Change default timestamp to 64 bits in all webrtc directories.

webrtc/voice_engine/test/auto_test/fakes/conference_transport.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 90 matching lines @@
   bool SendRtp(const uint8_t* data,
                size_t len,
                const webrtc::PacketOptions& options) override;
   bool SendRtcp(const uint8_t *data, size_t len) override;
 
  private:
   struct Packet {
     enum Type { Rtp, Rtcp, } type_;
 
     Packet() : len_(0) {}
-    Packet(Type type, const void* data, size_t len, uint32_t time_ms)
+    Packet(Type type, const void* data, size_t len, int64_t time_ms)
         : type_(type), len_(len), send_time_ms_(time_ms) {
       EXPECT_LE(len_, kMaxPacketSizeByte);
       memcpy(data_, data, len_);
     }
 
     uint8_t data_[kMaxPacketSizeByte];
     size_t len_;
-    uint32_t send_time_ms_;
+    int64_t send_time_ms_;
   };
 
   static bool Run(void* transport) {
     return static_cast<ConferenceTransport*>(transport)->DispatchPackets();
   }
 
   int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const;
   void StorePacket(Packet::Type type, const void* data, size_t len);
   void SendPacket(const Packet& packet);
   bool DispatchPackets();
@@ skipping 24 matching lines @@
   webrtc::VoENetwork* remote_network_;
   webrtc::VoEFile* remote_file_;
 
   LoudestFilter loudest_filter_;
 
   const std::unique_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
 };
 }  // namespace voetest
 
 #endif  // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_