Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

 /*
  *  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.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 
 #include <string.h>
 
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
 static const int kDtmfFrequencyHz = 8000;
 
 RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtpSender)
     : _clock(clock),
       _rtpSender(rtpSender),
-      _sendAudioCritsect(CriticalSectionWrapper::CreateCriticalSection()),
       _packetSizeSamples(160),
       _dtmfEventIsOn(false),
       _dtmfEventFirstPacketSent(false),
       _dtmfPayloadType(-1),
       _dtmfTimestamp(0),
       _dtmfKey(0),
       _dtmfLengthSamples(0),
       _dtmfLevel(0),
       _dtmfTimeLastSent(0),
       _dtmfTimestampLastSent(0),
       _REDPayloadType(-1),
       _inbandVADactive(false),
       _cngNBPayloadType(-1),
       _cngWBPayloadType(-1),
       _cngSWBPayloadType(-1),
       _cngFBPayloadType(-1),
       _lastPayloadType(-1),
       _audioLevel_dBov(0) {}
 
 RTPSenderAudio::~RTPSenderAudio() {}
 
 int RTPSenderAudio::AudioFrequency() const {
   return kDtmfFrequencyHz;
 }
 
 // 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;
 }
 
 int32_t RTPSenderAudio::RegisterAudioPayload(
     const char payloadName[RTP_PAYLOAD_NAME_SIZE],
     const int8_t payloadType,
     const uint32_t frequency,
     const size_t channels,
     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:
         _cngNBPayloadType = payloadType;
         break;
       case 16000:
         _cngWBPayloadType = payloadType;
         break;
       case 32000:
         _cngSWBPayloadType = payloadType;
         break;
       case 48000:
         _cngFBPayloadType = payloadType;
         break;
       default:
         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;
     return 0;
     // The default timestamp rate is 8000 Hz, but other rates may be defined.
   }
   *payload = new RtpUtility::Payload;
   (*payload)->typeSpecific.Audio.frequency = frequency;
   (*payload)->typeSpecific.Audio.channels = channels;
   (*payload)->typeSpecific.Audio.rate = rate;
   (*payload)->audio = true;
   (*payload)->name[RTP_PAYLOAD_NAME_SIZE - 1] = '\0';
   strncpy((*payload)->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
   return 0;
 }
 
 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) {
     if (payload_type != -1 && (_cngNBPayloadType == payload_type ||
                                _cngWBPayloadType == payload_type ||
                                _cngSWBPayloadType == payload_type ||
                                _cngFBPayloadType == payload_type)) {
       // Only set a marker bit when we change payload type to a non CNG
       return false;
     }
@@ skipping 37 matching lines @@
   // TODO(pwestin) Breakup function in smaller functions.
   size_t payloadSize = dataSize;
   size_t maxPayloadLength = _rtpSender->MaxPayloadLength();
   uint16_t dtmfLengthMS = 0;
   uint8_t key = 0;
   int red_payload_type;
   uint8_t audio_level_dbov;
   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;
     packet_size_samples = _packetSizeSamples;
   }
 
   // Check if we have pending DTMFs to send
   if (!_dtmfEventIsOn && PendingDTMF()) {
     int64_t delaySinceLastDTMF =
         _clock->TimeInMilliseconds() - _dtmfTimeLastSent;
@@ skipping 152 matching lines @@
              payloadData + fragmentation->fragmentationOffset[0],
              fragmentation->fragmentationLength[0]);
 
       payloadSize = fragmentation->fragmentationLength[0];
     } else {
       memcpy(dataBuffer + rtpHeaderLength, payloadData, payloadSize);
     }
   }
 
   {
-    CriticalSectionScoped cs(_sendAudioCritsect.get());
+    rtc::CritScope cs(&_sendAudioCritsect);
     _lastPayloadType = payloadType;
   }
   // Update audio level extension, if included.
   size_t packetSize = payloadSize + rtpHeaderLength;
   RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize);
   RTPHeader rtp_header;
   rtp_parser.Parse(&rtp_header);
   _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header,
                                (frameType == kAudioFrameSpeech),
                                audio_level_dbov);
   TRACE_EVENT_ASYNC_END2("webrtc", "Audio", captureTimeStamp, "timestamp",
                          _rtpSender->Timestamp(), "seqnum",
                          _rtpSender->SequenceNumber());
   int32_t send_result = _rtpSender->SendToNetwork(
       dataBuffer, payloadSize, rtpHeaderLength,
       TickTime::MillisecondTimestamp(), kAllowRetransmission,
       RtpPacketSender::kHighPriority);
   if (first_packet_sent_()) {
     LOG(LS_INFO) << "First audio RTP packet sent to pacer";
   }
   return send_result;
 }
 
 // Audio level magnitude and voice activity flag are set for each RTP packet
 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;
 }
 
 // Set payload type for Redundant Audio Data RFC 2198
 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;
   }
   *payloadType = _REDPayloadType;
   return 0;
 }
 
 // Send a TelephoneEvent tone using RFC 2833 (4733)
 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;
     }
   }
   return AddDTMF(key, time_ms, level);
 }
 
 int32_t RTPSenderAudio::SendTelephoneEventPacket(bool ended,
                                                  int8_t dtmf_payload_type,
@@ skipping 42 matching lines @@
                          dtmfTimeStamp, "seqnum", _rtpSender->SequenceNumber());
     retVal = _rtpSender->SendToNetwork(
         dtmfbuffer, 4, 12, TickTime::MillisecondTimestamp(),
         kAllowRetransmission, RtpPacketSender::kHighPriority);
     sendCount--;
   } while (sendCount > 0 && retVal == 0);
 
   return retVal;
 }
 }  // namespace webrtc