Piggybacking simulcast id and ALR experiment id into video content type extension.

webrtc/video/receive_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.
  */
 
 #include "webrtc/video/receive_statistics_proxy.h"
 
 #include <algorithm>
 #include <cmath>
+#include <sstream>
 #include <utility>
 
+#include "webrtc/modules/pacing/alr_detector.h"
 #include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/logging.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 
 namespace webrtc {
 namespace {
 // Periodic time interval for processing samples for |freq_offset_counter_|.
@@ skipping 14 matching lines @@
 const int kLowQpThresholdVp8 = 60;
 const int kHighQpThresholdVp8 = 70;
 const int kLowVarianceThreshold = 1;
 const int kHighVarianceThreshold = 2;
 
 // Some metrics are reported as a maximum over this period.
 const int kMovingMaxWindowMs = 10000;
 
 // How large window we use to calculate the framerate/bitrate.
 const int kRateStatisticsWindowSizeMs = 1000;
+
+std::string UmaPrefixForContentType(VideoContentType content_type) {
+  std::stringstream ss;
+  ss << "WebRTC.Video";
+  if (content_type.IsScreenshare()) {
+    ss << ".Screenshare";
+  }
+  return ss.str();
+}
+
+std::string UmaSuffixForContentType(VideoContentType content_type) {
+  std::stringstream ss;
+  int simulcast_id = content_type.GetSimulcastId();
+  if (simulcast_id > 0) {
+    ss << ".S" << simulcast_id - 1;
+  }
+  int experiment_id = content_type.GetExperimentId();
+  if (experiment_id > 0) {
+    ss << ".ExperimentGroup" << experiment_id - 1;
+  }
+  return ss.str();
+}
 }  // namespace
 
 ReceiveStatisticsProxy::ReceiveStatisticsProxy(
     const VideoReceiveStream::Config* config,
     Clock* clock)
     : clock_(clock),
       config_(*config),
       start_ms_(clock->TimeInMilliseconds()),
       last_sample_time_(clock->TimeInMilliseconds()),
       fps_threshold_(kLowFpsThreshold,
                      kHighFpsThreshold,
                      kBadFraction,
                      kNumMeasurements),
       qp_threshold_(kLowQpThresholdVp8,
                     kHighQpThresholdVp8,
                     kBadFraction,
                     kNumMeasurements),
       variance_threshold_(kLowVarianceThreshold,
                           kHighVarianceThreshold,
                           kBadFraction,
                           kNumMeasurementsVariance),
       num_bad_states_(0),
       num_certain_states_(0),
       // 1000ms window, scale 1000 for ms to s.
       decode_fps_estimator_(1000, 1000),
       renders_fps_estimator_(1000, 1000),
       render_fps_tracker_(100, 10u),
       render_pixel_tracker_(100, 10u),
       total_byte_tracker_(100, 10u),  // bucket_interval_ms, bucket_count
-      e2e_delay_max_ms_video_(-1),
-      e2e_delay_max_ms_screenshare_(-1),
-      interframe_delay_max_ms_video_(-1),
-      interframe_delay_max_ms_screenshare_(-1),
       interframe_delay_max_moving_(kMovingMaxWindowMs),
       freq_offset_counter_(clock, nullptr, kFreqOffsetProcessIntervalMs),
       first_report_block_time_ms_(-1),
       avg_rtt_ms_(0),
       last_content_type_(VideoContentType::UNSPECIFIED) {
   stats_.ssrc = config_.rtp.remote_ssrc;
   // TODO(brandtr): Replace |rtx_stats_| with a single instance of
   // StreamDataCounters.
   if (config_.rtp.rtx_ssrc) {
     rtx_stats_[config_.rtp.rtx_ssrc] = StreamDataCounters();
   }
 }
 
 ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {
   UpdateHistograms();
 }
 
 void ReceiveStatisticsProxy::UpdateHistograms() {
-  RTC_HISTOGRAM_COUNTS_100000(
-      "WebRTC.Video.ReceiveStreamLifetimeInSeconds",
-      (clock_->TimeInMilliseconds() - start_ms_) / 1000);
+  int stream_duration_sec = (clock_->TimeInMilliseconds() - start_ms_) / 1000;
+  RTC_HISTOGRAM_COUNTS_100000("WebRTC.Video.ReceiveStreamLifetimeInSeconds",
+                              stream_duration_sec);
+  LOG(LS_INFO) << "WebRTC.Video.ReceiveStreamLifetimeInSeconds "
+               << stream_duration_sec;

[sprang_webrtc, 2017/08/28 16:25:02]

Check that we have received at least one frame before reporting this. I think
it's all too common that applications create a bunch of streams that are never
used.

[ilnik, 2017/08/29 07:56:26]

Done.

 
   if (first_report_block_time_ms_ != -1 &&
       ((clock_->TimeInMilliseconds() - first_report_block_time_ms_) / 1000) >=
           metrics::kMinRunTimeInSeconds) {
     int fraction_lost = report_block_stats_.FractionLostInPercent();
     if (fraction_lost != -1) {
       RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
                                fraction_lost);
       LOG(LS_INFO) << "WebRTC.Video.ReceivedPacketsLostInPercent "
                    << fraction_lost;
     }
   }
 
   const int kMinRequiredSamples = 200;
   int samples = static_cast<int>(render_fps_tracker_.TotalSampleCount());
   if (samples >= kMinRequiredSamples) {
     RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.RenderFramesPerSecond",
                              round(render_fps_tracker_.ComputeTotalRate()));
     RTC_HISTOGRAM_COUNTS_100000(
         "WebRTC.Video.RenderSqrtPixelsPerSecond",
         round(render_pixel_tracker_.ComputeTotalRate()));
   }
-  int width = render_width_counter_.Avg(kMinRequiredSamples);
-  int height = render_height_counter_.Avg(kMinRequiredSamples);
-  if (width != -1) {
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedWidthInPixels", width);
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedHeightInPixels", height);
-    LOG(LS_INFO) << "WebRTC.Video.ReceivedWidthInPixels " << width;
-    LOG(LS_INFO) << "WebRTC.Video.ReceivedHeightInPixels " << height;
-  }
+
   int sync_offset_ms = sync_offset_counter_.Avg(kMinRequiredSamples);
   if (sync_offset_ms != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.AVSyncOffsetInMs", sync_offset_ms);
     LOG(LS_INFO) << "WebRTC.Video.AVSyncOffsetInMs " << sync_offset_ms;
   }
   AggregatedStats freq_offset_stats = freq_offset_counter_.GetStats();
   if (freq_offset_stats.num_samples > 0) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtpToNtpFreqOffsetInKhz",
                                freq_offset_stats.average);
     LOG(LS_INFO) << "WebRTC.Video.RtpToNtpFreqOffsetInKhz, "
@@ skipping 37 matching lines @@
   int current_delay_ms = current_delay_counter_.Avg(kMinRequiredSamples);
   if (current_delay_ms != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.CurrentDelayInMs",
                                current_delay_ms);
     LOG(LS_INFO) << "WebRTC.Video.CurrentDelayInMs " << current_delay_ms;
   }
   int delay_ms = delay_counter_.Avg(kMinRequiredSamples);
   if (delay_ms != -1)
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);
 
-  int e2e_delay_ms_video = e2e_delay_counter_video_.Avg(kMinRequiredSamples);
-  if (e2e_delay_ms_video != -1) {
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.EndToEndDelayInMs",
-                               e2e_delay_ms_video);
-    LOG(LS_INFO) << "WebRTC.Video.EndToEndDelayInMs " << e2e_delay_ms_video;
+  // Aggregate content_specific_stats_ by removing experiment or simulcast
+  // information;
+  std::map<VideoContentType, ContentSpecificStats> aggregated_stats;
+  for (auto it : content_specific_stats_) {
+    // Calculate simulcast specific metrics (".S0" ... ".S2" suffixes):
+    VideoContentType content_type = it.first;
+    if (content_type.GetSimulcastId() > 0) {
+      // Aggregate on experiment id.
+      content_type.SetExperimentId(0);
+      aggregated_stats[content_type].Add(it.second);
+    }
+    // Calculate experiment specific metrics:

[sprang_webrtc, 2017/08/28 16:25:02]

: ?

[ilnik, 2017/08/29 07:56:26]

Done.

+    content_type = it.first;
+    if (content_type.GetExperimentId() > 0) {
+      // Aggregate on simulcast id.
+      content_type.SetSimulcastId(0);
+      aggregated_stats[content_type].Add(it.second);
+    }
+    // calculate aggregated metrics (no suffixes. Aggregated on everything).

[sprang_webrtc, 2017/08/28 16:25:02]

nit: s/calculate/Calculate

[ilnik, 2017/08/29 07:56:26]

Done.

+    content_type = it.first;
+    content_type.SetSimulcastId(0);
+    content_type.SetExperimentId(0);
+    aggregated_stats[content_type].Add(it.second);

[sprang_webrtc, 2017/08/28 16:25:02]

We shouldn't do the addition if ss == 0 && exp == 0, right? Otherwise we add it
to itself.

[ilnik, 2017/08/29 07:56:27]

We add to the separate map. We need to make this addition. Making it in place is
tricky as we need to be very careful with the order of additions.

[sprang_webrtc, 2017/08/29 08:04:38]

Ah, you're right. I misread.

   }
 
-  int e2e_delay_ms_screenshare =
-      e2e_delay_counter_screenshare_.Avg(kMinRequiredSamples);
-  if (e2e_delay_ms_screenshare != -1) {
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.Screenshare.EndToEndDelayInMs",
-                               e2e_delay_ms_screenshare);
-  }
+  for (auto it : aggregated_stats) {
+    // for the metric Foo we report the following slices:

[sprang_webrtc, 2017/08/28 16:25:02]

nit: Capital F

[ilnik, 2017/08/29 07:56:26]

Done.

+    // WebRTC.Video.Foo,
+    // WebRTC.Video.Screenshare.Foo,
+    // WebRTC.Video.Foo.S[0-3],
+    // WebRTC.Video.Foo.ExperimentGroup[0-7],
+    // WebRTC.Video.Screenshare.Foo.S[0-3],
+    // WebRTC.Video.Screenshare.Foo.ExperimentGroup[0-7].
+    auto content_type = it.first;
+    auto stats = it.second;
+    std::string uma_prefix = UmaPrefixForContentType(content_type);
+    std::string uma_suffix = UmaSuffixForContentType(content_type);
+    // Metrics can be sliced on either simulcast id or experiment id but not
+    // both.
+    RTC_DCHECK(content_type.GetExperimentId() == 0 ||
+               content_type.GetSimulcastId() == 0);
 
-  int e2e_delay_max_ms_video = e2e_delay_max_ms_video_;
-  if (e2e_delay_max_ms_video != -1) {
-    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Video.EndToEndDelayMaxInMs",
-                                e2e_delay_max_ms_video);
-  }
+    int e2e_delay_ms = stats.e2e_delay_counter.Avg(kMinRequiredSamples);
+    if (e2e_delay_ms != -1) {
+      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+          uma_prefix + ".EndToEndDelayInMs" + uma_suffix, e2e_delay_ms);
+      LOG(LS_INFO) << uma_prefix << ".EndToEndDelayInMs" << uma_suffix << " "
+                   << e2e_delay_ms;
+    }
+    int e2e_delay_max_ms = stats.e2e_delay_counter.Max();
+    if (e2e_delay_max_ms != -1 && e2e_delay_ms != -1) {
+      RTC_HISTOGRAM_COUNTS_SPARSE_100000(
+          uma_prefix + ".EndToEndDelayMaxInMs" + uma_suffix, e2e_delay_max_ms);
+      LOG(LS_INFO) << uma_prefix << ".EndToEndDelayMaxInMs" << uma_suffix << " "
+                   << e2e_delay_max_ms;
+    }
+    int interframe_delay_ms =
+        stats.interframe_delay_counter.Avg(kMinRequiredSamples);
+    if (interframe_delay_ms != -1) {
+      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+          uma_prefix + ".InterframeDelayInMs" + uma_suffix,
+          interframe_delay_ms);
+      LOG(LS_INFO) << uma_prefix << ".InterframeDelayInMs" << uma_suffix << " "
+                   << interframe_delay_ms;
+    }
+    int interframe_delay_max_ms = stats.interframe_delay_counter.Max();
+    if (interframe_delay_max_ms != -1 && interframe_delay_ms != -1) {
+      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+          uma_prefix + ".InterframeDelayMaxInMs" + uma_suffix,
+          interframe_delay_max_ms);
+      LOG(LS_INFO) << uma_prefix << ".InterframeDelayMaxInMs" << uma_suffix
+                   << " " << interframe_delay_max_ms;
+    }
 
-  int e2e_delay_max_ms_screenshare = e2e_delay_max_ms_screenshare_;
-  if (e2e_delay_max_ms_screenshare != -1) {
-    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Video.Screenshare.EndToEndDelayMaxInMs",
-                                e2e_delay_max_ms_screenshare);
-  }
+    int width = stats.received_width.Avg(kMinRequiredSamples);
+    if (width != -1) {
+      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+          uma_prefix + ".ReceivedWidthInPixels" + uma_suffix, width);
+      LOG(LS_INFO) << uma_prefix << ".ReceivedWidthInPixels" << uma_suffix
+                   << " " << width;
+    }
 
-  int interframe_delay_ms_screenshare =
-      interframe_delay_counter_screenshare_.Avg(kMinRequiredSamples);
-  if (interframe_delay_ms_screenshare != -1) {
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.Screenshare.InterframeDelayInMs",
-                               interframe_delay_ms_screenshare);
-    RTC_DCHECK_GE(interframe_delay_max_ms_screenshare_,
-                  interframe_delay_ms_screenshare);
-    RTC_HISTOGRAM_COUNTS_10000(
-        "WebRTC.Video.Screenshare.InterframeDelayMaxInMs",
-        interframe_delay_max_ms_screenshare_);
-  }
+    int height = stats.received_height.Avg(kMinRequiredSamples);
+    if (height != -1) {
+      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+          uma_prefix + ".ReceivedHeightInPixels" + uma_suffix, height);
+      LOG(LS_INFO) << uma_prefix << ".ReceivedHeightInPixels" << uma_suffix
+                   << " " << height;
+    }
 
-  int interframe_delay_ms_video =
-      interframe_delay_counter_video_.Avg(kMinRequiredSamples);
-  if (interframe_delay_ms_video != -1) {
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.InterframeDelayInMs",
-                               interframe_delay_ms_video);
-    RTC_DCHECK_GE(interframe_delay_max_ms_video_, interframe_delay_ms_video);
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.InterframeDelayMaxInMs",
-                               interframe_delay_max_ms_video_);
+    if (content_type != VideoContentType::UNSPECIFIED) {
+      // Don't report these 3 metrics unsliced, as more precise variants
+      // are reported separately in this method.
+      float flow_duration_sec = stats.flow_duration_ms / 1000.0;
+      if (flow_duration_sec >= metrics::kMinRunTimeInSeconds) {
+        int media_bitrate_kbps = static_cast<int>(stats.total_media_bytes * 8 /
+                                                  flow_duration_sec / 1000);
+        RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+            uma_prefix + ".MediaBitrateReceivedInKbps" + uma_suffix,
+            media_bitrate_kbps);
+        LOG(LS_INFO) << uma_prefix << ".MediaBitrateReceivedInKbps"
+                     << uma_suffix << " " << media_bitrate_kbps;
+      }
+
+      int num_total_frames =
+          stats.frame_counts.key_frames + stats.frame_counts.delta_frames;
+      if (num_total_frames >= kMinRequiredSamples) {
+        int num_key_frames = stats.frame_counts.key_frames;
+        int key_frames_permille =
+            (num_key_frames * 1000 + num_total_frames / 2) / num_total_frames;
+        RTC_HISTOGRAM_COUNTS_SPARSE_1000(
+            uma_prefix + ".KeyFramesReceivedInPermille" + uma_suffix,
+            key_frames_permille);
+        LOG(LS_INFO) << uma_prefix << ".KeyFramesReceivedInPermille"
+                     << uma_suffix << " " << key_frames_permille;
+      }
+
+      int qp = stats.qp_counter.Avg(kMinRequiredSamples);
+      if (qp != -1) {
+        RTC_HISTOGRAM_COUNTS_SPARSE_200(
+            uma_prefix + ".Decoded.Vp8.Qp" + uma_suffix, qp);
+        LOG(LS_INFO) << uma_prefix << ".Decoded.Vp8.Qp" << uma_suffix << " "
+                     << qp;
+      }
+    }
   }
 
   StreamDataCounters rtp = stats_.rtp_stats;
   StreamDataCounters rtx;
   for (auto it : rtx_stats_)
     rtx.Add(it.second);
   StreamDataCounters rtp_rtx = rtp;
   rtp_rtx.Add(rtx);
   int64_t elapsed_sec =
       rtp_rtx.TimeSinceFirstPacketInMs(clock_->TimeInMilliseconds()) / 1000;
   if (elapsed_sec >= metrics::kMinRunTimeInSeconds) {
     RTC_HISTOGRAM_COUNTS_10000(
         "WebRTC.Video.BitrateReceivedInKbps",
         static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
                          1000));
-    RTC_HISTOGRAM_COUNTS_10000(
-        "WebRTC.Video.MediaBitrateReceivedInKbps",
-        static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
+    int media_bitrate_kbs =
+        static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000);
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.MediaBitrateReceivedInKbps",
+                               media_bitrate_kbs);
+    LOG(LS_INFO) << "WebRTC.Video.MediaBitrateReceivedInKbps "
+                 << media_bitrate_kbs;
     RTC_HISTOGRAM_COUNTS_10000(
         "WebRTC.Video.PaddingBitrateReceivedInKbps",
         static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
                          1000));
     RTC_HISTOGRAM_COUNTS_10000(
         "WebRTC.Video.RetransmittedBitrateReceivedInKbps",
         static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 / elapsed_sec /
                          1000));
     if (!rtx_stats_.empty()) {
       RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtxBitrateReceivedInKbps",
@@ skipping 256 matching lines @@
   }
   if (total_bytes > last_total_bytes)
     total_byte_tracker_.AddSamples(total_bytes - last_total_bytes);
 }
 
 void ReceiveStatisticsProxy::OnDecodedFrame(rtc::Optional<uint8_t> qp,
                                             VideoContentType content_type) {
   uint64_t now = clock_->TimeInMilliseconds();
 
   rtc::CritScope lock(&crit_);
+
+  ContentSpecificStats* content_specific_stats =
+      &content_specific_stats_[content_type];
   ++stats_.frames_decoded;
   if (qp) {
     if (!stats_.qp_sum) {
       if (stats_.frames_decoded != 1) {
         LOG(LS_WARNING)
             << "Frames decoded was not 1 when first qp value was received.";
         stats_.frames_decoded = 1;
       }
       stats_.qp_sum = rtc::Optional<uint64_t>(0);
     }
     *stats_.qp_sum += *qp;
+    content_specific_stats->qp_counter.Add(*qp);
   } else if (stats_.qp_sum) {
     LOG(LS_WARNING)
         << "QP sum was already set and no QP was given for a frame.";
     stats_.qp_sum = rtc::Optional<uint64_t>();
   }
   last_content_type_ = content_type;
   decode_fps_estimator_.Update(1, now);
   if (last_decoded_frame_time_ms_) {
     int64_t interframe_delay_ms = now - *last_decoded_frame_time_ms_;
     RTC_DCHECK_GE(interframe_delay_ms, 0);
     interframe_delay_max_moving_.Add(interframe_delay_ms, now);
-    if (last_content_type_ == VideoContentType::SCREENSHARE) {
-      interframe_delay_counter_screenshare_.Add(interframe_delay_ms);
-      if (interframe_delay_max_ms_screenshare_ < interframe_delay_ms) {
-        interframe_delay_max_ms_screenshare_ = interframe_delay_ms;
-      }
-    } else {
-      interframe_delay_counter_video_.Add(interframe_delay_ms);
-      if (interframe_delay_max_ms_video_ < interframe_delay_ms) {
-        interframe_delay_max_ms_video_ = interframe_delay_ms;
-      }
-    }
+    content_specific_stats->interframe_delay_counter.Add(interframe_delay_ms);
+    content_specific_stats->flow_duration_ms += interframe_delay_ms;
   }
   last_decoded_frame_time_ms_.emplace(now);
 }
 
 void ReceiveStatisticsProxy::OnRenderedFrame(const VideoFrame& frame) {
   int width = frame.width();
   int height = frame.height();
   RTC_DCHECK_GT(width, 0);
   RTC_DCHECK_GT(height, 0);
   uint64_t now = clock_->TimeInMilliseconds();
-
   rtc::CritScope lock(&crit_);
+  ContentSpecificStats* content_specific_stats =
+      &content_specific_stats_[last_content_type_];
   renders_fps_estimator_.Update(1, now);
   ++stats_.frames_rendered;
   stats_.width = width;
   stats_.height = height;
-  render_width_counter_.Add(width);
-  render_height_counter_.Add(height);
   render_fps_tracker_.AddSamples(1);
   render_pixel_tracker_.AddSamples(sqrt(width * height));
+  content_specific_stats->received_width.Add(width);
+  content_specific_stats->received_height.Add(height);
 
   if (frame.ntp_time_ms() > 0) {
     int64_t delay_ms = clock_->CurrentNtpInMilliseconds() - frame.ntp_time_ms();
     if (delay_ms >= 0) {
-      if (last_content_type_ == VideoContentType::SCREENSHARE) {
-        e2e_delay_max_ms_screenshare_ =
-            std::max(delay_ms, e2e_delay_max_ms_screenshare_);
-        e2e_delay_counter_screenshare_.Add(delay_ms);
-      } else {
-        e2e_delay_max_ms_video_ = std::max(delay_ms, e2e_delay_max_ms_video_);
-        e2e_delay_counter_video_.Add(delay_ms);
-      }
+      content_specific_stats->e2e_delay_counter.Add(delay_ms);
     }
   }
 }
 
 void ReceiveStatisticsProxy::OnSyncOffsetUpdated(int64_t sync_offset_ms,
                                                  double estimated_freq_khz) {
   rtc::CritScope lock(&crit_);
   sync_offset_counter_.Add(std::abs(sync_offset_ms));
   stats_.sync_offset_ms = sync_offset_ms;
 
   const double kMaxFreqKhz = 10000.0;
   int offset_khz = kMaxFreqKhz;
   // Should not be zero or negative. If so, report max.
   if (estimated_freq_khz < kMaxFreqKhz && estimated_freq_khz > 0.0)
     offset_khz = static_cast<int>(std::fabs(estimated_freq_khz - 90.0) + 0.5);
 
   freq_offset_counter_.Add(offset_khz);
 }
 
 void ReceiveStatisticsProxy::OnReceiveRatesUpdated(uint32_t bitRate,
                                                    uint32_t frameRate) {
 }
 
 void ReceiveStatisticsProxy::OnCompleteFrame(bool is_keyframe,
-                                             size_t size_bytes) {
+                                             size_t size_bytes,
+                                             VideoContentType content_type) {
   rtc::CritScope lock(&crit_);
   if (is_keyframe)
     ++stats_.frame_counts.key_frames;
   else
     ++stats_.frame_counts.delta_frames;
 
+  ContentSpecificStats* content_specific_stats =
+      &content_specific_stats_[content_type];
+
+  content_specific_stats->total_media_bytes += size_bytes;
+  if (is_keyframe)
+    ++content_specific_stats->frame_counts.key_frames;
+  else
+    ++content_specific_stats->frame_counts.delta_frames;

[sprang_webrtc, 2017/08/28 16:25:02]

nit: Use {} for if/else

[ilnik, 2017/08/29 07:56:27]

Done.

+
   int64_t now_ms = clock_->TimeInMilliseconds();
   frame_window_.insert(std::make_pair(now_ms, size_bytes));
   UpdateFramerate(now_ms);
 }
 
 void ReceiveStatisticsProxy::OnFrameCountsUpdated(
     const FrameCounts& frame_counts) {
   rtc::CritScope lock(&crit_);
   stats_.frame_counts = frame_counts;
 }
@@ skipping 20 matching lines @@
   // TODO(sprang): Figure out any other state that should be reset.
 
   rtc::CritScope lock(&crit_);
   // Don't report inter-frame delay if stream was paused.
   last_decoded_frame_time_ms_.reset();
 }
 
 void ReceiveStatisticsProxy::SampleCounter::Add(int sample) {
   sum += sample;
   ++num_samples;
+  if (!max || *max < sample) {

[sprang_webrtc, 2017/08/28 16:25:02]

nit: maybe more readable with if (!max || sample > *max)

[ilnik, 2017/08/29 07:56:26]

Done.

+    max.emplace(sample);
+  }
+}
+
+void ReceiveStatisticsProxy::SampleCounter::Add(const SampleCounter& other) {
+  sum += other.sum;
+  num_samples += other.num_samples;
+  if (other.max && (!max || *max < *other.max))
+    max = other.max;
 }
 
 int ReceiveStatisticsProxy::SampleCounter::Avg(
     int64_t min_required_samples) const {
   if (num_samples < min_required_samples || num_samples == 0)
     return -1;
   return static_cast<int>(sum / num_samples);
 }
 
+int ReceiveStatisticsProxy::SampleCounter::Max() const {
+  if (!max) {
+    return -1;
+  } else {
+    return *max;
+  }

[sprang_webrtc, 2017/08/28 16:25:02]

nit: return max.value_or(-1);

[ilnik, 2017/08/29 07:56:27]

Done.

+}
+
 void ReceiveStatisticsProxy::SampleCounter::Reset() {
   num_samples = 0;
   sum = 0;
+  max.reset();
 }
 
 void ReceiveStatisticsProxy::OnRttUpdate(int64_t avg_rtt_ms,
                                          int64_t max_rtt_ms) {
   rtc::CritScope lock(&crit_);
   avg_rtt_ms_ = avg_rtt_ms;
 }
 
+void ReceiveStatisticsProxy::ContentSpecificStats::Add(
+    const ContentSpecificStats& other) {
+  e2e_delay_counter.Add(other.e2e_delay_counter);
+  interframe_delay_counter.Add(other.interframe_delay_counter);
+  flow_duration_ms += other.flow_duration_ms;
+  total_media_bytes += other.total_media_bytes;
+  received_height.Add(other.received_height);
+  received_width.Add(other.received_width);
+  qp_counter.Add(other.qp_counter);
+  frame_counts.key_frames += other.frame_counts.key_frames;
+  frame_counts.delta_frames += other.frame_counts.delta_frames;
+}
+
 }  // namespace webrtc