Swap use of CriticalSectionWrapper with rtc::CriticalSection in voice_engine/

webrtc/voice_engine/test/auto_test/fakes/conference_transport.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
 #define WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
 
 #include <deque>
 #include <map>
 #include <utility>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 #include "webrtc/voice_engine/include/voe_file.h"
 #include "webrtc/voice_engine/include/voe_network.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h"
 
 static const size_t kMaxPacketSizeByte = 1500;
 
@@ skipping 86 matching lines @@
 
   static bool Run(void* transport) {
     return static_cast<ConferenceTransport*>(transport)->DispatchPackets();
   }
 
   int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const;
   void StorePacket(Packet::Type type, const void* data, size_t len);
   void SendPacket(const Packet& packet);
   bool DispatchPackets();
 
-  const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> pq_crit_;
-  const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> stream_crit_;
+  mutable rtc::CriticalSection pq_crit_;
+  mutable rtc::CriticalSection stream_crit_;
   const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_;
   rtc::PlatformThread thread_;
 
   unsigned int rtt_ms_;
   unsigned int stream_count_;
 
-  std::map<unsigned int, std::pair<int, int>> streams_
-    GUARDED_BY(stream_crit_.get());
-  std::deque<Packet> packet_queue_ GUARDED_BY(pq_crit_.get());
+  std::map<unsigned int, std::pair<int, int>> streams_ GUARDED_BY(stream_crit_);
+  std::deque<Packet> packet_queue_ GUARDED_BY(pq_crit_);
 
   int local_sender_;  // Channel Id of local sender
   int reflector_;
 
   webrtc::VoiceEngine* local_voe_;
   webrtc::VoEBase* local_base_;
   webrtc::VoERTP_RTCP* local_rtp_rtcp_;
   webrtc::VoENetwork* local_network_;
 
   webrtc::VoiceEngine* remote_voe_;
   webrtc::VoEBase* remote_base_;
   webrtc::VoECodec* remote_codec_;
   webrtc::VoERTP_RTCP* remote_rtp_rtcp_;
   webrtc::VoENetwork* remote_network_;
   webrtc::VoEFile* remote_file_;
 
   LoudestFilter loudest_filter_;
 
   const rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
 };
 }  // namespace voetest
 
 #endif  // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_