Create a private API for controlling WebRTC's AEC3

content/renderer/media/media_stream_audio_processor.h

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_
 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_
 
 #include "base/atomicops.h"
 #include "base/files/file.h"
 #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"
 #include "base/time/time.h"
 #include "content/common/content_export.h"
 #include "content/public/common/media_stream_request.h"
 #include "content/renderer/media/aec_dump_message_filter.h"
 #include "content/renderer/media/audio_repetition_detector.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "media/base/audio_converter.h"
@@ skipping 91 matching lines @@
   const media::AudioParameters& InputFormat() const;
   const media::AudioParameters& OutputFormat() const;
 
   // Accessor to check if the audio processing is enabled or not.
   bool has_audio_processing() const { return audio_processing_ != NULL; }
 
   // AecDumpMessageFilter::AecDumpDelegate implementation.
   // 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,
   // based on the |constraints| and |effects_flags| parsed from a user media
   // request. If the audio signal would not be modified, there is no need to
   // instantiate a MediaStreamAudioProcessor and feed audio through it. Doing so
   // would waste a non-trivial amount of memory and CPU resources.
   //
   // See media::AudioParameters::PlatformEffectsMask for interpretation of
   // |effects_flags|.
@@ skipping 50 matching lines @@
   // both the capture audio thread and the render audio thread.
   base::subtle::Atomic32 render_delay_ms_;
 
 #if ENABLE_AUDIO_REPETITION_DETECTOR
   // Module to detect and report (to UMA) bit exact audio repetition.
   std::unique_ptr<AudioRepetitionDetector> audio_repetition_detector_;
 #endif  // ENABLE_AUDIO_REPETITION_DETECTOR
 
   // 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_;
   // Receives processing output.
   std::unique_ptr<MediaStreamAudioBus> output_bus_;
 
   // FIFO to provide 10 ms render chunks when the AEC is enabled.
   std::unique_ptr<MediaStreamAudioFifo> render_fifo_;
 
   // These are mutated on the main render thread in OnCaptureFormatChanged().
@@ skipping 34 matching lines @@
   // Object for logging UMA stats for echo information when the AEC is enabled.
   // Accessed on the main render thread.
   std::unique_ptr<EchoInformation> echo_information_;
 
   DISALLOW_COPY_AND_ASSIGN(MediaStreamAudioProcessor);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_