Replace scoped_ptr with unique_ptr in webrtc/call/

webrtc/call/rtc_event_log_unittest.cc

 /*
  *  Copyright (c) 2015 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.
  */
 
 #ifdef ENABLE_RTC_EVENT_LOG
 
+#include <memory>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/random.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/call.h"
 #include "webrtc/call/rtc_event_log.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/test_suite.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
@@ skipping 434 matching lines @@
 
   // Find the name of the current test, in order to use it as a temporary
   // filename.
   auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
   const std::string temp_filename =
       test::OutputPath() + test_info->test_case_name() + test_info->name();
 
   // When log_dumper goes out of scope, it causes the log file to be flushed
   // to disk.
   {
-    rtc::scoped_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
+    std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
     log_dumper->LogVideoReceiveStreamConfig(receiver_config);
     log_dumper->LogVideoSendStreamConfig(sender_config);
     size_t rtcp_index = 1;
     size_t playout_index = 1;
     size_t bwe_loss_index = 1;
     for (size_t i = 1; i <= rtp_count; i++) {
       log_dumper->LogRtpHeader(
           (i % 2 == 0) ? kIncomingPacket : kOutgoingPacket,
           (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
           rtp_packets[i - 1].data(), rtp_packets[i - 1].size());
@@ skipping 145 matching lines @@
   GenerateVideoSendConfig(extensions_bitvector, &sender_config, &prng);
 
   // Find the name of the current test, in order to use it as a temporary
   // filename.
   auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
   const std::string temp_filename =
       test::OutputPath() + test_info->test_case_name() + test_info->name();
 
   // The log file will be flushed to disk when the log_dumper goes out of scope.
   {
-    rtc::scoped_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
+    std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
     // Reduce the time old events are stored to 50 ms.
     log_dumper->SetBufferDuration(50000);
     log_dumper->LogVideoReceiveStreamConfig(receiver_config);
     log_dumper->LogVideoSendStreamConfig(sender_config);
     log_dumper->LogRtpHeader(kOutgoingPacket, MediaType::AUDIO,
                              old_rtp_packet.data(), old_rtp_packet.size());
     log_dumper->LogRtcpPacket(kIncomingPacket, MediaType::AUDIO,
                               old_rtcp_packet.data(),
                               old_rtcp_packet.size());
     // Sleep 55 ms to let old events be removed from the queue.
@@ skipping 32 matching lines @@
   // Enable all header extensions
   uint32_t extensions = (1u << kNumExtensions) - 1;
   uint32_t csrcs_count = 2;
   DropOldEvents(extensions, csrcs_count, 141421356);
   DropOldEvents(extensions, csrcs_count, 173205080);
 }
 
 }  // namespace webrtc
 
 #endif  // ENABLE_RTC_EVENT_LOG