Swap use of CriticalSectionWrapper for rtc::CriticalSection in webrtc/video.

webrtc/video/payload_router.cc

 /*
  *  Copyright (c) 2015 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/video/payload_router.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
 PayloadRouter::PayloadRouter()
-    : crit_(CriticalSectionWrapper::CreateCriticalSection()),
-      active_(false) {}
+    : active_(false) {}
 
 PayloadRouter::~PayloadRouter() {}
 
 size_t PayloadRouter::DefaultMaxPayloadLength() {
   const size_t kIpUdpSrtpLength = 44;
   return IP_PACKET_SIZE - kIpUdpSrtpLength;
 }
 
 void PayloadRouter::SetSendingRtpModules(
     const std::list<RtpRtcp*>& rtp_modules) {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope lock(&crit_);
   rtp_modules_.clear();
   rtp_modules_.reserve(rtp_modules.size());
   for (auto* rtp_module : rtp_modules) {
     rtp_modules_.push_back(rtp_module);
   }
 }
 
 void PayloadRouter::set_active(bool active) {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope lock(&crit_);
   active_ = active;
 }
 
 bool PayloadRouter::active() {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope lock(&crit_);
   return active_ && !rtp_modules_.empty();
 }
 
 bool PayloadRouter::RoutePayload(FrameType frame_type,
                                  int8_t payload_type,
                                  uint32_t time_stamp,
                                  int64_t capture_time_ms,
                                  const uint8_t* payload_data,
                                  size_t payload_length,
                                  const RTPFragmentationHeader* fragmentation,
                                  const RTPVideoHeader* rtp_video_hdr) {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope lock(&crit_);
   if (!active_ || rtp_modules_.empty())
     return false;
 
   // The simulcast index might actually be larger than the number of modules in
   // case the encoder was processing a frame during a codec reconfig.
   if (rtp_video_hdr != NULL &&
       rtp_video_hdr->simulcastIdx >= rtp_modules_.size())
     return false;
 
   int stream_idx = 0;
   if (rtp_video_hdr != NULL)
     stream_idx = rtp_video_hdr->simulcastIdx;
   return rtp_modules_[stream_idx]->SendOutgoingData(
       frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
       payload_length, fragmentation, rtp_video_hdr) == 0 ? true : false;
 }
 
 void PayloadRouter::SetTargetSendBitrates(
     const std::vector<uint32_t>& stream_bitrates) {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope lock(&crit_);
   if (stream_bitrates.size() < rtp_modules_.size()) {
     // There can be a size mis-match during codec reconfiguration.
     return;
   }
   int idx = 0;
   for (auto* rtp_module : rtp_modules_) {
     rtp_module->SetTargetSendBitrate(stream_bitrates[idx++]);
   }
 }
 
 size_t PayloadRouter::MaxPayloadLength() const {
   size_t min_payload_length = DefaultMaxPayloadLength();
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope lock(&crit_);
   for (auto* rtp_module : rtp_modules_) {
     size_t module_payload_length = rtp_module->MaxDataPayloadLength();
     if (module_payload_length < min_payload_length)
       min_payload_length = module_payload_length;
   }
   return min_payload_length;
 }
 
 }  // namespace webrtc