Add histogram for percentage of sent frames that are limited in resolution due to bandwidth.

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 63 matching lines @@
       quality_limited_frame_counter_.Percent(kMinRequiredSamples);
   if (quality_limited != -1) {
     RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.QualityLimitedResolutionInPercent",
                              quality_limited);
   }
   int downscales = quality_downscales_counter_.Avg(kMinRequiredSamples);
   if (downscales != -1) {
     RTC_HISTOGRAM_ENUMERATION("WebRTC.Video.QualityLimitedResolutionDownscales",
                               downscales, 20);
   }
+  int bw_limited = bw_limited_frame_counter_.Percent(kMinRequiredSamples);
+  if (bw_limited != -1) {
+    RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BandwidthLimitedResolutionInPercent",
+        bw_limited);
+  }
+  int num_disabled = bw_resolutions_disabled_counter_.Avg(kMinRequiredSamples);
+  if (num_disabled != -1) {
+    RTC_HISTOGRAM_ENUMERATION(
+        "WebRTC.Video.BandwidthLimitedResolutionsDisabled", num_disabled, 10);
+  }
 }
 
 void SendStatisticsProxy::OnOutgoingRate(uint32_t framerate, uint32_t bitrate) {
   rtc::CritScope lock(&crit_);
   stats_.encode_frame_rate = framerate;
   stats_.media_bitrate_bps = bitrate;
 }
 
 void SendStatisticsProxy::CpuOveruseMetricsUpdated(
     const CpuOveruseMetrics& metrics) {
@@ skipping 89 matching lines @@
 
   if (encoded_image.adapt_reason_.quality_resolution_downscales != -1) {
     bool downscaled =
         encoded_image.adapt_reason_.quality_resolution_downscales > 0;
     quality_limited_frame_counter_.Add(downscaled);
     if (downscaled) {
       quality_downscales_counter_.Add(
           encoded_image.adapt_reason_.quality_resolution_downscales);
     }
   }
+  if (encoded_image.adapt_reason_.bw_resolutions_disabled != -1) {
+    bool bw_limited = encoded_image.adapt_reason_.bw_resolutions_disabled > 0;
+    bw_limited_frame_counter_.Add(bw_limited);
+    if (bw_limited) {
+      bw_resolutions_disabled_counter_.Add(
+         encoded_image.adapt_reason_.bw_resolutions_disabled);
+    }
+  }
 
   // TODO(asapersson): This is incorrect if simulcast layers are encoded on
   // different threads and there is no guarantee that one frame of all layers
   // are encoded before the next start.
   if (last_sent_frame_timestamp_ > 0 &&
       encoded_image._timeStamp != last_sent_frame_timestamp_) {
     sent_frame_rate_tracker_.AddSamples(1);
     sent_width_counter_.Add(max_sent_width_per_timestamp_);
     sent_height_counter_.Add(max_sent_height_per_timestamp_);
     max_sent_width_per_timestamp_ = 0;
@@ skipping 89 matching lines @@
 }
 
 void SendStatisticsProxy::SampleCounter::Add(int sample) {
   sum += sample;
   ++num_samples;
 }
 
 int SendStatisticsProxy::SampleCounter::Avg(int min_required_samples) const {
   if (num_samples < min_required_samples || num_samples == 0)
     return -1;
-  return sum / num_samples;
+  return (sum + (num_samples / 2)) / num_samples;
 }
 
 void SendStatisticsProxy::BoolSampleCounter::Add(bool sample) {
   if (sample)
     ++sum;
   ++num_samples;
 }
 
 int SendStatisticsProxy::BoolSampleCounter::Percent(
     int min_required_samples) const {
   return Fraction(min_required_samples, 100.0f);
 }
 
 int SendStatisticsProxy::BoolSampleCounter::Permille(
     int min_required_samples) const {
   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