Move RTCP histograms from vie_channel to video channel stats proxies.

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)
     : clock_(clock),
+      start_time_ms_(clock->TimeInMilliseconds()),
       // 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;
 }
 
 ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {
   UpdateHistograms();
@@ skipping 27 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);
+
+  int64_t elapsed_sec = (clock_->TimeInMilliseconds() - start_time_ms_) / 1000;
+  if (elapsed_sec >= metrics::kMinRunTimeInSeconds) {
+    const RtcpPacketTypeCounter& counters = stats_.rtcp_packet_type_counts;
+
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsSentPerMinute",
+                               counters.nack_packets * 60 / elapsed_sec);
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsSentPerMinute",
+                               counters.fir_packets * 60 / elapsed_sec);
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsSentPerMinute",
+                               counters.pli_packets * 60 / elapsed_sec);
+    if (counters.nack_requests > 0) {
+      RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.UniqueNackRequestsSentInPercent",
+                               counters.UniqueNackRequestsInPercent());
+    }
+  }
 }
 
 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 125 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