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

webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h

 /*
  *  Copyright (c) 2011 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 SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_
 #define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_
 
 #include <deque>
 
+#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/source/byte_io.h"
 #include "webrtc/system_wrappers/include/atomic32.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h"
 
 class TestErrorObserver;
 
 class LoopBackTransport : public webrtc::Transport {
  public:
   LoopBackTransport(webrtc::VoENetwork* voe_network, int channel)
-      : crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
-        packet_event_(webrtc::EventWrapper::Create()),
+      : packet_event_(webrtc::EventWrapper::Create()),
         thread_(NetworkProcess, this, "LoopBackTransport"),
         channel_(channel),
         voe_network_(voe_network),
         transmitted_packets_(0) {
     thread_.Start();
   }
 
   ~LoopBackTransport() { thread_.Stop(); }
 
   bool SendRtp(const uint8_t* data,
@@ skipping 12 matching lines @@
     enum {
       kSleepIntervalMs = 10
     };
     int32_t limit = transmitted_packets_.Value() + packet_count;
     while (transmitted_packets_.Value() < limit) {
       webrtc::SleepMs(kSleepIntervalMs);
     }
   }
 
   void AddChannel(uint32_t ssrc, int channel) {
-    webrtc::CriticalSectionScoped lock(crit_.get());
+    rtc::CritScope lock(&crit_);
     channels_[ssrc] = channel;
   }
 
  private:
   struct Packet {
     enum Type { Rtp, Rtcp, } type;
 
     Packet() : len(0) {}
     Packet(Type type, const void* data, size_t len)
         : type(type), len(len) {
       assert(len <= 1500);
       memcpy(this->data, data, len);
     }
 
     uint8_t data[1500];
     size_t len;
   };
 
   void StorePacket(Packet::Type type,
                    const void* data,
                    size_t len) {
     {
-      webrtc::CriticalSectionScoped lock(crit_.get());
+      rtc::CritScope lock(&crit_);
       packet_queue_.push_back(Packet(type, data, len));
     }
     packet_event_->Set();
   }
 
   static bool NetworkProcess(void* transport) {
     return static_cast<LoopBackTransport*>(transport)->SendPackets();
   }
 
   bool SendPackets() {
     switch (packet_event_->Wait(10)) {
       case webrtc::kEventSignaled:
         break;
       case webrtc::kEventTimeout:
         break;
       case webrtc::kEventError:
         // TODO(pbos): Log a warning here?
         return true;
     }
 
     while (true) {
       Packet p;
       int channel = channel_;
       {
-        webrtc::CriticalSectionScoped lock(crit_.get());
+        rtc::CritScope lock(&crit_);
         if (packet_queue_.empty())
           break;
         p = packet_queue_.front();
         packet_queue_.pop_front();
 
         if (p.type == Packet::Rtp) {
           uint32_t ssrc =
               webrtc::ByteReader<uint32_t>::ReadBigEndian(&p.data[8]);
           if (channels_[ssrc] != 0)
             channel = channels_[ssrc];
@@ skipping 12 matching lines @@
           break;
         case Packet::Rtcp:
           voe_network_->ReceivedRTCPPacket(channel, p.data, p.len);
           break;
       }
       ++transmitted_packets_;
     }
     return true;
   }
 
-  const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> crit_;
+  mutable rtc::CriticalSection crit_;
   const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_;
   rtc::PlatformThread thread_;
-  std::deque<Packet> packet_queue_ GUARDED_BY(crit_.get());
+  std::deque<Packet> packet_queue_ GUARDED_BY(crit_);
   const int channel_;
-  std::map<uint32_t, int> channels_ GUARDED_BY(crit_.get());
+  std::map<uint32_t, int> channels_ GUARDED_BY(crit_);
   webrtc::VoENetwork* const voe_network_;
   webrtc::Atomic32 transmitted_packets_;
 };
 
 // This fixture initializes the voice engine in addition to the work
 // done by the before-initialization fixture. It also registers an error
 // observer which will fail tests on error callbacks. This fixture is
 // useful to tests that want to run before we have started any form of
 // streaming through the voice engine.
 class AfterInitializationFixture : public BeforeInitializationFixture {
  public:
   AfterInitializationFixture();
   virtual ~AfterInitializationFixture();
 
  protected:
   rtc::scoped_ptr<TestErrorObserver> error_observer_;
 };
 
 #endif  // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_