Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module

webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
index 9de65abdbda8c54b6c11df136ba47fbc4562e35f..38b2830b79cbe9c3e903044b8c3b2eb746d49fdb 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
@@ -16,7 +16,6 @@
 
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy(
@@ -46,26 +45,26 @@ RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback)
 // Outband TelephoneEvent(DTMF) detection
 void RTPReceiverAudio::SetTelephoneEventForwardToDecoder(
     bool forward_to_decoder) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   telephone_event_forward_to_decoder_ = forward_to_decoder;
 }
 
 // Is forwarding of outband telephone events turned on/off?
 bool RTPReceiverAudio::TelephoneEventForwardToDecoder() const {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   return telephone_event_forward_to_decoder_;
 }
 
 bool RTPReceiverAudio::TelephoneEventPayloadType(
     int8_t payload_type) const {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   return telephone_event_payload_type_ == payload_type;
 }
 
 bool RTPReceiverAudio::CNGPayloadType(int8_t payload_type,
                                       uint32_t* frequency,
                                       bool* cng_payload_type_has_changed) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   *cng_payload_type_has_changed = false;
 
   //  We can have four CNG on 8000Hz, 16000Hz, 32000Hz and 48000Hz.
@@ -152,7 +151,7 @@ int32_t RTPReceiverAudio::OnNewPayloadTypeCreated(
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     int8_t payload_type,
     uint32_t frequency) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
 
   if (RtpUtility::StringCompare(payload_name, "telephone-event", 15)) {
     telephone_event_payload_type_ = payload_type;
@@ -206,7 +205,7 @@ int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
 }
 
 int RTPReceiverAudio::GetPayloadTypeFrequency() const {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   if (last_received_g722_) {
     return 8000;
   }
@@ -249,7 +248,7 @@ void RTPReceiverAudio::CheckPayloadChanged(int8_t payload_type,
 }
 
 int RTPReceiverAudio::Energy(uint8_t array_of_energy[kRtpCsrcSize]) const {
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
 
   assert(num_energy_ <= kRtpCsrcSize);
 
@@ -291,7 +290,7 @@ int32_t RTPReceiverAudio::ParseAudioCodecSpecific(
   bool telephone_event_packet =
       TelephoneEventPayloadType(rtp_header->header.payloadType);
   if (telephone_event_packet) {
-    CriticalSectionScoped lock(crit_sect_.get());
+    rtc::CritScope lock(&crit_sect_);
 
     // RFC 4733 2.3
     // 0                   1                   2                   3
@@ -336,7 +335,7 @@ int32_t RTPReceiverAudio::ParseAudioCodecSpecific(
   }
 
   {
-    CriticalSectionScoped lock(crit_sect_.get());
+    rtc::CritScope lock(&crit_sect_);
 
     if (!telephone_event_packet) {
       last_received_frequency_ = audio_specific.frequency;