Change default timestamp to 64 bits in all webrtc directories.

webrtc/api/dtmfsender_unittest.cc

 /*
  *  Copyright 2012 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 67 matching lines @@
     return (can_insert_dtmf_tracks_.count(track_label) != 0);
   }
 
   bool InsertDtmf(const std::string& track_label,
                   int code,
                   int duration) override {
     int gap = 0;
     // TODO(ronghuawu): Make the timer (basically the rtc::TimeNanos)
     // mockable and use a fake timer in the unit tests.
     if (last_insert_dtmf_call_ > 0) {
-      gap = static_cast<int>(rtc::Time() - last_insert_dtmf_call_);
+      gap = static_cast<int>(rtc::TimeMillis() - last_insert_dtmf_call_);
     }
-    last_insert_dtmf_call_ = rtc::Time();
+    last_insert_dtmf_call_ = rtc::TimeMillis();
 
     LOG(LS_VERBOSE) << "FakeDtmfProvider::InsertDtmf code=" << code
                     << " duration=" << duration
                     << " gap=" << gap << ".";
     dtmf_info_queue_.push_back(DtmfInfo(code, duration, gap));
     return true;
   }
 
   sigslot::signal0<>* GetOnDestroyedSignal() override {
     return &SignalDestroyed;
@@ skipping 233 matching lines @@
   std::string tones = "3,4";
   int duration = 100;
   int inter_tone_gap = 50;
 
   EXPECT_FALSE(dtmf_->InsertDtmf(tones, 6001, inter_tone_gap));
   EXPECT_FALSE(dtmf_->InsertDtmf(tones, 69, inter_tone_gap));
   EXPECT_FALSE(dtmf_->InsertDtmf(tones, duration, 49));
 
   EXPECT_TRUE(dtmf_->InsertDtmf(tones, duration, inter_tone_gap));
 }