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 11 matching lines @@
 const int kHighFpsThreshold = 14;
 // For qp and fps variance:
 // Low means low enough to be good, high means high enough to be bad
 const int kLowQpThresholdVp8 = 60;
 const int kHighQpThresholdVp8 = 70;
 const int kLowVarianceThreshold = 1;
 const int kHighVarianceThreshold = 2;
 
 // 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;
+  }
+  return ss.str();
+}
+
+std::string UmaPrefixForExperiment(VideoContentType content_type) {
+  std::stringstream ss;
+  ss << "WebRTC.Video.AlrExperiment";
+  if (content_type.IsScreenshare()) {
+    ss << ".Screenshare";
+  }
+  return ss.str();
+}
+
+std::string UmaSuffixForExperiment(VideoContentType content_type) {
+  std::stringstream ss;
+  int experiment_id = content_type.GetExperimentId();
+  if (experiment_id > 0) {
+    ss << ".Group" << 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),
       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() {
+  // Use to report experiment-specific metrics.
+  int experiment_id = last_content_type_.GetExperimentId();
+
   RTC_HISTOGRAM_COUNTS_100000(
       "WebRTC.Video.ReceiveStreamLifetimeInSeconds",
       (clock_->TimeInMilliseconds() - start_ms_) / 1000);
 
   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",
@@ skipping 70 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;
+  int avg_interframe_delay = 0;
+  int max_interframe_delay = -1;
+  int interframe_delay_slices = 0;
+  int avg_e2e_delay = 0;
+  int max_e2e_delay = -1;
+  int e2e_delay_slices = 0;
+
+  // Slice on content type and simulcast id.
+  for (auto it : content_specific_stats_) {
+    auto content_type = it.first;
+    auto stats = it.second;
+    std::string uma_prefix = UmaPrefixForContentType(content_type);
+    std::string uma_suffix = UmaSuffixForContentType(content_type);
+    // The same line of code can't report different histograms because of how
+    // macro is done. Additional index needed to be different for all different
+    // names. Since experiment can't change during call, we could report 4*2
+    // different content types (4 simulcast_ids x 2 content types).
+    // Conveniently all of these are stored in 3 lower bits in VideoContentType.
+    int idx = content_type.content_type & 0x07;
+    int e2e_delay_ms = stats.e2e_delay_counter.Avg(kMinRequiredSamples);
+    if (e2e_delay_ms != -1) {
+      RTC_HISTOGRAMS_COUNTS_10000(
+          idx, uma_prefix + ".EndToEndDelayInMs" + uma_suffix, e2e_delay_ms);
+      LOG(LS_INFO) << uma_prefix << ".EndToEndDelayInMs" << uma_suffix << " "
+                   << e2e_delay_ms;
+      avg_e2e_delay += e2e_delay_ms;
+      ++e2e_delay_slices;
+    }
+    int e2e_delay_max_ms = stats.e2e_delay_counter.Max();
+
+    if (e2e_delay_max_ms != -1 && e2e_delay_ms != -1) {
+      RTC_HISTOGRAMS_COUNTS_100000(
+          idx, uma_prefix + ".EndToEndDelayMaxInMs" + uma_suffix,
+          e2e_delay_max_ms);
+      LOG(LS_INFO) << uma_prefix << ".EndToEndDelayMaxInMs" << uma_suffix << " "
+                   << e2e_delay_max_ms;
+      if (e2e_delay_max_ms > max_e2e_delay)
+        max_e2e_delay = e2e_delay_max_ms;
+    }
+    int interframe_delay_ms =
+        stats.interframe_delay_counter.Avg(kMinRequiredSamples);
+    if (interframe_delay_ms != -1) {
+      RTC_HISTOGRAMS_COUNTS_10000(
+          idx, uma_prefix + ".InterframeDelayInMs" + uma_suffix,
+          interframe_delay_ms);
+      LOG(LS_INFO) << uma_prefix << ".InterframeDelayInMs" << uma_suffix << " "
+                   << interframe_delay_ms;
+      avg_interframe_delay += interframe_delay_ms;
+      ++interframe_delay_slices;
+    }
+    int interframe_delay_max_ms = stats.interframe_delay_counter.Max();
+    if (interframe_delay_max_ms != -1 && interframe_delay_ms != -1) {
+      RTC_HISTOGRAMS_COUNTS_10000(
+          idx, uma_prefix + ".InterframeDelayMaxInMs" + uma_suffix,
+          interframe_delay_max_ms);
+      LOG(LS_INFO) << uma_prefix << ".InterframeDelayMaxInMs" << uma_suffix
+                   << " " << interframe_delay_max_ms;
+      if (interframe_delay_max_ms > max_interframe_delay)
+        max_interframe_delay = interframe_delay_max_ms;
+    }
+  }
+  if (e2e_delay_slices > 0) {
+    avg_e2e_delay /= e2e_delay_slices;
+  } else {
+    avg_e2e_delay = -1;
+  }
+  if (interframe_delay_slices > 0) {
+    avg_interframe_delay /= interframe_delay_slices;
+  } else {
+    avg_interframe_delay = -1;
   }
 
-  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);
-  }
-
-  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_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 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 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_);
-  }
+  int media_bitrate = 0;
 
   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));
+    media_bitrate =
+        static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000);
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.MediaBitrateReceivedInKbps",
+                               media_bitrate);
     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 11 matching lines @@
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsSentPerMinute",
                                counters.fir_packets * 60 / elapsed_sec);
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsSentPerMinute",
                                counters.pli_packets * 60 / elapsed_sec);
     if (counters.nack_requests > 0) {
       RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.UniqueNackRequestsSentInPercent",
                                counters.UniqueNackRequestsInPercent());
     }
   }
 
+  // Report metrics sliced on ALR experiment group.
+  if (experiment_id > 0 && content_specific_stats_.size() > 0) {
+    VideoContentType content_type = content_specific_stats_.begin()->first;
+    std::string uma_prefix = UmaPrefixForExperiment(content_type);
+    std::string uma_suffix = UmaSuffixForExperiment(content_type);
+    if (avg_e2e_delay != -1) {
+      RTC_HISTOGRAM_COUNTS_10000(uma_prefix + ".EndToEndDelayInMs" + uma_suffix,
+                                 avg_e2e_delay);
+      LOG(LS_INFO) << uma_prefix << ".EndToEndDelayInMs" << uma_suffix << " "
+                   << avg_e2e_delay;
+    }
+    if (max_e2e_delay != -1) {
+      RTC_HISTOGRAM_COUNTS_100000(
+          uma_prefix + ".EndToEndDelayMaxInMs" + uma_suffix, max_e2e_delay);
+      LOG(LS_INFO) << uma_prefix << ".EndToEndDelayMaxInMs" << uma_suffix << " "
+                   << max_e2e_delay;
+    }
+    if (avg_interframe_delay != -1) {
+      RTC_HISTOGRAM_COUNTS_10000(
+          uma_prefix + ".InterframeDelayInMs" + uma_suffix,
+          avg_interframe_delay);
+      LOG(LS_INFO) << uma_prefix << ".InterframeDelayInMs" << uma_suffix << " "
+                   << avg_interframe_delay;
+    }
+    if (max_interframe_delay != -1) {
+      RTC_HISTOGRAM_COUNTS_10000(
+          uma_prefix + ".InterframeDelayMaxInMs" + uma_suffix,
+          max_interframe_delay);
+      LOG(LS_INFO) << uma_prefix << ".InterframeDelayMaxInMs" << uma_suffix
+                   << " " << max_interframe_delay;
+    }
+    if (media_bitrate != 0) {
+      RTC_HISTOGRAM_COUNTS_10000(
+          uma_prefix + ".MediaBitrateReceivedInKbps" + uma_suffix,
+          media_bitrate);
+      LOG(LS_INFO) << uma_prefix << ".MediaBitrateReceivedInKbps" << uma_suffix
+                   << " " << media_bitrate;
+    }
+  }
+

[sprang_webrtc, 2017/08/24 09:13:16]

Why isn't this reported in the same way as for the other content type aspects
above?

[ilnik, 2017/08/25 12:35:07]

Because currently we can't get slice on both simulcast id and experiment id.
Thus, if the are slicing on experiment id, we aggregate all the different
spatial layers. There's a problem if the stream received both screenshare and
video during the call, but it never happens currently.

   if (num_certain_states_ >= kBadCallMinRequiredSamples) {
     RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BadCall.Any",
                              100 * num_bad_states_ / num_certain_states_);
   }
   rtc::Optional<double> fps_fraction =
       fps_threshold_.FractionHigh(kBadCallMinRequiredSamples);
   if (fps_fraction) {
     RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BadCall.FrameRate",
                              static_cast<int>(100 * (1 - *fps_fraction)));
   }
@@ skipping 245 matching lines @@
     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);
     stats_.interframe_delay_sum_ms += interframe_delay_ms;
-    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;
-      }
+    auto it = content_specific_stats_.find(last_content_type_);
+    if (it == content_specific_stats_.end()) {
+      content_specific_stats_[last_content_type_] = ContentSpecificStats();
+      it = content_specific_stats_.find(last_content_type_);
     }

[sprang_webrtc, 2017/08/24 09:13:16]

I think you can just do 

auto it = content_specific_stats_.emplace(last_content_content_type_,
ContentSpecificStats()).first;

Or use a utility method for getting with lazy initalization.

[ilnik, 2017/08/25 12:35:07]

Redone using plain [] operator. According to documentation, if there's no entry
for the key, default will be constructed and reference to that returned. Just
that I need.

+    ContentSpecificStats* stats = &it->second;
+    stats->interframe_delay_counter.Add(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_);
   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));
 
   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);
+      auto it = content_specific_stats_.find(last_content_type_);
+      if (it == content_specific_stats_.end()) {
+        content_specific_stats_[last_content_type_] = ContentSpecificStats();
+        it = content_specific_stats_.find(last_content_type_);
       }
+      ContentSpecificStats* stats = &it->second;
+      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;
 
@@ skipping 51 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) {
+    max.emplace(sample);
+  }
 }
 
 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;
+  }
+}
+
 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;
 }
 
 }  // namespace webrtc