Test submission of complete AEC-dump refactoring.

webrtc/modules/audio_processing/test/audio_processing_simulator.h

Index: webrtc/modules/audio_processing/test/audio_processing_simulator.h
diff --git a/webrtc/modules/audio_processing/test/audio_processing_simulator.h b/webrtc/modules/audio_processing/test/audio_processing_simulator.h
index f597fa101a76e7a1a705464957458308fb1b0f81..50162126f0866f18d0cd0cbb94a5722ee1c63a09 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.h
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.h
@@ -17,9 +17,10 @@
 #include <memory>
 #include <string>
 
-#include "webrtc/base/timeutils.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/optional.h"
+#include "webrtc/base/task_queue.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
@@ -149,6 +150,8 @@ class AudioProcessingSimulator {
                                   int reverse_output_num_channels);
 
   const SimulationSettings settings_;
+  rtc::TaskQueue worker_queue_;  // Important to make this outlive last
+                                 // call to StopDebugRecording.
   std::unique_ptr<AudioProcessing> ap_;
 
   std::unique_ptr<ChannelBuffer<float>> in_buf_;