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 44 matching lines @@
   }
   int sent_width = sent_width_counter_.Avg(kMinRequiredSamples);
   int sent_height = sent_height_counter_.Avg(kMinRequiredSamples);
   if (sent_width != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.SentWidthInPixels", sent_width);
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.SentHeightInPixels", sent_height);
   }
   int encode_ms = encode_time_counter_.Avg(kMinRequiredSamples);
   if (encode_ms != -1)
     RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.EncodeTimeInMs", encode_ms);
+
+  int quality_limited =
+      quality_limited_sent_frame_counter_.Percent(kMinRequiredSamples);

[stefan-webrtc, 2015/09/16 07:36:57]

I think "sent" is implied here and that you can leave it out.

[åsapersson, 2015/09/17 14:24:45]

Done.

+  if (quality_limited != -1) {
+    RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.QualityLimitedResolutionInPercent",
+                             quality_limited);
+  }
 }
 
 void SendStatisticsProxy::OutgoingRate(const int video_channel,
                                        const unsigned int framerate,
                                        const unsigned int bitrate) {
   rtc::CritScope lock(&crit_);
   stats_.encode_frame_rate = framerate;
   stats_.media_bitrate_bps = bitrate;
 }
 
@@ skipping 80 matching lines @@
 
   rtc::CritScope lock(&crit_);
   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();
 
+  if (encoded_image._length > 0) {
+    if (encoded_image.adapt_reason_.quality_limited_resolution_available) {
+      quality_limited_sent_frame_counter_.Add(
+          encoded_image.adapt_reason_.quality_limited_resolution);
+    }
+  }
+
   // 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_total_.Update(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 92 matching lines @@
   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;
 }
 
+void SendStatisticsProxy::BoolSampleCounter::Add(bool sample) {
+  if (sample)
+    ++sum;
+  ++num_samples;
+}
+
+int SendStatisticsProxy::BoolSampleCounter::Percent(
+    int min_required_samples) const {
+  if (num_samples < min_required_samples || num_samples == 0)
+    return -1;
+  return sum * 100 / num_samples;
+}
 }  // namespace webrtc