Revert of New statistics interface for APM

webrtc/modules/audio_processing/include/mock_audio_processing.h

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 165 matching lines @@
   MOCK_CONST_METHOD0(delay_offset_ms, int());
   MOCK_METHOD2(StartDebugRecording, int(const char filename[kMaxFilenameSize],
                                         int64_t max_log_size_bytes));
   MOCK_METHOD2(StartDebugRecording, int(FILE* handle,
                                         int64_t max_log_size_bytes));
   MOCK_METHOD1(StartDebugRecording, int (FILE* handle));
   MOCK_METHOD1(StartDebugRecordingForPlatformFile,
                int(rtc::PlatformFile handle));
   MOCK_METHOD0(StopDebugRecording, int());
   MOCK_METHOD0(UpdateHistogramsOnCallEnd, void());
-  MOCK_CONST_METHOD0(GetStatistics, AudioProcessingStatistics());
   virtual MockEchoCancellation* echo_cancellation() const {
     return echo_cancellation_.get();
   }
   virtual MockEchoControlMobile* echo_control_mobile() const {
     return echo_control_mobile_.get();
   }
   virtual MockGainControl* gain_control() const {
     return gain_control_.get();
   }
   virtual MockHighPassFilter* high_pass_filter() const {
@@ skipping 16 matching lines @@
   std::unique_ptr<MockHighPassFilter> high_pass_filter_;
   std::unique_ptr<MockLevelEstimator> level_estimator_;
   std::unique_ptr<MockNoiseSuppression> noise_suppression_;
   std::unique_ptr<MockVoiceDetection> voice_detection_;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_