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

webrtc/call/audio_receive_stream.h

 /*
  *  Copyright (c) 2015 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 31 matching lines @@
     uint32_t packets_lost = 0;
     float fraction_lost = 0.0f;
     std::string codec_name;
     rtc::Optional<int> codec_payload_type;
     uint32_t ext_seqnum = 0;
     uint32_t jitter_ms = 0;
     uint32_t jitter_buffer_ms = 0;
     uint32_t jitter_buffer_preferred_ms = 0;
     uint32_t delay_estimate_ms = 0;
     int32_t audio_level = -1;
+    // See description of "totalAudioEnergy" in the WebRTC stats spec:
+    // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
+    double total_output_energy = 0.0;
+    double total_output_duration = 0.0;
     float expand_rate = 0.0f;
     float speech_expand_rate = 0.0f;
     float secondary_decoded_rate = 0.0f;
     float accelerate_rate = 0.0f;
     float preemptive_expand_rate = 0.0f;
     int32_t decoding_calls_to_silence_generator = 0;
     int32_t decoding_calls_to_neteq = 0;
     int32_t decoding_normal = 0;
     int32_t decoding_plc = 0;
     int32_t decoding_cng = 0;
@@ skipping 73 matching lines @@
   virtual void SetGain(float gain) = 0;
 
   virtual std::vector<RtpSource> GetSources() const = 0;
 
  protected:
   virtual ~AudioReceiveStream() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_