Replace scoped_ptr with unique_ptr in webrtc/p2p/

webrtc/p2p/stunprober/stunprober.cc

 /*
  *  Copyright 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 <map>
+#include <memory>
 #include <set>
 #include <string>
 
 #include "webrtc/base/asyncpacketsocket.h"
 #include "webrtc/base/asyncresolverinterface.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
@@ skipping 61 matching lines @@
   bool Done() {
     return static_cast<size_t>(num_request_sent_) == server_ips_.size();
   }
 
  private:
   Request* GetRequestByAddress(const rtc::IPAddress& ip);
 
   StunProber* prober_;
 
   // The socket for this session.
-  rtc::scoped_ptr<rtc::AsyncPacketSocket> socket_;
+  std::unique_ptr<rtc::AsyncPacketSocket> socket_;
 
   // Temporary SocketAddress and buffer for RecvFrom.
   rtc::SocketAddress addr_;
-  rtc::scoped_ptr<rtc::ByteBufferWriter> response_packet_;
+  std::unique_ptr<rtc::ByteBufferWriter> response_packet_;
 
   std::vector<Request*> requests_;
   std::vector<rtc::SocketAddress> server_ips_;
   int16_t num_request_sent_ = 0;
   int16_t num_response_received_ = 0;
 
   rtc::ThreadChecker& thread_checker_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Requester);
 };
@@ skipping 26 matching lines @@
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   requests_.push_back(new Request());
   Request& request = *(requests_.back());
   cricket::StunMessage message;
 
   // Random transaction ID, STUN_BINDING_REQUEST
   message.SetTransactionID(
       rtc::CreateRandomString(cricket::kStunTransactionIdLength));
   message.SetType(cricket::STUN_BINDING_REQUEST);
 
-  rtc::scoped_ptr<rtc::ByteBufferWriter> request_packet(
+  std::unique_ptr<rtc::ByteBufferWriter> request_packet(
       new rtc::ByteBufferWriter(nullptr, kMaxUdpBufferSize));
   if (!message.Write(request_packet.get())) {
     prober_->ReportOnFinished(WRITE_FAILED);
     return;
   }
 
   auto addr = server_ips_[num_request_sent_];
   request.server_addr = addr.ipaddr();
 
   // The write must succeed immediately. Otherwise, the calculating of the STUN
@@ skipping 183 matching lines @@
     return;
   }
 
   // Dedupe.
   std::set<rtc::SocketAddress> addrs(all_servers_addrs_.begin(),
                                      all_servers_addrs_.end());
   all_servers_addrs_.assign(addrs.begin(), addrs.end());
 
   // Prepare all the sockets beforehand. All of them will bind to "any" address.
   while (sockets_.size() < total_socket_required()) {
-    rtc::scoped_ptr<rtc::AsyncPacketSocket> socket(
+    std::unique_ptr<rtc::AsyncPacketSocket> socket(
         socket_factory_->CreateUdpSocket(rtc::SocketAddress(INADDR_ANY, 0), 0,
                                          0));
     if (!socket) {
       ReportOnPrepared(GENERIC_FAILURE);
       return;
     }
     // Chrome and WebRTC behave differently in terms of the state of a socket
     // once returned from PacketSocketFactory::CreateUdpSocket.
     if (socket->GetState() == rtc::AsyncPacketSocket::STATE_BINDING) {
       socket->SignalAddressReady.connect(this, &StunProber::OnSocketReady);
@@ skipping 203 matching lines @@
   }
 }
 
 void StunProber::ReportOnFinished(StunProber::Status status) {
   if (observer_) {
     observer_->OnFinished(this, status);
   }
 }
 
 }  // namespace stunprober