Fix Chromium clang plugin warnings

webrtc/modules/audio_processing/test/audio_processing_simulator.cc

 /*
  *  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 26 matching lines @@
 std::string GetIndexedOutputWavFilename(const std::string& wav_name,
                                         int counter) {
   std::stringstream ss;
   ss << wav_name.substr(0, wav_name.size() - 4) << "_" << counter
      << wav_name.substr(wav_name.size() - 4);
   return ss.str();
 }
 
 }  // namespace
 
+SimulationSettings::SimulationSettings() = default;
+SimulationSettings::SimulationSettings(const SimulationSettings&) = default;
+SimulationSettings::~SimulationSettings() = default;
+
 void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) {
   RTC_CHECK_EQ(src.num_channels(), dest->num_channels_);
   RTC_CHECK_EQ(src.num_frames(), dest->samples_per_channel_);
   for (size_t ch = 0; ch < dest->num_channels_; ++ch) {
     for (size_t sample = 0; sample < dest->samples_per_channel_; ++sample) {
       dest->data_[sample * dest->num_channels_ + ch] =
           src.channels()[ch][sample] * 32767;
     }
   }
 }
 
+AudioProcessingSimulator::AudioProcessingSimulator(
+    const SimulationSettings& settings)
+    : settings_(settings) {}
+
+AudioProcessingSimulator::~AudioProcessingSimulator() = default;
+
 AudioProcessingSimulator::ScopedTimer::~ScopedTimer() {
   int64_t interval = rtc::TimeNanos() - start_time_;
   proc_time_->sum += interval;
   proc_time_->max = std::max(proc_time_->max, interval);
   proc_time_->min = std::min(proc_time_->min, interval);
 }
 
 void AudioProcessingSimulator::ProcessStream(bool fixed_interface) {
   if (fixed_interface) {
     {
@@ skipping 261 matching lines @@
     size_t kMaxFilenameSize = AudioProcessing::kMaxFilenameSize;
     RTC_CHECK_LE(settings_.aec_dump_output_filename->size(), kMaxFilenameSize);
     RTC_CHECK_EQ(AudioProcessing::kNoError,
                  ap_->StartDebugRecording(
                      settings_.aec_dump_output_filename->c_str(), -1));
   }
 }
 
 }  // namespace test
 }  // namespace webrtc