rtt calculation handles time go backwards

webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc

 /*
  *  Copyright (c) 2012 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/rtcp_receiver.h"
 
 #include <assert.h>
 #include <string.h>
 
-#include <algorithm>
-
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
 #include "webrtc/modules/rtp_rtcp/source/time_util.h"
 #include "webrtc/system_wrappers/include/ntp_time.h"
 
 namespace webrtc {
@@ skipping 486 matching lines @@
       rb.ExtendedHighestSequenceNumber;
   reportBlock->remoteReceiveBlock.jitter = rb.Jitter;
   reportBlock->remoteReceiveBlock.delaySinceLastSR = rb.DelayLastSR;
   reportBlock->remoteReceiveBlock.lastSR = rb.LastSR;
 
   if (rtcpPacket.ReportBlockItem.Jitter > reportBlock->remoteMaxJitter) {
     reportBlock->remoteMaxJitter = rtcpPacket.ReportBlockItem.Jitter;
   }
 
   uint32_t send_time = rtcpPacket.ReportBlockItem.LastSR;
-  uint32_t rtt = 0;
+  int64_t rtt = 0;
 
   if (send_time > 0) {
     uint32_t delay = rtcpPacket.ReportBlockItem.DelayLastSR;
     // Local NTP time.
     uint32_t receive_time = CompactNtp(NtpTime(*_clock));
 
     // RTT in 1/(2^16) seconds.
     uint32_t rtt_ntp = receive_time - delay - send_time;
     // Convert to 1/1000 seconds (milliseconds).
-    uint32_t rtt_ms = CompactNtpIntervalToMs(rtt_ntp);
-    rtt = std::max<uint32_t>(rtt_ms, 1);
+    rtt = CompactNtpRttToMs(rtt_ntp);

[åsapersson, 2016/03/08 12:59:44]

I think this will now also result in RTT estimates for receive streams (earlier
prevented by _rtpRtcp.SendTimeOfSendReport > 0) and accounted for several times
in the CallStats class? Maybe add a check that this is not a receive_only
channel (and the opposite for XR rtt)?

[danilchap, 2016/03/08 15:02:02]

It shouldn't: send_time should be set to 0 to indicate there is no send time.
That is checked instead of previous _rtpRtcp.SendTimeOfSendReport > 0. Same is
for XR rtt.
Added comment for SR, fixed comment for XR.

[åsapersson, 2016/03/08 15:22:30]

The receive stream(s) has the same main_ssrc_ as the send stream. The received
RTCP packets are input to all streams, therefore i think receive streams also
will get an RTT estimate (which is picked up the the CallStats class).

     if (rtt > reportBlock->maxRTT) {
       // Store max RTT.
       reportBlock->maxRTT = rtt;
     }
     if (reportBlock->minRTT == 0) {
       // First RTT.
       reportBlock->minRTT = rtt;
     } else if (rtt < reportBlock->minRTT) {
       // Store min RTT.
       reportBlock->minRTT = rtt;
@@ skipping 374 matching lines @@
   uint32_t send_time = packet.XRDLRRReportBlockItem.LastRR;
   // RFC3411, section 4.5, LRR field discription states:
   // If no such block has been received, the field is set to zero.
   if (send_time == 0)
     return;
 
   uint32_t delay_rr = packet.XRDLRRReportBlockItem.DelayLastRR;
   uint32_t now = CompactNtp(NtpTime(*_clock));
 
   uint32_t rtt_ntp = now - delay_rr - send_time;
-  uint32_t rtt_ms = CompactNtpIntervalToMs(rtt_ntp);
-  xr_rr_rtt_ms_ = std::max<uint32_t>(rtt_ms, 1);
+  xr_rr_rtt_ms_ = CompactNtpRttToMs(rtt_ntp);
 
   rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpXrDlrrReportBlock;
 }
 
 void
 RTCPReceiver::HandleXRVOIPMetric(RTCPUtility::RTCPParserV2& rtcpParser,
                                  RTCPPacketInformation& rtcpPacketInformation)
 {
     const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
 
@@ skipping 503 matching lines @@
         return -1;
       }
       num += receiveInfo->TmmbrSet.lengthOfSet();
       receiveInfoIt++;
     }
   }
   return num;
 }
 
 }  // namespace webrtc