Create rtc::AtomicInt POD struct.

webrtc/modules/pacing/packet_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/modules/pacing/packet_router.h"
 
 #include "webrtc/base/atomicops.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/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 
 namespace webrtc {
 
-PacketRouter::PacketRouter() : transport_seq_(0) {
-}
+PacketRouter::PacketRouter() {}
 
 PacketRouter::~PacketRouter() {
   RTC_DCHECK(rtp_modules_.empty());
 }
 
 void PacketRouter::AddRtpModule(RtpRtcp* rtp_module) {
   rtc::CritScope cs(&modules_lock_);
   RTC_DCHECK(std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module) ==
              rtp_modules_.end());
   rtp_modules_.push_back(rtp_module);
@@ skipping 29 matching lines @@
           module->TimeToSendPadding(bytes_to_send - total_bytes_sent);
       total_bytes_sent += bytes_sent;
       if (total_bytes_sent >= bytes_to_send)
         break;
     }
   }
   return total_bytes_sent;
 }
 
 void PacketRouter::SetTransportWideSequenceNumber(uint16_t sequence_number) {
-  rtc::AtomicOps::ReleaseStore(&transport_seq_, sequence_number);
+  rtc::AtomicInt::ReleaseStore(&transport_seq_, sequence_number);
 }
 
 uint16_t PacketRouter::AllocateSequenceNumber() {
-  int prev_seq = rtc::AtomicOps::AcquireLoad(&transport_seq_);
+  int prev_seq = rtc::AtomicInt::AcquireLoad(&transport_seq_);
   int desired_prev_seq;
   int new_seq;
   do {
     desired_prev_seq = prev_seq;
     new_seq = (desired_prev_seq + 1) & 0xFFFF;
     // Note: CompareAndSwap returns the actual value of transport_seq at the
     // time the CAS operation was executed. Thus, if prev_seq is returned, the
     // operation was successful - otherwise we need to retry. Saving the
     // return value saves us a load on retry.
-    prev_seq = rtc::AtomicOps::CompareAndSwap(&transport_seq_, desired_prev_seq,
+    prev_seq = rtc::AtomicInt::CompareAndSwap(&transport_seq_, desired_prev_seq,
                                               new_seq);
   } while (prev_seq != desired_prev_seq);
 
   return new_seq;
 }
 
 bool PacketRouter::SendFeedback(rtcp::TransportFeedback* packet) {
   rtc::CritScope cs(&modules_lock_);
   for (auto* rtp_module : rtp_modules_) {
     packet->WithPacketSenderSsrc(rtp_module->SSRC());
     if (rtp_module->SendFeedbackPacket(*packet))
       return true;
   }
   return false;
 }
 
 }  // namespace webrtc