Fix Chromium clang plugin warnings

webrtc/modules/audio_processing/test/audio_processing_simulator.h

 /*
  *  Copyright (c) 2016 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 10 matching lines @@
 #include "webrtc/base/optional.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 
 namespace webrtc {
 namespace test {
 
 // Holds all the parameters available for controlling the simulation.
 struct SimulationSettings {
+  SimulationSettings();
+  SimulationSettings(const SimulationSettings&);
+  ~SimulationSettings();
   rtc::Optional<int> stream_delay;
   rtc::Optional<int> stream_drift_samples;
   rtc::Optional<int> output_sample_rate_hz;
   rtc::Optional<int> output_num_channels;
   rtc::Optional<int> reverse_output_sample_rate_hz;
   rtc::Optional<int> reverse_output_num_channels;
   rtc::Optional<std::string> microphone_positions;
   int target_angle_degrees = 90;
   rtc::Optional<std::string> output_filename;
   rtc::Optional<std::string> reverse_output_filename;
@@ skipping 44 matching lines @@
 };
 
 // Copies samples present in a ChannelBuffer into an AudioFrame.
 void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest);
 
 // Provides common functionality for performing audioprocessing simulations.
 class AudioProcessingSimulator {
  public:
   static const int kChunksPerSecond = 1000 / AudioProcessing::kChunkSizeMs;
 
-  explicit AudioProcessingSimulator(const SimulationSettings& settings)
-      : settings_(settings) {}
-  virtual ~AudioProcessingSimulator() {}
+  explicit AudioProcessingSimulator(const SimulationSettings& settings);
+  virtual ~AudioProcessingSimulator();
 
   // Processes the data in the input.
   virtual void Process() = 0;
 
   // Returns the execution time of all AudioProcessing calls.
   const TickIntervalStats& proc_time() const { return proc_time_; }
 
   // Reports whether the processed recording was bitexact.
   bool OutputWasBitexact() { return bitexact_output_; }
 
@@ skipping 59 matching lines @@
   std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_writer_;
   TickIntervalStats proc_time_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator);
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_