Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/

webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc

 /*
  *  Copyright (c) 2013 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 <string.h>
 
+#include <memory>
+
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h"
 
 namespace webrtc {
 
 namespace acm2 {
 
 namespace {
 
 const uint8_t kAudioPayloadType = 0;
@@ skipping 48 matching lines @@
     InitRtpInfo(&rtp_info_);
   }
 
   void GetNextRtpHeader(WebRtcRTPHeader* rtp_info,
                         uint32_t* rtp_receive_timestamp) const {
     memcpy(rtp_info, &rtp_info_, sizeof(*rtp_info));
     *rtp_receive_timestamp = rtp_receive_timestamp_;
     NextRtpHeader(rtp_info, rtp_receive_timestamp);
   }
 
-  rtc::scoped_ptr<InitialDelayManager> manager_;
+  std::unique_ptr<InitialDelayManager> manager_;
   WebRtcRTPHeader rtp_info_;
   uint32_t rtp_receive_timestamp_;
 };
 
 TEST_F(InitialDelayManagerTest, Init) {
   EXPECT_TRUE(manager_->buffering());
   EXPECT_FALSE(manager_->PacketBuffered());
   manager_->DisableBuffering();
   EXPECT_FALSE(manager_->buffering());
   InitialDelayManager::SyncStream sync_stream;
@@ skipping 275 matching lines @@
   manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
                                      InitialDelayManager::kAudioPacket,
                                      false, kSamplingRateHz, &sync_stream);
   EXPECT_EQ(0, sync_stream.num_sync_packets);
   EXPECT_FALSE(manager_->buffering());
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc