Fix the video buffer size should take rtt into consideration

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.
  */
 
@@ skipping 256 matching lines @@
 
 int FrameBuffer::InsertFrame(std::unique_ptr<FrameObject> frame) {
   TRACE_EVENT0("webrtc", "FrameBuffer::InsertFrame");
   RTC_DCHECK(frame);
   if (stats_callback_)
     stats_callback_->OnCompleteFrame(frame->num_references == 0, frame->size());
   FrameKey key(frame->picture_id, frame->spatial_layer);
 
   rtc::CritScope lock(&crit_);
 
+  if (frame->delayed_by_retransmission())
+    jitter_estimator_->FrameNacked();

[philipel2, 2017/07/24 14:17:22]

I think we should move this inside the if block on line 131.

+
   int last_continuous_picture_id =
       last_continuous_frame_it_ == frames_.end()
           ? -1
           : last_continuous_frame_it_->first.picture_id;
 
   if (!ValidReferences(*frame)) {
     LOG(LS_WARNING) << "Frame with (picture_id:spatial_id) (" << key.picture_id
                     << ":" << static_cast<int>(key.spatial_layer)
                     << ") has invalid frame references, dropping frame.";
     return last_continuous_picture_id;
@@ skipping 260 matching lines @@
   frames_.clear();
   last_decoded_frame_it_ = frames_.end();
   last_continuous_frame_it_ = frames_.end();
   next_frame_it_ = frames_.end();
   num_frames_history_ = 0;
   num_frames_buffered_ = 0;
 }
 
 }  // namespace video_coding
 }  // namespace webrtc