Change default timestamp to 64 bits in all webrtc directories.

webrtc/media/base/rtpdump.cc

Index: webrtc/media/base/rtpdump.cc
diff --git a/webrtc/media/base/rtpdump.cc b/webrtc/media/base/rtpdump.cc
index a109f2d8e269bcfd135794a0c7847d2b8e51d79a..6ffa311cb2eb08059dcd8c2459dbc850be33e9a9 100644
--- a/webrtc/media/base/rtpdump.cc
+++ b/webrtc/media/base/rtpdump.cc
@@ -28,13 +28,12 @@ namespace cricket {
 
 const char RtpDumpFileHeader::kFirstLine[] = "#!rtpplay1.0 0.0.0.0/0\n";
 
-RtpDumpFileHeader::RtpDumpFileHeader(uint32_t start_ms, uint32_t s, uint16_t p)
-    : start_sec(start_ms / 1000),
-      start_usec(start_ms % 1000 * 1000),
+RtpDumpFileHeader::RtpDumpFileHeader(int64_t start_ms, uint32_t s, uint16_t p)
+    : start_sec(static_cast<uint32_t>(start_ms / 1000)),

[Taylor Brandstetter, 2016/04/05 01:08:15]

Why a uint32_t?

[pthatcher1, 2016/04/11 20:56:59]

I agree.  Why not just change to uint64_t everywhere here?

[honghaiz3, 2016/04/18 23:39:04]

RtpDumpFileHeader defines the format of the RtpDumpFile header and the output
may be used elsewhere. If we change the structure here, it may cause issue when
other apps (like wireshark) try to read the dumped file. 
So I think we should avoid changing it. 

+      start_usec(static_cast<uint32_t>(start_ms % 1000 * 1000)),
       source(s),
       port(p),
-      padding(0) {
-}
+      padding(0) {}
 
 void RtpDumpFileHeader::WriteToByteBuffer(rtc::ByteBufferWriter* buf) {
   buf->WriteUInt32(start_sec);
@@ -44,7 +43,7 @@ void RtpDumpFileHeader::WriteToByteBuffer(rtc::ByteBufferWriter* buf) {
   buf->WriteUInt16(padding);
 }
 
-static const uint32_t kDefaultTimeIncrease = 30;
+static const int kDefaultTimeIncrease = 30;
 
 bool RtpDumpPacket::IsValidRtpPacket() const {
   return original_data_len >= data.size() &&
@@ -315,7 +314,7 @@ void RtpDumpWriter::set_packet_filter(int filter) {
 }
 
 uint32_t RtpDumpWriter::GetElapsedTime() const {
-  return rtc::TimeSince(start_time_ms_);
+  return static_cast<uint32_t>(rtc::Time() - start_time_ms_);
 }
 
 rtc::StreamResult RtpDumpWriter::WriteFileHeader() {
@@ -395,10 +394,10 @@ size_t RtpDumpWriter::FilterPacket(const void* data, size_t data_len,
 
 rtc::StreamResult RtpDumpWriter::WriteToStream(
     const void* data, size_t data_len) {
-  uint32_t before = rtc::Time();
+  int64_t before = rtc::Time();
   rtc::StreamResult result =
       stream_->WriteAll(data, data_len, NULL, NULL);
-  uint32_t delay = rtc::TimeSince(before);
+  uint32_t delay = static_cast<uint32_t>(rtc::TimeSince(before));
   if (delay >= warn_slow_writes_delay_) {
     LOG(LS_WARNING) << "Slow RtpDump: took " << delay << "ms to write "
                     << data_len << " bytes.";