[rtp_rtcp] time helper functions

webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h

 /*
  *  Copyright (c) 2013 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.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
 
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 
 #include <algorithm>
 #include <map>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/ntp_time.h"
 
 namespace webrtc {
 
 class CriticalSectionWrapper;
 
 class StreamStatisticianImpl : public StreamStatistician {
  public:
   StreamStatisticianImpl(Clock* clock,
                          RtcpStatisticsCallback* rtcp_callback,
                          StreamDataCountersCallback* rtp_callback);
@@ skipping 13 matching lines @@
                       size_t packet_length,
                       bool retransmitted);
   void FecPacketReceived(const RTPHeader& header, size_t packet_length);
   void SetMaxReorderingThreshold(int max_reordering_threshold);
   void ProcessBitrate();
   virtual void LastReceiveTimeNtp(uint32_t* secs, uint32_t* frac) const;
 
  private:
   bool InOrderPacketInternal(uint16_t sequence_number) const;
   RtcpStatistics CalculateRtcpStatistics();
-  void UpdateJitter(const RTPHeader& header,
-                    uint32_t receive_time_secs,
-                    uint32_t receive_time_frac);
+  void UpdateJitter(const RTPHeader& header, NtpTime receive_time);
   void UpdateCounters(const RTPHeader& rtp_header,
                       size_t packet_length,
                       bool retransmitted);
   void NotifyRtpCallback() LOCKS_EXCLUDED(stream_lock_.get());
   void NotifyRtcpCallback() LOCKS_EXCLUDED(stream_lock_.get());
 
   Clock* clock_;
   rtc::scoped_ptr<CriticalSectionWrapper> stream_lock_;
   Bitrate incoming_bitrate_;
   uint32_t ssrc_;
   int max_reordering_threshold_;  // In number of packets or sequence numbers.
 
   // Stats on received RTP packets.
   uint32_t jitter_q4_;
   uint32_t cumulative_loss_;
   uint32_t jitter_q4_transmission_time_offset_;
 
   int64_t last_receive_time_ms_;
-  uint32_t last_receive_time_secs_;
-  uint32_t last_receive_time_frac_;
+  NtpTime last_receive_time_ntp_;
   uint32_t last_received_timestamp_;
   int32_t last_received_transmission_time_offset_;
   uint16_t received_seq_first_;
   uint16_t received_seq_max_;
   uint16_t received_seq_wraps_;
 
   // Current counter values.
   size_t received_packet_overhead_;
   StreamDataCounters receive_counters_;
 
@@ skipping 47 matching lines @@
   Clock* clock_;
   rtc::scoped_ptr<CriticalSectionWrapper> receive_statistics_lock_;
   int64_t last_rate_update_ms_;
   StatisticianImplMap statisticians_;
 
   RtcpStatisticsCallback* rtcp_stats_callback_;
   StreamDataCountersCallback* rtp_stats_callback_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_