Create a private API for controlling WebRTC's AEC3

content/renderer/media/media_stream_audio_processor.h

Index: content/renderer/media/media_stream_audio_processor.h
diff --git a/content/renderer/media/media_stream_audio_processor.h b/content/renderer/media/media_stream_audio_processor.h
index ed50687ffe20675d62225f04ffe35c4df3ef6479..fa4c8ff71fe1e7c77c3e4fc75a98c7149b9d8bef 100644
--- a/content/renderer/media/media_stream_audio_processor.h
+++ b/content/renderer/media/media_stream_audio_processor.h
@@ -10,6 +10,7 @@
 #include "base/gtest_prod_util.h"
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
+#include "base/optional.h"
 #include "base/single_thread_task_runner.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
@@ -121,6 +122,7 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
   // Called on the main render thread.
   void OnAecDumpFile(const IPC::PlatformFileForTransit& file_handle) override;
   void OnDisableAecDump() override;
+  void OnAec3Enable(bool enable) override;
   void OnIpcClosing() override;
 
   // Returns true if MediaStreamAudioProcessor would modify the audio signal,
@@ -191,6 +193,12 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
 
   // Module to handle processing and format conversion.
   std::unique_ptr<webrtc::AudioProcessing> audio_processing_;
+  bool has_echo_cancellation_;
+  // When this variable is not set, the use of AEC3 is governed by the Finch
+  // experiment and/or WebRTC's own default. When set to true/false, Finch and
+  // WebRTC defaults will be overridden, and AEC3/AEC2 (respectively) will be
+  // used.
+  base::Optional<bool> override_aec3_;
 
   // FIFO to provide 10 ms capture chunks.
   std::unique_ptr<MediaStreamAudioFifo> capture_fifo_;