Change default timestamp to 64 bits in all webrtc directories.

webrtc/voice_engine/test/auto_test/voe_conference_test.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 61 matching lines @@
   const int kStatsRequestIntervalMs = 1000;
   const int kStatsBufferSize = 3;
 
   uint32_t deadline = rtc::TimeAfter(kMaxRunTimeMs);
   // Run the following up to |kMaxRunTimeMs| milliseconds.
   int successive_pass = 0;
   webrtc::CallStatistics stats_1;
   webrtc::CallStatistics stats_2;
   std::queue<Stats> stats_buffer;
 
-  while (rtc::TimeIsLater(rtc::Time(), deadline) &&
-      successive_pass < kNeedSuccessivePass) {
+  while (rtc::Time() < deadline && successive_pass < kNeedSuccessivePass) {
     webrtc::SleepMs(kStatsRequestIntervalMs);
 
     EXPECT_TRUE(trans.GetReceiverStatistics(id_1, &stats_1));
     EXPECT_TRUE(trans.GetReceiverStatistics(id_2, &stats_2));
 
     // It is not easy to verify the NTP time directly. We verify it by testing
     // the difference of two start NTP times.
     int64_t captured_start_ntp_delay = stats_2.capture_start_ntp_time_ms_ -
         stats_1.capture_start_ntp_time_ms_;
 
@@ skipping 73 matching lines @@
     // depends on the sleep timer. So we allow a small off from |kPackets|.
     EXPECT_NEAR(stats_1.packetsReceived, kPackets, 2);
     EXPECT_NEAR(stats_2.packetsReceived, kPackets, 2);
     EXPECT_NEAR(stats_3.packetsReceived, kPackets, 2);
   }
 
   remove(silence_file.c_str());
 }
 
 }  // namespace voetest