Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module

webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h

 /*
  *  Copyright (c) 2012 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_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
 
 #include <set>
 
 #include "webrtc/base/onetimeevent.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
-class CriticalSectionWrapper;
-
 // Handles audio RTP packets. This class is thread-safe.
 class RTPReceiverAudio : public RTPReceiverStrategy,
                          public TelephoneEventHandler {
  public:
   explicit RTPReceiverAudio(RtpData* data_callback);
   virtual ~RTPReceiverAudio() {}
 
   // The following three methods implement the TelephoneEventHandler interface.
   // Forward DTMFs to decoder for playout.
   void SetTelephoneEventForwardToDecoder(bool forward_to_decoder);
@@ skipping 80 matching lines @@
   bool last_received_g722_;
 
   uint8_t num_energy_;
   uint8_t current_remote_energy_[kRtpCsrcSize];
 
   ThreadUnsafeOneTimeEvent first_packet_received_;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_