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

webrtc/video/vie_sync_module.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/video/vie_sync_module.h"
 
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/video_coding/include/video_coding.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/video/stream_synchronization.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 
 namespace webrtc {
 
 int UpdateMeasurements(StreamSynchronization::Measurements* stream,
                        const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) {
   if (!receiver.Timestamp(&stream->latest_timestamp))
     return -1;
   if (!receiver.LastReceivedTimeMs(&stream->latest_receive_time_ms))
@@ skipping 13 matching lines @@
   bool new_rtcp_sr = false;
   if (!UpdateRtcpList(
       ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) {
     return -1;
   }
 
   return 0;
 }
 
 ViESyncModule::ViESyncModule(VideoCodingModule* vcm)
-    : data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-      vcm_(vcm),
+    : vcm_(vcm),
       video_receiver_(NULL),
       video_rtp_rtcp_(NULL),
       voe_channel_id_(-1),
       voe_sync_interface_(NULL),
       last_sync_time_(TickTime::Now()),
       sync_() {
 }
 
 ViESyncModule::~ViESyncModule() {
 }
 
 int ViESyncModule::ConfigureSync(int voe_channel_id,
                                  VoEVideoSync* voe_sync_interface,
                                  RtpRtcp* video_rtcp_module,
                                  RtpReceiver* video_receiver) {
-  CriticalSectionScoped cs(data_cs_.get());
+  rtc::CritScope lock(&data_cs_);
   // Prevent expensive no-ops.
   if (voe_channel_id_ == voe_channel_id &&
       voe_sync_interface_ == voe_sync_interface &&
       video_receiver_ == video_receiver &&
       video_rtp_rtcp_ == video_rtcp_module) {
     return 0;
   }
   voe_channel_id_ = voe_channel_id;
   voe_sync_interface_ = voe_sync_interface;
   video_receiver_ = video_receiver;
@@ skipping 15 matching lines @@
 int ViESyncModule::VoiceChannel() {
   return voe_channel_id_;
 }
 
 int64_t ViESyncModule::TimeUntilNextProcess() {
   const int64_t kSyncIntervalMs = 1000;
   return kSyncIntervalMs - (TickTime::Now() - last_sync_time_).Milliseconds();
 }
 
 int32_t ViESyncModule::Process() {
-  CriticalSectionScoped cs(data_cs_.get());
+  rtc::CritScope lock(&data_cs_);
   last_sync_time_ = TickTime::Now();
 
   const int current_video_delay_ms = vcm_->Delay();
 
   if (voe_channel_id_ == -1) {
     return 0;
   }
   assert(video_rtp_rtcp_ && voe_sync_interface_);
   assert(sync_.get());
 
@@ skipping 49 matching lines @@
 
   if (voe_sync_interface_->SetMinimumPlayoutDelay(
       voe_channel_id_, target_audio_delay_ms) == -1) {
     LOG(LS_ERROR) << "Error setting voice delay.";
   }
   vcm_->SetMinimumPlayoutDelay(target_video_delay_ms);
   return 0;
 }
 
 }  // namespace webrtc