabled,
+        constraints: constraints,
+        disableMode: disableMode,
+      );
+    });
 
     return result;
   }

packages/stream_video/lib/src/call/session/call_session_factory.dart

@@ -51,6 +51,7 @@ class CallSessionFactory {
     required StreamVideo streamVideo,
     ClientPublishOptions? clientPublishOptions,
     List<TraceRecord> leftoverTraceRecords = const [],
+    AudioConfigurationPolicy? audioConfigurationPolicy,
   }) async {
     final finalSessionId = sessionId ?? const Uuid().v4();
     _logger.d(() => '[makeCallSession] sessionId: $finalSessionId($sessionId)');
@@ -98,6 +99,7 @@ class CallSessionFactory {
       statsOptions: statsOptions,
       streamVideo: streamVideo,
       tracer: tracer,
+      audioConfigurationPolicy: audioConfigurationPolicy,
     );
   }