Using 64-bit timestamp in webrtc/p2p

webrtc/p2p/base/faketransportcontroller.h

 /*
  *  Copyright 2009 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.
  */
 
@@ skipping 187 matching lines @@
     if (flags != PF_SRTP_BYPASS && flags != 0) {
       return -1;
     }
 
     PacketMessageData* packet = new PacketMessageData(data, len);
     if (async_) {
       rtc::Thread::Current()->Post(this, 0, packet);
     } else {
       rtc::Thread::Current()->Send(this, 0, packet);
     }
-    rtc::SentPacket sent_packet(options.packet_id, rtc::Time());
+    rtc::SentPacket sent_packet(options.packet_id, rtc::Time64());

[honghaiz3, 2016/03/14 22:35:59]

SentPacket take int64_t as the second parameter, so we can use Time64() there
directly. 

     SignalSentPacket(this, sent_packet);
     return static_cast<int>(len);
   }
   int SetOption(rtc::Socket::Option opt, int value) override { return true; }
   bool GetOption(rtc::Socket::Option opt, int* value) override { return true; }
   int GetError() override { return 0; }
 
   void AddRemoteCandidate(const Candidate& candidate) override {
     remote_candidates_.push_back(candidate);
   }
@@ skipping 311 matching lines @@
     }
   }
 
  private:
   bool fail_create_channel_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_P2P_BASE_FAKETRANSPORTCONTROLLER_H_