Change default timestamp to 64 bits in all webrtc directories.

webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h

 /*
  *  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 11 matching lines @@
   /* ForwardThisPacket()
    * Decide whether to forward a RTP packet, given its header.
    *
    * Input:
    *   rtp_header : Header of the RTP packet of interest.
    */
   bool ForwardThisPacket(const webrtc::RTPHeader& rtp_header);
 
  private:
   struct Status {
-    void Set(int audio_level, uint32_t last_time_ms) {
+    void Set(int audio_level, int64_t last_time_ms) {
       this->audio_level = audio_level;
       this->last_time_ms = last_time_ms;
     }
     int audio_level;
-    uint32_t last_time_ms;
+    int64_t last_time_ms;
   };
 
-  void RemoveTimeoutStreams(uint32_t time_ms);
+  void RemoveTimeoutStreams(int64_t time_ms);
   unsigned int FindQuietestStream();
 
   // Keeps the streams being forwarded in pair<SSRC, Status>.
   std::map<unsigned int, Status> stream_levels_;
 
   const int32_t kStreamTimeOutMs = 5000;
   const size_t kMaxMixSize = 3;
   const int kInvalidAudioLevel = 128;
 };
 
 
 }  // namespace voetest
 
 #endif  // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_