Report interframe delay sum in old GetStats

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.
  */
 
@@ skipping 533 matching lines @@
   } 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);
+    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;
       }
@@ skipping 114 matching lines @@
   sum = 0;
 }
 
 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