Adding stats that can be used to compute output audio levels.

webrtc/api/statstypes.cc

 /*
  *  Copyright 2014 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 574 matching lines @@
     case kStatsValueNameSecondaryDecodedRate:
       return "googSecondaryDecodedRate";
     case kStatsValueNameSendPacketsDiscarded:
       return "packetsDiscardedOnSend";
     case kStatsValueNameSpeechExpandRate:
       return "googSpeechExpandRate";
     case kStatsValueNameSrtpCipher:
       return "srtpCipher";
     case kStatsValueNameTargetEncBitrate:
       return "googTargetEncBitrate";
+    case kStatsValueNameTotalAudioEnergy:
+      return "totalAudioEnergy";
+    case kStatsValueNameTotalSamplesDuration:
+      return "totalSamplesDuration";
     case kStatsValueNameTransmitBitrate:
       return "googTransmitBitrate";
     case kStatsValueNameTransportType:
       return "googTransportType";
     case kStatsValueNameTrackId:
       return "googTrackId";
     case kStatsValueNameTypingNoiseState:
       return "googTypingNoiseState";
     case kStatsValueNameWritable:
       return "googWritable";
@@ skipping 177 matching lines @@
 // Looks for a report with the given |id|.  If one is not found, null
 // will be returned.
 StatsReport* StatsCollection::Find(const StatsReport::Id& id) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   Container::iterator it = std::find_if(list_.begin(), list_.end(),
       [&id](const StatsReport* r)->bool { return r->id()->Equals(id); });
   return it == list_.end() ? nullptr : *it;
 }
 
 }  // namespace webrtc