Test submission of complete AEC-dump refactoring.

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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_
 
 #include <memory>
 
+#include "webrtc/modules/audio_processing/include/aec_dump.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/test/gmock.h"
 
 namespace webrtc {
 
 namespace test {
 
 class MockEchoCancellation : public EchoCancellation {
  public:
   virtual ~MockEchoCancellation() {}
@@ skipping 141 matching lines @@
   MOCK_METHOD4(ProcessReverseStream, int(const float* const* src,
                                          const StreamConfig& input_config,
                                          const StreamConfig& output_config,
                                          float* const* dest));
   MOCK_METHOD1(set_stream_delay_ms, int(int delay));
   MOCK_CONST_METHOD0(stream_delay_ms, int());
   MOCK_CONST_METHOD0(was_stream_delay_set, bool());
   MOCK_METHOD1(set_stream_key_pressed, void(bool key_pressed));
   MOCK_METHOD1(set_delay_offset_ms, void(int offset));
   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_METHOD1(StartDebugRecording,
+  //              void(std::unique_ptr<AecDump> aec_dump));
+  MOCK_METHOD0(StopDebugRecording, void());
   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 {
     return high_pass_filter_.get();
   }
   virtual MockLevelEstimator* level_estimator() const {
     return level_estimator_.get();
   }
   virtual MockNoiseSuppression* noise_suppression() const {
     return noise_suppression_.get();
   }
   virtual MockVoiceDetection* voice_detection() const {
     return voice_detection_.get();
   }
 
+  virtual void StartDebugRecording(std::unique_ptr<AecDump> aec_dump) {}
+
  private:
   std::unique_ptr<MockEchoCancellation> echo_cancellation_;
   std::unique_ptr<MockEchoControlMobile> echo_control_mobile_;
   std::unique_ptr<MockGainControl> gain_control_;
   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_