Change default timestamp to 64 bits in all webrtc directories.

webrtc/voice_engine/test/auto_test/fakes/loudest_filter.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.
  */
 
 #include "webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h"
 
 #include "webrtc/base/checks.h"
 
 namespace voetest {
 
-void LoudestFilter::RemoveTimeoutStreams(uint32_t time_ms) {
+void LoudestFilter::RemoveTimeoutStreams(int64_t time_ms) {
   auto it = stream_levels_.begin();
   while (it != stream_levels_.end()) {
-    if (rtc::TimeDiff(time_ms, it->second.last_time_ms) >
-        kStreamTimeOutMs) {
+    if (rtc::TimeDiff(time_ms, it->second.last_time_ms) > kStreamTimeOutMs) {
       stream_levels_.erase(it++);
     } else {
       ++it;
     }
   }
 }
 
 unsigned int LoudestFilter::FindQuietestStream() {
   int quietest_level = kInvalidAudioLevel;
   unsigned int quietest_ssrc = 0;
   for (auto stream : stream_levels_) {
     // A smaller value if audio level corresponds to a louder sound.
     if (quietest_level == kInvalidAudioLevel ||
         stream.second.audio_level > quietest_level) {
       quietest_level = stream.second.audio_level;
       quietest_ssrc = stream.first;
     }
   }
   return quietest_ssrc;
 }
 
 bool LoudestFilter::ForwardThisPacket(const webrtc::RTPHeader& rtp_header) {
-  uint32_t time_now_ms = rtc::Time();
+  int64_t time_now_ms = rtc::TimeMillis();
   RemoveTimeoutStreams(time_now_ms);
 
   int source_ssrc = rtp_header.ssrc;
   int audio_level = rtp_header.extension.hasAudioLevel ?
       rtp_header.extension.audioLevel : kInvalidAudioLevel;
 
   if (audio_level == kInvalidAudioLevel) {
     // Always forward streams with unknown audio level, and don't keep their
     // states.
     return true;
@@ skipping 18 matching lines @@
   if (audio_level < stream_levels_[quietest_ssrc].audio_level) {
     stream_levels_.erase(quietest_ssrc);
     stream_levels_[source_ssrc].Set(audio_level, time_now_ms);
     return true;
   }
   return false;
 }
 
 }  // namespace voetest