Replace Atomic32 with webrtc/base/atomicops.h.

webrtc/video_engine/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_engine/payload_router.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 
 namespace webrtc {
 
 PayloadRouter::PayloadRouter()
     : crit_(CriticalSectionWrapper::CreateCriticalSection()),
-      active_(false) {}
+      active_(false),
+      ref_count_(0) {}
 
 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) {
@@ skipping 59 matching lines @@
   CriticalSectionScoped cs(crit_.get());
   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