Replace scoped_ptr with unique_ptr in webrtc/common_audio/

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 277 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 = rtc::ScopedToUnique(AudioConverter::Create(
+      render_.render_converter = 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 1137 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