Move histogram for number of pause events to per stream.

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 304 matching lines @@
             kIndex, uma_prefix_ + "UniqueNackRequestsReceivedInPercent",
             counters.UniqueNackRequestsInPercent());
       }
     }
   }
 
   if (first_rtp_stats_time_ms_ != -1) {
     int64_t elapsed_sec =
         (clock_->TimeInMilliseconds() - first_rtp_stats_time_ms_) / 1000;
     if (elapsed_sec >= metrics::kMinRunTimeInSeconds) {
+      RTC_HISTOGRAMS_COUNTS_100(kIndex, uma_prefix_ + "NumberOfPauseEvents",
+                                target_rate_updates_.pause_resume_events);
+
+      int paused_time_percent =
+          paused_time_counter_.Percent(metrics::kMinRunTimeInSeconds * 1000);

[stefan-webrtc, 2016/12/13 12:18:01]

Why is this argument passed in? We already know it has been running that long
based on line 324, ,right?

[åsapersson, 2016/12/13 12:51:32]

Right, should be about the same if OnSetEncoderTargetRate is called at the
beginning and the end of a call. Maybe good to have kMinRunTimeInSeconds as
argument in case this would change.

[stefan-webrtc, 2016/12/19 12:37:15]

Acknowledged.

+      if (paused_time_percent != -1) {
+        RTC_HISTOGRAMS_PERCENTAGE(kIndex, uma_prefix_ + "PausedTimeInPercent",
+                                  paused_time_percent);
+      }
+
       StreamDataCounters rtp;
       StreamDataCounters rtx;
       AccumulateRtxStats(current_stats, rtp_config.rtx.ssrcs, &rtp, &rtx);
       StreamDataCounters start_rtp;
       StreamDataCounters start_rtx;
       AccumulateRtxStats(start_stats_, rtp_config.rtx.ssrcs, &start_rtp,
                          &start_rtx);
       rtp.Subtract(start_rtp);
       rtx.Subtract(start_rtx);
       StreamDataCounters rtp_rtx = rtp;
@@ skipping 125 matching lines @@
     return;
 
   stats->total_bitrate_bps = 0;
   stats->retransmit_bitrate_bps = 0;
   stats->height = 0;
   stats->width = 0;
 }
 
 void SendStatisticsProxy::OnSetEncoderTargetRate(uint32_t bitrate_bps) {
   rtc::CritScope lock(&crit_);
+  if (uma_container_->target_rate_updates_.last_ms == -1 && bitrate_bps == 0)
+    return;  // Start on first non-zero bitrate, may initially be zero.
+
+  int64_t now = clock_->TimeInMilliseconds();
+  if (uma_container_->target_rate_updates_.last_ms != -1) {
+    bool was_paused = stats_.target_media_bitrate_bps == 0;
+    int64_t diff_ms = now - uma_container_->target_rate_updates_.last_ms;
+    uma_container_->paused_time_counter_.Add(was_paused, diff_ms);
+
+    // Use last to not include update when stream is stopped and video disabled.
+    if (uma_container_->target_rate_updates_.last_paused_or_resumed)
+      ++uma_container_->target_rate_updates_.pause_resume_events;
+
+    // Check if video is paused/resumed.
+    uma_container_->target_rate_updates_.last_paused_or_resumed =
+        (bitrate_bps == 0) != was_paused;
+  }
+  uma_container_->target_rate_updates_.last_ms = now;
+
   stats_.target_media_bitrate_bps = bitrate_bps;
 }
 
 void SendStatisticsProxy::OnSendEncodedImage(
     const EncodedImage& encoded_image,
     const CodecSpecificInfo* codec_info) {
   size_t simulcast_idx = 0;
 
   rtc::CritScope lock(&crit_);
   ++stats_.frames_encoded;
@@ skipping 219 matching lines @@
     return -1;
   return (sum + (num_samples / 2)) / num_samples;
 }
 
 void SendStatisticsProxy::BoolSampleCounter::Add(bool sample) {
   if (sample)
     ++sum;
   ++num_samples;
 }
 
+void SendStatisticsProxy::BoolSampleCounter::Add(bool sample, int64_t count) {
+  if (sample)
+    sum += count;
+  num_samples += count;
+}
 int SendStatisticsProxy::BoolSampleCounter::Percent(
-    int min_required_samples) const {
+    int64_t min_required_samples) const {
   return Fraction(min_required_samples, 100.0f);
 }
 
 int SendStatisticsProxy::BoolSampleCounter::Permille(
-    int min_required_samples) const {
+    int64_t min_required_samples) const {
   return Fraction(min_required_samples, 1000.0f);
 }
 
 int SendStatisticsProxy::BoolSampleCounter::Fraction(
-    int min_required_samples, float multiplier) const {
+    int64_t 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