Report interframe delay sum in old GetStats

webrtc/media/base/mediachannel.h

 /*
  *  Copyright (c) 2004 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 722 matching lines @@
         frame_width(0),
         frame_height(0),
         framerate_rcvd(0),
         framerate_decoded(0),
         framerate_output(0),
         framerate_render_input(0),
         framerate_render_output(0),
         frames_received(0),
         frames_decoded(0),
         frames_rendered(0),
+        interframe_delay_sum_ms(0),
         decode_ms(0),
         max_decode_ms(0),
         jitter_buffer_ms(0),
         min_playout_delay_ms(0),
         render_delay_ms(0),
         target_delay_ms(0),
         current_delay_ms(0),
         capture_start_ntp_time_ms(-1) {
   }
 
@@ skipping 10 matching lines @@
   int framerate_decoded;
   int framerate_output;
   // Framerate as sent to the renderer.
   int framerate_render_input;
   // Framerate that the renderer reports.
   int framerate_render_output;
   uint32_t frames_received;
   uint32_t frames_decoded;
   uint32_t frames_rendered;
   rtc::Optional<uint64_t> qp_sum;
+  uint64_t interframe_delay_sum_ms;
 
   // All stats below are gathered per-VideoReceiver, but some will be correlated
   // across MediaStreamTracks.  NOTE(hta): when sinking stats into per-SSRC
   // structures, reflect this in the new layout.
 
   // Current frame decode latency.
   int decode_ms;
   // Maximum observed frame decode latency.
   int max_decode_ms;
   // Jitter (network-related) latency.
@@ skipping 423 matching lines @@
                    const char*,
                    size_t> SignalDataReceived;
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_