Change default timestamp to 64 bits in all webrtc directories.

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

Index: webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
index 8fd745771128c70bd91067dd2e5fe781ced13a70..bbdf5015d4af3e0768ef638f81866b3496ad837a 100644
--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
@@ -108,7 +108,7 @@ class ConferenceTransport: public webrtc::Transport {
     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_);
@@ -116,7 +116,7 @@ class ConferenceTransport: public webrtc::Transport {
 
     uint8_t data_[kMaxPacketSizeByte];
     size_t len_;
-    uint32_t send_time_ms_;
+    int64_t send_time_ms_;
   };
 
   static bool Run(void* transport) {