Remove ResetStatistics from RTP feedback.

webrtc/video_engine/vie_channel.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.
  */
 
@@ skipping 226 matching lines @@
       if (rtcp_counter.nack_requests > 0) {
         RTC_HISTOGRAM_PERCENTAGE(
             "WebRTC.Video.UniqueNackRequestsReceivedInPercent",
                 rtcp_counter.UniqueNackRequestsInPercent());
       }
       int fraction_lost = report_block_stats_sender_->FractionLostInPercent();
       if (fraction_lost != -1) {
         RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.SentPacketsLostInPercent",
             fraction_lost);
       }
+
+      StreamDataCounters rtp;

[åsapersson, 2015/07/07 12:18:43]

move this outside if-statement above

[pbos-webrtc, 2015/07/07 13:31:26]

Done.

+      StreamDataCounters rtx;
+      GetSendStreamDataCounters(&rtp, &rtx);
+      StreamDataCounters rtp_rtx = rtp;
+      rtp_rtx.Add(rtx);
+      elapsed_sec =
+          rtp_rtx.TimeSinceFirstPacketInMs(
+              Clock::GetRealTimeClock()->TimeInMilliseconds()) /
+          1000;
+      if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
+        RTC_HISTOGRAM_COUNTS_100000(
+            "WebRTC.Video.BitrateSentInKbps",
+            static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 /
+                             elapsed_sec / 1000));
+        RTC_HISTOGRAM_COUNTS_10000(
+            "WebRTC.Video.MediaBitrateSentInKbps",
+            static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
+        RTC_HISTOGRAM_COUNTS_10000(
+            "WebRTC.Video.PaddingBitrateSentInKbps",
+            static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 /
+                             elapsed_sec / 1000));
+        RTC_HISTOGRAM_COUNTS_10000(
+            "WebRTC.Video.RetransmittedBitrateSentInKbps",
+            static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 /
+                             elapsed_sec / 1000));
+        if (rtp_rtcp_->RtxSendStatus() != kRtxOff) {
+          RTC_HISTOGRAM_COUNTS_10000(
+              "WebRTC.Video.RtxBitrateSentInKbps",
+              static_cast<int>(rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
+                               1000));
+        }
+        bool fec_enabled = false;
+        uint8_t pltype_red;
+        uint8_t pltype_fec;
+        rtp_rtcp_->GenericFECStatus(fec_enabled, pltype_red, pltype_fec);
+        if (fec_enabled) {
+          RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FecBitrateSentInKbps",
+                                     static_cast<int>(rtp_rtx.fec.TotalBytes() *
+                                                      8 / elapsed_sec / 1000));
+        }
+      }
     }
   } else if (vie_receiver_.GetRemoteSsrc() > 0) {
     // Get receive stats if we are receiving packets, i.e. there is a remote
     // ssrc.
     RtcpPacketTypeCounter rtcp_counter;
     GetReceiveRtcpPacketTypeCounter(&rtcp_counter);
     int64_t elapsed_sec = rtcp_counter.TimeSinceFirstPacketInMs(now) / 1000;
     if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
       RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsSentPerMinute",
           rtcp_counter.nack_packets * 60 / elapsed_sec);
@@ skipping 38 matching lines @@
       }
       if (vie_receiver_.IsFecEnabled()) {
         RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FecBitrateReceivedInKbps",
                                    static_cast<int>(rtp_rtx.fec.TotalBytes() *
                                                     8 / elapsed_sec / 1000));
       }
     }
   }
 }
 
-void ViEChannel::UpdateHistogramsAtStopSend() {
-  StreamDataCounters rtp;
-  StreamDataCounters rtx;
-  GetSendStreamDataCounters(&rtp, &rtx);
-  StreamDataCounters rtp_rtx = rtp;
-  rtp_rtx.Add(rtx);
-
-  int64_t elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs(
-      Clock::GetRealTimeClock()->TimeInMilliseconds()) / 1000;
-  if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
-    return;
-  }
-  RTC_HISTOGRAM_COUNTS_100000(
-      "WebRTC.Video.BitrateSentInKbps",
-      static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
-                       1000));
-  RTC_HISTOGRAM_COUNTS_10000(
-      "WebRTC.Video.MediaBitrateSentInKbps",
-      static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
-  RTC_HISTOGRAM_COUNTS_10000(
-      "WebRTC.Video.PaddingBitrateSentInKbps",
-      static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
-                       1000));
-  RTC_HISTOGRAM_COUNTS_10000(
-      "WebRTC.Video.RetransmittedBitrateSentInKbps",
-      static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 / elapsed_sec /
-                       1000));
-  if (rtp_rtcp_->RtxSendStatus() != kRtxOff) {
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtxBitrateSentInKbps",
-                               static_cast<int>(rtx.transmitted.TotalBytes() *
-                                                8 / elapsed_sec / 1000));
-  }
-  bool fec_enabled = false;
-  uint8_t pltype_red;
-  uint8_t pltype_fec;
-  rtp_rtcp_->GenericFECStatus(fec_enabled, pltype_red, pltype_fec);
-  if (fec_enabled) {
-    RTC_HISTOGRAM_COUNTS_10000(
-        "WebRTC.Video.FecBitrateSentInKbps",
-        static_cast<int>(rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec / 1000));
-  }
-}
-
 int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec,
                                  bool new_stream) {
   if (!sender_) {
     return 0;
   }
   if (video_codec.codecType == kVideoCodecRED ||
       video_codec.codecType == kVideoCodecULPFEC) {
     LOG_F(LS_ERROR) << "Not a valid send codec " << video_codec.codecType;
     return -1;
   }
@@ skipping 970 matching lines @@
        it++) {
     RtpRtcp* rtp_rtcp = *it;
     rtp_rtcp->SetSendingMediaStatus(true);
     rtp_rtcp->SetSendingStatus(true);
   }
   send_payload_router_->set_active(true);
   return 0;
 }
 
 int32_t ViEChannel::StopSend() {
-  UpdateHistogramsAtStopSend();
   send_payload_router_->set_active(false);
   CriticalSectionScoped cs(rtp_rtcp_cs_.get());
   rtp_rtcp_->SetSendingMediaStatus(false);
   for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
        it != simulcast_rtp_rtcp_.end();
        it++) {
     RtpRtcp* rtp_rtcp = *it;
     rtp_rtcp->SetSendingMediaStatus(false);
   }
   if (!rtp_rtcp_->Sending()) {
     return -1;
   }
 
-  // Reset.
-  rtp_rtcp_->ResetSendDataCountersRTP();
   if (rtp_rtcp_->SetSendingStatus(false) != 0) {
     return -1;
   }
   for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
        it != simulcast_rtp_rtcp_.end();
        it++) {
     RtpRtcp* rtp_rtcp = *it;
-    rtp_rtcp->ResetSendDataCountersRTP();
     rtp_rtcp->SetSendingStatus(false);
   }
   return 0;
 }
 
 bool ViEChannel::Sending() {
   return rtp_rtcp_->Sending();
 }
 
 int32_t ViEChannel::StartReceive() {
@@ skipping 327 matching lines @@
 
 }
 
 void ViEChannel::OnIncomingCSRCChanged(const int32_t id,
                                        const uint32_t CSRC,
                                        const bool added) {
   assert(channel_id_ == ChannelId(id));
   CriticalSectionScoped cs(callback_cs_.get());
 }
 
-void ViEChannel::ResetStatistics(uint32_t ssrc) {
-  StreamStatistician* statistician =
-      vie_receiver_.GetReceiveStatistics()->GetStatistician(ssrc);
-  if (statistician)
-    statistician->ResetStatistics();
-}
-
 void ViEChannel::RegisterSendFrameCountObserver(
     FrameCountObserver* observer) {
   send_frame_count_observer_.Set(observer);
 }
 
 void ViEChannel::RegisterReceiveStatisticsProxy(
     ReceiveStatisticsProxy* receive_statistics_proxy) {
   CriticalSectionScoped cs(callback_cs_.get());
   vcm_receive_stats_callback_ = receive_statistics_proxy;
 }
 
 void ViEChannel::SetIncomingVideoStream(
     IncomingVideoStream* incoming_video_stream) {
   CriticalSectionScoped cs(callback_cs_.get());
   incoming_video_stream_ = incoming_video_stream;
 }
 }  // namespace webrtc