Add detection for repeated audio in capturing.

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/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"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"
 
 namespace blink {
 class WebMediaConstraints;
 }
 
 namespace media {
@@ skipping 79 matching lines @@
   // Called on the main render thread.
   void OnAecDumpFile(const IPC::PlatformFileForTransit& file_handle) override;
   void OnDisableAecDump() override;
   void OnIpcClosing() override;
 
  protected:
   ~MediaStreamAudioProcessor() override;
 
  private:
   friend class MediaStreamAudioProcessorTest;
+
   FRIEND_TEST_ALL_PREFIXES(MediaStreamAudioProcessorTest,
                            GetAecDumpMessageFilter);
 
   // WebRtcPlayoutDataSource::Sink implementation.
   void OnPlayoutData(media::AudioBus* audio_bus,
                      int sample_rate,
                      int audio_delay_milliseconds) override;
   void OnPlayoutDataSourceChanged() override;
 
   // webrtc::AudioProcessorInterface implementation.
@@ skipping 20 matching lines @@
                   int process_frames,
                   base::TimeDelta capture_delay,
                   int volume,
                   bool key_pressed,
                   float* const* output_ptrs);
 
   // Cached value for the render delay latency. This member is accessed by
   // both the capture audio thread and the render audio thread.
   base::subtle::Atomic32 render_delay_ms_;
 
+  // Module to detect and report (to UMA) bit exact audio repetition.
+  scoped_ptr<AudioRepetitionDetector> audio_repetition_detector_;
+
   // Module to handle processing and format conversion.
   scoped_ptr<webrtc::AudioProcessing> audio_processing_;
 
   // FIFO to provide 10 ms capture chunks.
   scoped_ptr<MediaStreamAudioFifo> capture_fifo_;
   // Receives processing output.
   scoped_ptr<MediaStreamAudioBus> output_bus_;
 
   // FIFO to provide 10 ms render chunks when the AEC is enabled.
   scoped_ptr<MediaStreamAudioFifo> render_fifo_;
@@ skipping 35 matching lines @@
   // Object for logging echo information when the AEC is enabled. Accessible by
   // the libjingle thread through GetStats().
   scoped_ptr<EchoInformation> echo_information_;
 
   DISALLOW_COPY_AND_ASSIGN(MediaStreamAudioProcessor);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_