Report interframe delay sum in old GetStats

webrtc/api/statstypes.h

 /*
  *  Copyright 2012 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 89 matching lines @@
     kStatsValueNameActiveConnection,
     kStatsValueNameAecDivergentFilterFraction,
     kStatsValueNameAudioInputLevel,
     kStatsValueNameAudioOutputLevel,
     kStatsValueNameBytesReceived,
     kStatsValueNameBytesSent,
     kStatsValueNameCodecImplementationName,
     kStatsValueNameDataChannelId,
     kStatsValueNameFramesDecoded,
     kStatsValueNameFramesEncoded,
+    kStatsValueNameInterframeDelaySumMs,
     kStatsValueNameMediaType,
     kStatsValueNamePacketsLost,
     kStatsValueNamePacketsReceived,
     kStatsValueNamePacketsSent,
     kStatsValueNameProtocol,
     kStatsValueNameQpSum,
     kStatsValueNameReceiving,
     kStatsValueNameSelectedCandidatePairId,
     kStatsValueNameSsrc,
     kStatsValueNameState,
@@ skipping 301 matching lines @@
   StatsReport* Find(const StatsReport::Id& id);
 
  private:
   Container list_;
   rtc::ThreadChecker thread_checker_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_STATSTYPES_H_