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/rtc_base/trace_event.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 {
@@ skipping 12 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";
+  int simulcast_id = content_type.GetSimulcastId();
+  if (content_type.IsScreenshare()) {
+    ss << ".Screenshare";
+  }
+  if (simulcast_id > 0) {
+    ss << "_S" << simulcast_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() {
+  // Set synthetic field trial based on received experiment id.
+  // Since experiment is set at the send side and is not changed during the
+  // call, any received content type may be used.

[sprang_webrtc, 2017/07/26 14:13:43]

Unfortunately this is not necessarily correct. I think we need to update these
histograms as soon as the content type changes. :/

[ilnik, 2017/07/26 14:49:48]

This is a big problem, I realize now. What if several senders have different
experiment groups. This method should be one big critical section between all
receive statistic proxies.

Experiment should be set to "" at the end of the method. This way it will be
something strange only during reporting these values (I assume nothing else will
report UMA metrics? How about audio?

+  int experiment_id = last_content_type_.GetExperimentId();
+  if (experiment_id != 0) {  // 0 means no experiment is active.
+    std::stringstream group_name;
+    group_name << "Group-" << experiment_id - 1;
+    field_trial::RegisterSyntheticFieldTrial(
+        AlrDetector::kScreenshareProbingBweExperimentNameOnReceiveSide,
+        group_name.str());
+  } else {
+    // No experiment group specified.
+    field_trial::RegisterSyntheticFieldTrial(
+        AlrDetector::kScreenshareProbingBweExperimentNameOnReceiveSide, "");
+  }
+
   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;
+  for (auto it : content_specific_stats_) {

[sprang_webrtc, 2017/07/26 14:13:43]

const auto& it ?

+    auto content_type = it.first;
+    auto stats = it.second;

[sprang_webrtc, 2017/07/26 14:13:43]

auto is good for |it| here, since its type is kinda verbose.
For |content_type| and |stats| I would prefer explicit types however. It's not
obvious here what type |it.second| has...

+    std::string uma_prefix = UmaPrefixForContentType(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",
+                                  e2e_delay_ms);
+      LOG(LS_INFO) << uma_prefix + ".EndToEndDelayInMs " << e2e_delay_ms;
+    }
+    int e2e_delay_max_ms = stats.e2e_delay_counter.Max();
+
+    if (e2e_delay_max_ms != -1) {
+      RTC_HISTOGRAMS_COUNTS_100000(idx, uma_prefix + ".EndToEndDelayMaxInMs",
+                                   e2e_delay_max_ms);
+      LOG(LS_INFO) << uma_prefix + ".EndToEndDelayMaxInMs " << 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",
+                                  interframe_delay_ms);
+      LOG(LS_INFO) << uma_prefix + ".InterframeDelayInMs "
+                   << interframe_delay_ms;
+    }
+    int interframe_delay_max_ms = stats.interframe_delay_counter.Max();
+    if (interframe_delay_max_ms != -1) {
+      RTC_HISTOGRAMS_COUNTS_10000(idx, uma_prefix + ".InterframeDelayMaxInMs",
+                                  interframe_delay_max_ms);
+      LOG(LS_INFO) << uma_prefix + ".InterframeDelayMaxInMs "
+                   << interframe_delay_max_ms;
+    }
   }
 
-  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);
-  }
-
-  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);
-  }
-
-  int interframe_delay_max_ms_screenshare =
-      interframe_delay_max_ms_screenshare_;
-  if (interframe_delay_max_ms_screenshare != -1) {
-    RTC_HISTOGRAM_COUNTS_10000(
-        "WebRTC.Video.Screenshare.InterframeDelayMaxInMs",
-        interframe_delay_ms_screenshare);
-  }
-
-  int interframe_delay_max_ms_video = interframe_delay_max_ms_video_;
-  if (interframe_delay_max_ms_video != -1) {
-    RTC_HISTOGRAM_COUNTS_10000(
-        "WebRTC.Video.InterframeDelayMaxInMs",
-        interframe_delay_ms_video);
-  }
-
-
   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(
@@ skipping 312 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_);
     }
+    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);
     }
   }
 
   TRACE_EVENT_INSTANT2("webrtc_stats", "WebRTC.Video.ReceivedWidthInPixels",
                        "width", width, "ssrc", stats_.ssrc);
   TRACE_EVENT_INSTANT2("webrtc_stats", "WebRTC.Video.ReceivedHeightInPixels",
                        "height", height, "ssrc", stats_.ssrc);
   TRACE_EVENT_INSTANT1("webrtc_stats", "WebRTC.Video.OnRenderedFrame",
                        "ssrc", stats_.ssrc);
 }
@@ skipping 50 matching lines @@
   if (codec_specific_info->codecType == kVideoCodecVP8) {
     qp_counters_.vp8.Add(encoded_image.qp_);
     rtc::CritScope lock(&crit_);
     qp_sample_.Add(encoded_image.qp_);
   }
 }
 
 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 static_cast<int>(*max);
+  }

[sprang_webrtc, 2017/07/26 14:13:43]

return max.value_or(-1);

+}
+
 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