Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/test/

webrtc/modules/audio_coding/test/EncodeDecodeTest.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.
  */
 
 #include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
 
+#include <memory>
 #include <sstream>
 #include <stdio.h>
 #include <stdlib.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
 #include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
 TestPacketization::TestPacketization(RTPStream *rtpStream, uint16_t frequency)
@@ skipping 239 matching lines @@
 void EncodeDecodeTest::Perform() {
   int numCodecs = 1;
   int codePars[3];  // Frequency, packet size, rate.
   int numPars[52];  // Number of codec parameters sets (freq, pacsize, rate)
                     // to test, for a given codec.
 
   codePars[0] = 0;
   codePars[1] = 0;
   codePars[2] = 0;
 
-  rtc::scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
   struct CodecInst sendCodecTmp;
   numCodecs = acm->NumberOfCodecs();
 
   if (_testMode != 2) {
     for (int n = 0; n < numCodecs; n++) {
       EXPECT_EQ(0, acm->Codec(n, &sendCodecTmp));
       if (STR_CASE_CMP(sendCodecTmp.plname, "telephone-event") == 0) {
         numPars[n] = 0;
       } else if (STR_CASE_CMP(sendCodecTmp.plname, "cn") == 0) {
         numPars[n] = 0;
@@ skipping 35 matching lines @@
   // End tracing.
   if (_testMode == 1) {
     Trace::ReturnTrace();
   }
 }
 
 std::string EncodeDecodeTest::EncodeToFile(int fileType,
                                            int codeId,
                                            int* codePars,
                                            int testMode) {
-  rtc::scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(1));
+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(1));
   RTPFile rtpFile;
   std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
                                                     "encode_decode_rtp");
   rtpFile.Open(fileName.c_str(), "wb+");
   rtpFile.WriteHeader();
 
   // Store for auto_test and logging.
   _sender.testMode = testMode;
   _sender.codeId = codeId;
 
   _sender.Setup(acm.get(), &rtpFile, "audio_coding/testfile32kHz", 32000, 1);
   if (acm->SendCodec()) {
     _sender.Run();
   }
   _sender.Teardown();
   rtpFile.Close();
 
   return fileName;
 }
 
 }  // namespace webrtc