Replace scoped_ptr with unique_ptr in webrtc/call/

webrtc/call/bitrate_estimator_tests.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 <functional>
 #include <list>
+#include <memory>
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/audio_state.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/call.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/fake_decoder.h"
 #include "webrtc/test/fake_encoder.h"
 #include "webrtc/test/mock_voice_engine.h"
@@ skipping 205 matching lines @@
         is_sending_receiving_ = false;
       }
     }
 
    private:
     BitrateEstimatorTest* test_;
     bool is_sending_receiving_;
     VideoSendStream* send_stream_;
     AudioReceiveStream* audio_receive_stream_;
     VideoReceiveStream* video_receive_stream_;
-    rtc::scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
+    std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
     test::FakeEncoder fake_encoder_;
     test::FakeDecoder fake_decoder_;
   };
 
   testing::NiceMock<test::MockVoiceEngine> mock_voice_engine_;
   LogObserver receiver_log_;
-  rtc::scoped_ptr<test::DirectTransport> send_transport_;
-  rtc::scoped_ptr<test::DirectTransport> receive_transport_;
-  rtc::scoped_ptr<Call> sender_call_;
-  rtc::scoped_ptr<Call> receiver_call_;
+  std::unique_ptr<test::DirectTransport> send_transport_;
+  std::unique_ptr<test::DirectTransport> receive_transport_;
+  std::unique_ptr<Call> sender_call_;
+  std::unique_ptr<Call> receiver_call_;
   VideoReceiveStream::Config receive_config_;
   std::vector<Stream*> streams_;
 };
 
 static const char* kAbsSendTimeLog =
     "RemoteBitrateEstimatorAbsSendTime: Instantiating.";
 static const char* kSingleStreamLog =
     "RemoteBitrateEstimatorSingleStream: Instantiating.";
 
 TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
@@ skipping 55 matching lines @@
       RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
   streams_.push_back(new Stream(this, false));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_TRUE(receiver_log_.Wait());
 }
 }  // namespace webrtc