Replace scoped_ptr with unique_ptr in webrtc/call/

webrtc/call/call.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 <map>
+#include <memory>
 #include <vector>
 
 #include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/scoped_voe_interface.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call.h"
 #include "webrtc/call/bitrate_allocator.h"
 #include "webrtc/call/rtc_event_log.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
@@ skipping 80 matching lines @@
     else
       return nullptr;
   }
 
   void UpdateSendHistograms() EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
   void UpdateReceiveHistograms();
 
   Clock* const clock_;
 
   const int num_cpu_cores_;
-  const rtc::scoped_ptr<ProcessThread> module_process_thread_;
-  const rtc::scoped_ptr<ProcessThread> pacer_thread_;
-  const rtc::scoped_ptr<CallStats> call_stats_;
-  const rtc::scoped_ptr<BitrateAllocator> bitrate_allocator_;
+  const std::unique_ptr<ProcessThread> module_process_thread_;
+  const std::unique_ptr<ProcessThread> pacer_thread_;
+  const std::unique_ptr<CallStats> call_stats_;
+  const std::unique_ptr<BitrateAllocator> bitrate_allocator_;
   Call::Config config_;
   rtc::ThreadChecker configuration_thread_checker_;
 
   bool network_enabled_;
 
-  rtc::scoped_ptr<RWLockWrapper> receive_crit_;
+  std::unique_ptr<RWLockWrapper> receive_crit_;
   // Audio and Video receive streams are owned by the client that creates them.
   std::map<uint32_t, AudioReceiveStream*> audio_receive_ssrcs_
       GUARDED_BY(receive_crit_);
   std::map<uint32_t, VideoReceiveStream*> video_receive_ssrcs_
       GUARDED_BY(receive_crit_);
   std::set<VideoReceiveStream*> video_receive_streams_
       GUARDED_BY(receive_crit_);
   std::map<std::string, AudioReceiveStream*> sync_stream_mapping_
       GUARDED_BY(receive_crit_);
 
-  rtc::scoped_ptr<RWLockWrapper> send_crit_;
+  std::unique_ptr<RWLockWrapper> send_crit_;
   // Audio and Video send streams are owned by the client that creates them.
   std::map<uint32_t, AudioSendStream*> audio_send_ssrcs_ GUARDED_BY(send_crit_);
   std::map<uint32_t, VideoSendStream*> video_send_ssrcs_ GUARDED_BY(send_crit_);
   std::set<VideoSendStream*> video_send_streams_ GUARDED_BY(send_crit_);
 
   VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;
 
   RtcEventLog* event_log_ = nullptr;
 
   // The following members are only accessed (exclusively) from one thread and
   // from the destructor, and therefore doesn't need any explicit
   // synchronization.
   int64_t received_video_bytes_;
   int64_t received_audio_bytes_;
   int64_t received_rtcp_bytes_;
   int64_t first_rtp_packet_received_ms_;
   int64_t last_rtp_packet_received_ms_;
   int64_t first_packet_sent_ms_;
 
   // TODO(holmer): Remove this lock once BitrateController no longer calls
   // OnNetworkChanged from multiple threads.
   rtc::CriticalSection bitrate_crit_;
   int64_t estimated_send_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);
   int64_t pacer_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);
   int64_t num_bitrate_updates_ GUARDED_BY(&bitrate_crit_);
 
   VieRemb remb_;
-  const rtc::scoped_ptr<CongestionController> congestion_controller_;
+  const std::unique_ptr<CongestionController> congestion_controller_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Call);
 };
 }  // namespace internal
 
 Call* Call::Create(const Call::Config& config) {
   return new internal::Call(config);
 }
 
 namespace internal {
 
 Call::Call(const Call::Config& config)
     : clock_(Clock::GetRealTimeClock()),
       num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
-      module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
-      pacer_thread_(ProcessThread::Create("PacerThread")),
+      module_process_thread_(
+          rtc::ScopedToUnique(ProcessThread::Create("ModuleProcessThread"))),
+      pacer_thread_(rtc::ScopedToUnique(ProcessThread::Create("PacerThread"))),
       call_stats_(new CallStats(clock_)),
       bitrate_allocator_(new BitrateAllocator()),
       config_(config),
       network_enabled_(true),
       receive_crit_(RWLockWrapper::CreateRWLock()),
       send_crit_(RWLockWrapper::CreateRWLock()),
       received_video_bytes_(0),
       received_audio_bytes_(0),
       received_rtcp_bytes_(0),
       first_rtp_packet_received_ms_(-1),
@@ skipping 546 matching lines @@
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc