Move histograms for rtp receive counters to ReceiveStatisticsProxy

webrtc/video/receive_statistics_proxy.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/video/receive_statistics_proxy.h"
 
 #include <cmath>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 
 namespace webrtc {
 
-ReceiveStatisticsProxy::ReceiveStatisticsProxy(uint32_t ssrc, Clock* clock)
+ReceiveStatisticsProxy::ReceiveStatisticsProxy(
+    const VideoReceiveStream::Config& config,
+    Clock* clock)
     : clock_(clock),
+      config_(config),
       // 1000ms window, scale 1000 for ms to s.
       decode_fps_estimator_(1000, 1000),
       renders_fps_estimator_(1000, 1000),
       render_fps_tracker_(100u, 10u),
       render_pixel_tracker_(100u, 10u) {
-  stats_.ssrc = ssrc;
+  stats_.ssrc = config.rtp.remote_ssrc;
+  for (auto it : config.rtp.rtx)
+    rtx_stats_[it.second.ssrc] = StreamDataCounters();
 }
 
 ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {
   UpdateHistograms();
 }
 
 void ReceiveStatisticsProxy::UpdateHistograms() {
   int fraction_lost = report_block_stats_.FractionLostInPercent();
   if (fraction_lost != -1) {
     RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
@@ skipping 21 matching lines @@
   // TODO(asapersson): DecoderTiming() is call periodically (each 1000ms) and
   // not per frame. Change decode time to include every frame.
   const int kMinRequiredDecodeSamples = 5;
   int decode_ms = decode_time_counter_.Avg(kMinRequiredDecodeSamples);
   if (decode_ms != -1)
     RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);
 
   int delay_ms = delay_counter_.Avg(kMinRequiredDecodeSamples);
   if (delay_ms != -1)
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);
+
+  StreamDataCounters rtp = stats_.rtp_stats;
+  StreamDataCounters rtx;
+  for (auto it : rtx_stats_)
+    rtx.Add(it.second);
+  StreamDataCounters rtp_rtx = rtp;
+  rtp_rtx.Add(rtx);
+  int64_t elapsed_sec =
+      rtp_rtx.TimeSinceFirstPacketInMs(clock_->TimeInMilliseconds()) / 1000;
+  if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
+    RTC_HISTOGRAM_COUNTS_10000(
+        "WebRTC.Video.BitrateReceivedInKbps",
+        static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
+                         1000));
+    RTC_HISTOGRAM_COUNTS_10000(
+        "WebRTC.Video.MediaBitrateReceivedInKbps",
+        static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
+    RTC_HISTOGRAM_COUNTS_10000(
+        "WebRTC.Video.PaddingBitrateReceivedInKbps",
+        static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
+                         1000));
+    RTC_HISTOGRAM_COUNTS_10000(
+        "WebRTC.Video.RetransmittedBitrateReceivedInKbps",
+        static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 / elapsed_sec /
+                         1000));
+    if (!rtx_stats_.empty()) {
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtxBitrateReceivedInKbps",
+                                 static_cast<int>(rtx.transmitted.TotalBytes() *
+                                                  8 / elapsed_sec / 1000));
+    }
+    if (config_.rtp.fec.ulpfec_payload_type != -1) {
+      RTC_HISTOGRAM_COUNTS_10000(
+          "WebRTC.Video.FecBitrateReceivedInKbps",
+          static_cast<int>(rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec / 1000));
+    }
+  }
 }
 
 VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const {
   rtc::CritScope lock(&crit_);
   return stats_;
 }
 
 void ReceiveStatisticsProxy::OnIncomingPayloadType(int payload_type) {
   rtc::CritScope lock(&crit_);
   stats_.current_payload_type = payload_type;
@@ skipping 60 matching lines @@
   // receive stats from one of them.
   if (stats_.ssrc != ssrc)
     return;
   stats_.c_name = cname;
 }
 
 void ReceiveStatisticsProxy::DataCountersUpdated(
     const webrtc::StreamDataCounters& counters,
     uint32_t ssrc) {
   rtc::CritScope lock(&crit_);
-  if (stats_.ssrc != ssrc)
-    return;
-  stats_.rtp_stats = counters;
+  if (ssrc == stats_.ssrc) {
+    stats_.rtp_stats = counters;
+  } else {
+    auto it = rtx_stats_.find(ssrc);
+    if (it != rtx_stats_.end()) {
+      it->second = counters;
+    } else {
+      RTC_NOTREACHED() << "Unexpected stream ssrc: " << ssrc;
+    }
+  }
 }
 
 void ReceiveStatisticsProxy::OnDecodedFrame() {
   uint64_t now = clock_->TimeInMilliseconds();
 
   rtc::CritScope lock(&crit_);
   decode_fps_estimator_.Update(1, now);
   stats_.decode_frame_rate = decode_fps_estimator_.Rate(now);
 }
 
@@ skipping 42 matching lines @@
   ++num_samples;
 }
 
 int ReceiveStatisticsProxy::SampleCounter::Avg(int min_required_samples) const {
   if (num_samples < min_required_samples || num_samples == 0)
     return -1;
   return sum / num_samples;
 }
 
 }  // namespace webrtc