Change default timestamp to 64 bits in all webrtc directories.

webrtc/media/base/rtpdump.h

 /*
  *  Copyright (c) 2010 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 22 matching lines @@
 enum RtpDumpPacketFilter {
   PF_NONE = 0x0,
   PF_RTPHEADER = 0x1,
   PF_RTPPACKET = 0x3,  // includes header
   // PF_RTCPHEADER = 0x4,  // TODO(juberti)
   PF_RTCPPACKET = 0xC,  // includes header
   PF_ALL = 0xF
 };
 
 struct RtpDumpFileHeader {
-  RtpDumpFileHeader(uint32_t start_ms, uint32_t s, uint16_t p);
+  RtpDumpFileHeader(int64_t start_ms, uint32_t s, uint16_t p);
   void WriteToByteBuffer(rtc::ByteBufferWriter* buf);
 
   static const char kFirstLine[];
   static const size_t kHeaderLength = 16;
   uint32_t start_sec;   // start of recording, the seconds part.
   uint32_t start_usec;  // start of recording, the microseconds part.
   uint32_t source;      // network source (multicast address).
   uint16_t port;        // UDP port.
   uint16_t padding;     // 2 bytes padding.
 };
@@ skipping 52 matching lines @@
     return stream_->SetPosition(first_line_and_file_header_len_);
   }
 
  private:
   // Check if its matches "#!rtpplay1.0 address/port\n".
   bool CheckFirstLine(const std::string& first_line);
 
   rtc::StreamInterface* stream_;
   bool file_header_read_;
   size_t first_line_and_file_header_len_;
-  uint32_t start_time_ms_;
+  int64_t start_time_ms_;
   uint32_t ssrc_override_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(RtpDumpReader);
 };
 
 // RtpDumpLoopReader reads RTP dump packets from the input stream and rewinds
 // the stream when it ends. RtpDumpLoopReader maintains the elapsed time, the
 // RTP sequence number and the RTP timestamp properly. RtpDumpLoopReader can
 // handle both RTP dump and RTCP dump. We assume that the dump does not mix
 // RTP packets and RTCP packets.
@@ skipping 25 matching lines @@
   uint32_t rtp_timestamp_increase_;
   // How many RTP packets and how many payload frames in the input stream. RTP
   // packets belong to the same frame have the same RTP timestamp, different
   // dump timestamp, and different RTP sequence number.
   uint32_t packet_count_;
   uint32_t frame_count_;
   // The elapsed time, RTP sequence number, and RTP timestamp of the first and
   // the previous dump packets in the input stream.
   uint32_t first_elapsed_time_;
   int first_rtp_seq_num_;
-  uint32_t first_rtp_timestamp_;
+  int64_t first_rtp_timestamp_;
   uint32_t prev_elapsed_time_;
   int prev_rtp_seq_num_;
-  uint32_t prev_rtp_timestamp_;
+  int64_t prev_rtp_timestamp_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(RtpDumpLoopReader);
 };
 
 class RtpDumpWriter {
  public:
   explicit RtpDumpWriter(rtc::StreamInterface* stream);
 
   // Filter to control what packets we actually record.
   void set_packet_filter(int filter);
@@ skipping 24 matching lines @@
   rtc::StreamResult WritePacket(const void* data,
                                 size_t data_len,
                                 uint32_t elapsed,
                                 bool rtcp);
   size_t FilterPacket(const void* data, size_t data_len, bool rtcp);
   rtc::StreamResult WriteToStream(const void* data, size_t data_len);
 
   rtc::StreamInterface* stream_;
   int packet_filter_;
   bool file_header_written_;
-  uint32_t start_time_ms_;  // Time when the record starts.
+  int64_t start_time_ms_;  // Time when the record starts.
   // If writing to the stream takes longer than this many ms, log a warning.
-  uint32_t warn_slow_writes_delay_;
+  int64_t warn_slow_writes_delay_;
   RTC_DISALLOW_COPY_AND_ASSIGN(RtpDumpWriter);
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_RTPDUMP_H_