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

webrtc/api/stats/rtcstats_objects.h

 /*
  *  Copyright 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 254 matching lines @@
   RTCStatsMember<uint32_t> frames_decoded;
   RTCStatsMember<uint32_t> frames_dropped;
   // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/659137
   RTCStatsMember<uint32_t> frames_corrupted;
   // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/659137
   RTCStatsMember<uint32_t> partial_frames_lost;
   // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/659137
   RTCStatsMember<uint32_t> full_frames_lost;
   // Audio-only members
   RTCStatsMember<double> audio_level;
+  RTCStatsMember<double> total_audio_energy;
+  RTCStatsMember<double> total_samples_duration;
   RTCStatsMember<double> echo_return_loss;
   RTCStatsMember<double> echo_return_loss_enhancement;
 };
 
 // https://w3c.github.io/webrtc-stats/#pcstats-dict*
 class RTCPeerConnectionStats final : public RTCStats {
  public:
   WEBRTC_RTCSTATS_DECL();
 
   RTCPeerConnectionStats(const std::string& id, int64_t timestamp_us);
@@ skipping 120 matching lines @@
   // TODO(hbos): Support enum types? "RTCStatsMember<RTCDtlsTransportState>"?
   RTCStatsMember<std::string> dtls_state;
   RTCStatsMember<std::string> selected_candidate_pair_id;
   RTCStatsMember<std::string> local_certificate_id;
   RTCStatsMember<std::string> remote_certificate_id;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_STATS_RTCSTATS_OBJECTS_H_