Now run EndToEndTest with the WebRTC-NewVideoJitterBuffer experiment.

webrtc/modules/video_coding/frame_buffer2.cc

 /*
  *  Copyright (c) 2016 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/modules/video_coding/frame_buffer2.h"
 
 #include <algorithm>
 #include <cstring>
 #include <queue>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/video_coding/jitter_estimator.h"
 #include "webrtc/modules/video_coding/timing.h"
 #include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/metrics.h"
 
 namespace webrtc {
 namespace video_coding {
 
 namespace {
 // Max number of frames the buffer will hold.
 constexpr int kMaxFramesBuffered = 600;
 
 // Max number of decoded frame info that will be saved.
 constexpr int kMaxFramesHistory = 50;
 }  // namespace
 
 FrameBuffer::FrameBuffer(Clock* clock,
                          VCMJitterEstimator* jitter_estimator,
                          VCMTiming* timing)
     : clock_(clock),
       new_countinuous_frame_event_(false, false),
       jitter_estimator_(jitter_estimator),
       timing_(timing),
       inter_frame_delay_(clock_->TimeInMilliseconds()),
       last_decoded_frame_it_(frames_.end()),
       last_continuous_frame_it_(frames_.end()),
       num_frames_history_(0),
       num_frames_buffered_(0),
       stopped_(false),
-      protection_mode_(kProtectionNack) {}
+      protection_mode_(kProtectionNack),
+      num_total_frames_(0),
+      num_key_frames_(0) {}
+
+FrameBuffer::~FrameBuffer() {
+  UpdateHistograms();
+}
 
 FrameBuffer::ReturnReason FrameBuffer::NextFrame(
     int64_t max_wait_time_ms,
     std::unique_ptr<FrameObject>* frame_out) {
   int64_t latest_return_time = clock_->TimeInMilliseconds() + max_wait_time_ms;
   int64_t wait_ms = max_wait_time_ms;
   FrameMap::iterator next_frame_it;
 
   do {
     int64_t now_ms = clock_->TimeInMilliseconds();
@@ skipping 90 matching lines @@
 void FrameBuffer::Stop() {
   rtc::CritScope lock(&crit_);
   stopped_ = true;
   new_countinuous_frame_event_.Set();
 }
 
 int FrameBuffer::InsertFrame(std::unique_ptr<FrameObject> frame) {
   rtc::CritScope lock(&crit_);
   RTC_DCHECK(frame);
 
+  ++num_total_frames_;
+  if (frame->num_references == 0)
+    ++num_key_frames_;
+
   FrameKey key(frame->picture_id, frame->spatial_layer);
   int last_continuous_picture_id =
       last_continuous_frame_it_ == frames_.end()
           ? -1
           : last_continuous_frame_it_->first.picture_id;
 
   if (num_frames_buffered_ >= kMaxFramesBuffered) {
     LOG(LS_WARNING) << "Frame with (picture_id:spatial_id) (" << key.picture_id
                     << ":" << static_cast<int>(key.spatial_layer)
                     << ") could not be inserted due to the frame "
@@ skipping 178 matching lines @@
     RTC_DCHECK_LE(ref_info->second.num_missing_continuous,
                   ref_info->second.num_missing_decodable);
   }
 
   RTC_DCHECK_LE(info->second.num_missing_continuous,
                 info->second.num_missing_decodable);
 
   return true;
 }
 
+void FrameBuffer::UpdateHistograms() const {
+  rtc::CritScope lock(&crit_);
+  if (num_total_frames_ > 0) {
+    int key_frames_permille = (static_cast<float>(num_key_frames_) * 1000.0f /
+                                   static_cast<float>(num_total_frames_) +
+                               0.5f);
+    RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesReceivedInPermille",
+                              key_frames_permille);
+  }
+}
+
 }  // namespace video_coding
 }  // namespace webrtc