Move RTP stats histograms from VieChannel to SendStatisticsProxy.

webrtc/video/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 146 matching lines @@
     }
     vcm_->RegisterFrameTypeCallback(this);
     vcm_->RegisterReceiveStatisticsCallback(this);
     vcm_->RegisterDecoderTimingCallback(this);
     vcm_->SetRenderDelay(kDefaultRenderDelayMs);
   }
   return 0;
 }
 
 ViEChannel::~ViEChannel() {
-  UpdateHistograms();
   // Make sure we don't get more callbacks from the RTP module.
   module_process_thread_->DeRegisterModule(
       vie_receiver_.GetReceiveStatistics());
   if (sender_) {
     send_payload_router_->SetSendingRtpModules(std::vector<RtpRtcp*>());
   }
   for (size_t i = 0; i < num_active_rtp_rtcp_modules_; ++i)
     packet_router_->RemoveRtpModule(rtp_rtcp_modules_[i], sender_);
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     module_process_thread_->DeRegisterModule(rtp_rtcp);
     delete rtp_rtcp;
   }
 }
 
-void ViEChannel::UpdateHistograms() {
-  if (sender_) {
-    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) {
-      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_modules_[0]->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_modules_[0]->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) {
   RTC_DCHECK(sender_);
   if (video_codec.codecType == kVideoCodecRED ||
       video_codec.codecType == kVideoCodecULPFEC) {
     LOG_F(LS_ERROR) << "Not a valid send codec " << video_codec.codecType;
     return -1;
   }
   if (kMaxSimulcastStreams < video_codec.numberOfSimulcastStreams) {
     LOG_F(LS_ERROR) << "Incorrect config "
@@ skipping 491 matching lines @@
   rtc::CritScope lock(&crit_);
   receive_stats_callback_ = receive_statistics_proxy;
 }
 
 void ViEChannel::SetIncomingVideoStream(
     IncomingVideoStream* incoming_video_stream) {
   rtc::CritScope lock(&crit_);
   incoming_video_stream_ = incoming_video_stream;
 }
 }  // namespace webrtc