Change default timestamp to 64 bits in all webrtc directories.

webrtc/voice_engine/test/auto_test/fakes/conference_transport.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.
  */
 
@@ skipping 113 matching lines @@
     return it->second.second;
   }
   return -1;
 }
 
 void ConferenceTransport::StorePacket(Packet::Type type,
                                       const void* data,
                                       size_t len) {
   {
     rtc::CritScope lock(&pq_crit_);
-    packet_queue_.push_back(Packet(type, data, len, rtc::Time()));
+    packet_queue_.push_back(Packet(type, data, len, rtc::TimeMillis()));
   }
   packet_event_->Set();
 }
 
 // This simulates the flow of RTP and RTCP packets. Complications like that
 // a packet is first sent to the reflector, and then forwarded to the receiver
 // are simplified, in this particular case, to a direct link between the sender
 // and the receiver.
 void ConferenceTransport::SendPacket(const Packet& packet) {
   int destination = -1;
@@ skipping 132 matching lines @@
 bool ConferenceTransport::GetReceiverStatistics(unsigned int id,
                                                 webrtc::CallStatistics* stats) {
   int dst = GetReceiverChannelForSsrc(id);
   if (dst == -1) {
     return false;
   }
   EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats));
   return true;
 }
 }  // namespace voetest