Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index c85a19781dab9f36a0a037491292c528982449d6..6d0f7a4627bc49cad1680b11f34fb26090b9ff1e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -25,7 +25,6 @@ static const int kDtmfFrequencyHz = 8000;
 RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtpSender)
     : _clock(clock),
       _rtpSender(rtpSender),
-      _sendAudioCritsect(CriticalSectionWrapper::CreateCriticalSection()),
       _packetSizeSamples(160),
       _dtmfEventIsOn(false),
       _dtmfEventFirstPacketSent(false),
@@ -54,7 +53,7 @@ int RTPSenderAudio::AudioFrequency() const {
 // set audio packet size, used to determine when it's time to send a DTMF packet
 // in silence (CNG)
 int32_t RTPSenderAudio::SetAudioPacketSize(uint16_t packetSizeSamples) {
-  CriticalSectionScoped cs(_sendAudioCritsect.get());
+  rtc::CritScope cs(&_sendAudioCritsect);
 
   _packetSizeSamples = packetSizeSamples;
   return 0;
@@ -68,7 +67,7 @@ int32_t RTPSenderAudio::RegisterAudioPayload(
     const uint32_t rate,
     RtpUtility::Payload** payload) {
   if (RtpUtility::StringCompare(payloadName, "cn", 2)) {
-    CriticalSectionScoped cs(_sendAudioCritsect.get());
+    rtc::CritScope cs(&_sendAudioCritsect);
     //  we can have multiple CNG payload types
     switch (frequency) {
       case 8000:
@@ -87,7 +86,7 @@ int32_t RTPSenderAudio::RegisterAudioPayload(
         return -1;
     }
   } else if (RtpUtility::StringCompare(payloadName, "telephone-event", 15)) {
-    CriticalSectionScoped cs(_sendAudioCritsect.get());
+    rtc::CritScope cs(&_sendAudioCritsect);
     // Don't add it to the list
     // we dont want to allow send with a DTMF payloadtype
     _dtmfPayloadType = payloadType;
@@ -105,7 +104,7 @@ int32_t RTPSenderAudio::RegisterAudioPayload(
 }
 
 bool RTPSenderAudio::MarkerBit(FrameType frameType, int8_t payload_type) {
-  CriticalSectionScoped cs(_sendAudioCritsect.get());
+  rtc::CritScope cs(&_sendAudioCritsect);
   // for audio true for first packet in a speech burst
   bool markerBit = false;
   if (_lastPayloadType != payload_type) {
@@ -163,7 +162,7 @@ int32_t RTPSenderAudio::SendAudio(FrameType frameType,
   int8_t dtmf_payload_type;
   uint16_t packet_size_samples;
   {
-    CriticalSectionScoped cs(_sendAudioCritsect.get());
+    rtc::CritScope cs(&_sendAudioCritsect);
     red_payload_type = _REDPayloadType;
     audio_level_dbov = _audioLevel_dBov;
     dtmf_payload_type = _dtmfPayloadType;
@@ -336,7 +335,7 @@ int32_t RTPSenderAudio::SendAudio(FrameType frameType,
   }
 
   {
-    CriticalSectionScoped cs(_sendAudioCritsect.get());
+    rtc::CritScope cs(&_sendAudioCritsect);
     _lastPayloadType = payloadType;
   }
   // Update audio level extension, if included.
@@ -365,7 +364,7 @@ int32_t RTPSenderAudio::SetAudioLevel(uint8_t level_dBov) {
   if (level_dBov > 127) {
     return -1;
   }
-  CriticalSectionScoped cs(_sendAudioCritsect.get());
+  rtc::CritScope cs(&_sendAudioCritsect);
   _audioLevel_dBov = level_dBov;
   return 0;
 }
@@ -375,14 +374,14 @@ int32_t RTPSenderAudio::SetRED(int8_t payloadType) {
   if (payloadType < -1) {
     return -1;
   }
-  CriticalSectionScoped cs(_sendAudioCritsect.get());
+  rtc::CritScope cs(&_sendAudioCritsect);
   _REDPayloadType = payloadType;
   return 0;
 }
 
 // Get payload type for Redundant Audio Data RFC 2198
 int32_t RTPSenderAudio::RED(int8_t* payloadType) const {
-  CriticalSectionScoped cs(_sendAudioCritsect.get());
+  rtc::CritScope cs(&_sendAudioCritsect);
   if (_REDPayloadType == -1) {
     // not configured
     return -1;
@@ -396,7 +395,7 @@ int32_t RTPSenderAudio::SendTelephoneEvent(uint8_t key,
                                            uint16_t time_ms,
                                            uint8_t level) {
   {
-    CriticalSectionScoped lock(_sendAudioCritsect.get());
+    rtc::CritScope lock(&_sendAudioCritsect);
     if (_dtmfPayloadType < 0) {
       // TelephoneEvent payloadtype not configured
       return -1;