Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/

webrtc/modules/audio_processing/audio_processing_impl.cc

 /*
  *  Copyright (c) 2012 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 72 matching lines @@
 
 struct AudioProcessingImpl::ApmPublicSubmodules {
   ApmPublicSubmodules()
       : echo_cancellation(nullptr),
         echo_control_mobile(nullptr),
         gain_control(nullptr) {}
   // Accessed externally of APM without any lock acquired.
   EchoCancellationImpl* echo_cancellation;
   EchoControlMobileImpl* echo_control_mobile;
   GainControlImpl* gain_control;
-  rtc::scoped_ptr<HighPassFilterImpl> high_pass_filter;
-  rtc::scoped_ptr<LevelEstimatorImpl> level_estimator;
-  rtc::scoped_ptr<NoiseSuppressionImpl> noise_suppression;
-  rtc::scoped_ptr<VoiceDetectionImpl> voice_detection;
-  rtc::scoped_ptr<GainControlForExperimentalAgc>
+  std::unique_ptr<HighPassFilterImpl> high_pass_filter;
+  std::unique_ptr<LevelEstimatorImpl> level_estimator;
+  std::unique_ptr<NoiseSuppressionImpl> noise_suppression;
+  std::unique_ptr<VoiceDetectionImpl> voice_detection;
+  std::unique_ptr<GainControlForExperimentalAgc>
       gain_control_for_experimental_agc;
 
   // Accessed internally from both render and capture.
-  rtc::scoped_ptr<TransientSuppressor> transient_suppressor;
-  rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer;
+  std::unique_ptr<TransientSuppressor> transient_suppressor;
+  std::unique_ptr<IntelligibilityEnhancer> intelligibility_enhancer;
 };
 
 struct AudioProcessingImpl::ApmPrivateSubmodules {
   explicit ApmPrivateSubmodules(Beamformer<float>* beamformer)
       : beamformer(beamformer) {}
   // Accessed internally from capture or during initialization
   std::list<ProcessingComponent*> component_list;
-  rtc::scoped_ptr<Beamformer<float>> beamformer;
-  rtc::scoped_ptr<AgcManagerDirect> agc_manager;
+  std::unique_ptr<Beamformer<float>> beamformer;
+  std::unique_ptr<AgcManagerDirect> agc_manager;
 };
 
 const int AudioProcessing::kNativeSampleRatesHz[] = {
     AudioProcessing::kSampleRate8kHz,
     AudioProcessing::kSampleRate16kHz,
     AudioProcessing::kSampleRate32kHz,
     AudioProcessing::kSampleRate48kHz};
 const size_t AudioProcessing::kNumNativeSampleRates =
     arraysize(AudioProcessing::kNativeSampleRatesHz);
 const int AudioProcessing::kMaxNativeSampleRateHz = AudioProcessing::
@@ skipping 168 matching lines @@
           ? formats_.rev_proc_format.num_frames()
           : formats_.api_format.reverse_output_stream().num_frames();
   if (formats_.api_format.reverse_input_stream().num_channels() > 0) {
     render_.render_audio.reset(new AudioBuffer(
         formats_.api_format.reverse_input_stream().num_frames(),
         formats_.api_format.reverse_input_stream().num_channels(),
         formats_.rev_proc_format.num_frames(),
         formats_.rev_proc_format.num_channels(),
         rev_audio_buffer_out_num_frames));
     if (rev_conversion_needed()) {
-      render_.render_converter = AudioConverter::Create(
+      render_.render_converter = rtc::ScopedToUnique(AudioConverter::Create(
           formats_.api_format.reverse_input_stream().num_channels(),
           formats_.api_format.reverse_input_stream().num_frames(),
           formats_.api_format.reverse_output_stream().num_channels(),
-          formats_.api_format.reverse_output_stream().num_frames());
+          formats_.api_format.reverse_output_stream().num_frames()));
     } else {
       render_.render_converter.reset(nullptr);
     }
   } else {
     render_.render_audio.reset(nullptr);
     render_.render_converter.reset(nullptr);
   }
   capture_.capture_audio.reset(
       new AudioBuffer(formats_.api_format.input_stream().num_frames(),
                       formats_.api_format.input_stream().num_channels(),
@@ skipping 1164 matching lines @@
   debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
 
   RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                         &debug_dump_.num_bytes_left_for_log_,
                                         &crit_debug_, &debug_dump_.capture));
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc