Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()

webrtc/video/send_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.
  */
 
@@ skipping 71 matching lines @@
 }
 
 SendStatisticsProxy::~SendStatisticsProxy() {}
 
 SendStatisticsProxy::UmaSamplesContainer::UmaSamplesContainer(
     const char* prefix)
     : uma_prefix_(prefix),
       max_sent_width_per_timestamp_(0),
       max_sent_height_per_timestamp_(0),
       input_frame_rate_tracker_(100u, 10u),
-      sent_frame_rate_tracker_(100u, 10u) {}
+      sent_frame_rate_tracker_(100u, 10u),
+      first_rtcp_stats_time_ms_(-1) {}
 
 SendStatisticsProxy::UmaSamplesContainer::~UmaSamplesContainer() {
   UpdateHistograms();
 }
 
 void SendStatisticsProxy::UmaSamplesContainer::UpdateHistograms() {
   RTC_DCHECK(uma_prefix_ == kRealtimePrefix || uma_prefix_ == kScreenPrefix);
   const int kIndex = uma_prefix_ == kScreenPrefix ? 1 : 0;
   const int kMinRequiredSamples = 200;
   int in_width = input_width_counter_.Avg(kMinRequiredSamples);
@@ skipping 56 matching lines @@
   int delay_ms = delay_counter_.Avg(kMinRequiredSamples);
   if (delay_ms != -1)
     RTC_HISTOGRAMS_COUNTS_100000(kIndex, uma_prefix_ + "SendSideDelayInMs",
                                  delay_ms);
 
   int max_delay_ms = max_delay_counter_.Avg(kMinRequiredSamples);
   if (max_delay_ms != -1) {
     RTC_HISTOGRAMS_COUNTS_100000(kIndex, uma_prefix_ + "SendSideDelayMaxInMs",
                                  max_delay_ms);
   }
+  int fraction_lost = report_block_stats_.FractionLostInPercent();
+  if (first_rtcp_stats_time_ms_ != -1) {
+    int64_t elapsed_time_ms = Clock::GetRealTimeClock()->TimeInMilliseconds() -
+                              first_rtcp_stats_time_ms_;
+    if (elapsed_time_ms / 1000 >= metrics::kMinRunTimeInSeconds &&
+        fraction_lost != -1) {
+      RTC_HISTOGRAMS_PERCENTAGE(
+          kIndex, uma_prefix_ + "SentPacketsLostInPercent", fraction_lost);
+    }
+  }
 }
 
 void SendStatisticsProxy::SetContentType(
     VideoEncoderConfig::ContentType content_type) {
   rtc::CritScope lock(&crit_);
   if (content_type_ != content_type) {
     uma_container_.reset(new UmaSamplesContainer(GetUmaPrefix(content_type)));
     content_type_ = content_type;
   }
 }
@@ skipping 168 matching lines @@
 }
 
 void SendStatisticsProxy::StatisticsUpdated(const RtcpStatistics& statistics,
                                             uint32_t ssrc) {
   rtc::CritScope lock(&crit_);
   VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
   if (stats == nullptr)
     return;
 
   stats->rtcp_stats = statistics;
+  uma_container_->report_block_stats_.Store(statistics, 0, ssrc);
+  if (uma_container_->first_rtcp_stats_time_ms_ == -1)
+    uma_container_->first_rtcp_stats_time_ms_ = clock_->TimeInMilliseconds();
 }
 
 void SendStatisticsProxy::CNameChanged(const char* cname, uint32_t ssrc) {
 }
 
 void SendStatisticsProxy::DataCountersUpdated(
     const StreamDataCounters& counters,
     uint32_t ssrc) {
   rtc::CritScope lock(&crit_);
   VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
@@ skipping 66 matching lines @@
   return Fraction(min_required_samples, 1000.0f);
 }
 
 int SendStatisticsProxy::BoolSampleCounter::Fraction(
     int min_required_samples, float multiplier) const {
   if (num_samples < min_required_samples || num_samples == 0)
     return -1;
   return static_cast<int>((sum * multiplier / num_samples) + 0.5f);
 }
 }  // namespace webrtc