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

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 52 matching lines @@
   }
   int encode_ms = encode_time_counter_.Avg(kMinRequiredSamples);
   if (encode_ms != -1)
     RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.EncodeTimeInMs", encode_ms);
 
   int key_frames_permille = key_frame_counter_.Permille(kMinRequiredSamples);
   if (key_frames_permille != -1) {
     RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesSentInPermille",
         key_frames_permille);
   }
+  int quality_limited =
+      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);
+  }
 }
 
 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 80 matching lines @@
   VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
   if (stats == nullptr)
     return;
 
   stats->width = encoded_image._encodedWidth;
   stats->height = encoded_image._encodedHeight;
   update_times_[ssrc].resolution_update_ms = clock_->TimeInMilliseconds();
 
   key_frame_counter_.Add(encoded_image._frameType == kKeyFrame);
 
+  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);
+    }
+  }
+
   // 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;
     max_sent_height_per_timestamp_ = 0;
@@ skipping 115 matching lines @@
 }
 
 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