[rtp_rtcp] time helper functions

webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc

 /*
  *  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.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h"
 
 #include <math.h>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/time_util.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
 const int64_t kStatisticsTimeoutMs = 8000;
 const int64_t kStatisticsProcessIntervalMs = 1000;
 
 StreamStatistician::~StreamStatistician() {}
 
 StreamStatisticianImpl::StreamStatisticianImpl(
     Clock* clock,
     RtcpStatisticsCallback* rtcp_callback,
     StreamDataCountersCallback* rtp_callback)
     : clock_(clock),
       stream_lock_(CriticalSectionWrapper::CreateCriticalSection()),
       incoming_bitrate_(clock, NULL),
       ssrc_(0),
       max_reordering_threshold_(kDefaultMaxReorderingThreshold),
       jitter_q4_(0),
       cumulative_loss_(0),
       jitter_q4_transmission_time_offset_(0),
       last_receive_time_ms_(0),
-      last_receive_time_secs_(0),
-      last_receive_time_frac_(0),
       last_received_timestamp_(0),
       last_received_transmission_time_offset_(0),
       received_seq_first_(0),
       received_seq_max_(0),
       received_seq_wraps_(0),
       received_packet_overhead_(12),
       last_report_inorder_packets_(0),
       last_report_old_packets_(0),
       last_report_seq_max_(0),
       rtcp_callback_(rtcp_callback),
@@ skipping 20 matching lines @@
 
   if (receive_counters_.transmitted.packets == 1) {
     received_seq_first_ = header.sequenceNumber;
     receive_counters_.first_packet_time_ms = clock_->TimeInMilliseconds();
   }
 
   // Count only the new packets received. That is, if packets 1, 2, 3, 5, 4, 6
   // are received, 4 will be ignored.
   if (in_order) {
     // Current time in samples.
-    uint32_t receive_time_secs;
-    uint32_t receive_time_frac;
-    clock_->CurrentNtp(receive_time_secs, receive_time_frac);
+    NtpTime receive_time(*clock_);
 
     // Wrong if we use RetransmitOfOldPacket.
     if (receive_counters_.transmitted.packets > 1 &&
         received_seq_max_ > header.sequenceNumber) {
       // Wrap around detected.
       received_seq_wraps_++;
     }
     // New max.
     received_seq_max_ = header.sequenceNumber;
 
     // If new time stamp and more than one in-order packet received, calculate
     // new jitter statistics.
     if (header.timestamp != last_received_timestamp_ &&
         (receive_counters_.transmitted.packets -
          receive_counters_.retransmitted.packets) > 1) {
-      UpdateJitter(header, receive_time_secs, receive_time_frac);
+      UpdateJitter(header, receive_time);
     }
     last_received_timestamp_ = header.timestamp;
-    last_receive_time_secs_ = receive_time_secs;
-    last_receive_time_frac_ = receive_time_frac;
+    last_receive_time_ntp_ = receive_time;
     last_receive_time_ms_ = clock_->TimeInMilliseconds();
   }
 
   size_t packet_oh = header.headerLength + header.paddingLength;
 
   // Our measured overhead. Filter from RFC 5104 4.2.1.2:
   // avg_OH (new) = 15/16*avg_OH (old) + 1/16*pckt_OH,
   received_packet_overhead_ = (15 * received_packet_overhead_ + packet_oh) >> 4;
 }
 
 void StreamStatisticianImpl::UpdateJitter(const RTPHeader& header,
-                                          uint32_t receive_time_secs,
-                                          uint32_t receive_time_frac) {
-  uint32_t receive_time_rtp = RtpUtility::ConvertNTPTimeToRTP(
-      receive_time_secs, receive_time_frac, header.payload_type_frequency);
+                                          NtpTime receive_time) {
+  uint32_t receive_time_rtp =
+      NtpToRtp(receive_time, header.payload_type_frequency);
   uint32_t last_receive_time_rtp =
-      RtpUtility::ConvertNTPTimeToRTP(last_receive_time_secs_,
-                                      last_receive_time_frac_,
-                                      header.payload_type_frequency);
+      NtpToRtp(last_receive_time_ntp_, header.payload_type_frequency);
   int32_t time_diff_samples = (receive_time_rtp - last_receive_time_rtp) -
       (header.timestamp - last_received_timestamp_);
 
   time_diff_samples = abs(time_diff_samples);
 
   // lib_jingle sometimes deliver crazy jumps in TS for the same stream.
   // If this happens, don't update jitter value. Use 5 secs video frequency
   // as the threshold.
   if (time_diff_samples < 450000) {
     // Note we calculate in Q4 to avoid using float.
@@ skipping 178 matching lines @@
 }
 
 void StreamStatisticianImpl::ProcessBitrate() {
   CriticalSectionScoped cs(stream_lock_.get());
   incoming_bitrate_.Process();
 }
 
 void StreamStatisticianImpl::LastReceiveTimeNtp(uint32_t* secs,
                                                 uint32_t* frac) const {
   CriticalSectionScoped cs(stream_lock_.get());
-  *secs = last_receive_time_secs_;
-  *frac = last_receive_time_frac_;
+  *secs = last_receive_time_ntp_.seconds();
+  *frac = last_receive_time_ntp_.fractions();
 }
 
 bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
     const RTPHeader& header, int64_t min_rtt) const {
   CriticalSectionScoped cs(stream_lock_.get());
   if (InOrderPacketInternal(header.sequenceNumber)) {
     return false;
   }
   uint32_t frequency_khz = header.payload_type_frequency / 1000;
   assert(frequency_khz > 0);
@@ skipping 205 matching lines @@
 
 int32_t NullReceiveStatistics::Process() { return 0; }
 
 void NullReceiveStatistics::RegisterRtcpStatisticsCallback(
     RtcpStatisticsCallback* callback) {}
 
 void NullReceiveStatistics::RegisterRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {}
 
 }  // namespace webrtc