Change default timestamp to 64 bits in all webrtc directories.

webrtc/base/asynctcpsocket.cc

 /*
  *  Copyright 2004 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 278 matching lines @@
   AppendToOutBuffer(&pkt_len, kPacketLenSize);
   AppendToOutBuffer(pv, cb);
 
   int res = FlushOutBuffer();
   if (res <= 0) {
     // drop packet if we made no progress
     ClearOutBuffer();
     return res;
   }
 
-  rtc::SentPacket sent_packet(options.packet_id, rtc::Time());
+  rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis());
   SignalSentPacket(this, sent_packet);
 
   // We claim to have sent the whole thing, even if we only sent partial
   return static_cast<int>(cb);
 }
 
 void AsyncTCPSocket::ProcessInput(char * data, size_t* len) {
   SocketAddress remote_addr(GetRemoteAddress());
 
   while (true) {
@@ skipping 12 matching lines @@
       memmove(data, data + kPacketLenSize + pkt_len, *len);
     }
   }
 }
 
 void AsyncTCPSocket::HandleIncomingConnection(AsyncSocket* socket) {
   SignalNewConnection(this, new AsyncTCPSocket(socket, false));
 }
 
 }  // namespace rtc