Replace scoped_ptr with unique_ptr in webrtc/video/

webrtc/video/vie_remb_unittest.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.
  */
 
 
 // This file includes unit tests for ViERemb.
 
+#include <memory>
 #include <vector>
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "webrtc/modules/utility/include/mock/mock_process_thread.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/video/vie_remb.h"
 
 using ::testing::_;
 using ::testing::AnyNumber;
 using ::testing::NiceMock;
 using ::testing::Return;
 
 namespace webrtc {
 
 class ViERembTest : public ::testing::Test {
  public:
   ViERembTest() : fake_clock_(12345) {}
 
  protected:
   virtual void SetUp() {
     process_thread_.reset(new NiceMock<MockProcessThread>);
     vie_remb_.reset(new VieRemb(&fake_clock_));
   }
   SimulatedClock fake_clock_;
-  rtc::scoped_ptr<MockProcessThread> process_thread_;
-  rtc::scoped_ptr<VieRemb> vie_remb_;
+  std::unique_ptr<MockProcessThread> process_thread_;
+  std::unique_ptr<VieRemb> vie_remb_;
 };
 
 TEST_F(ViERembTest, OneModuleTestForSendingRemb) {
   MockRtpRtcp rtp;
   vie_remb_->AddReceiveChannel(&rtp);
   vie_remb_->AddRembSender(&rtp);
 
   const uint32_t bitrate_estimate = 456;
   uint32_t ssrc = 1234;
   std::vector<uint32_t> ssrcs(&ssrc, &ssrc + 1);
@@ skipping 190 matching lines @@
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
 
   // Lower the estimate to trigger a new packet REMB packet.
   bitrate_estimate = bitrate_estimate - 100;
   EXPECT_CALL(rtp, SetREMBData(_, _))
       .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
 }
 
 }  // namespace webrtc