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

webrtc/call/audio_send_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 29 matching lines @@
     int64_t bytes_sent = 0;
     int32_t packets_sent = 0;
     int32_t packets_lost = -1;
     float fraction_lost = -1.0f;
     std::string codec_name;
     rtc::Optional<int> codec_payload_type;
     int32_t ext_seqnum = -1;
     int32_t jitter_ms = -1;
     int64_t rtt_ms = -1;
     int32_t audio_level = -1;
+    // See description of "totalAudioEnergy" in the WebRTC stats spec.
+    double total_input_energy = 0.0;
+    double total_input_duration = 0.0;
     float aec_quality_min = -1.0f;
     int32_t echo_delay_median_ms = -1;
     int32_t echo_delay_std_ms = -1;
     int32_t echo_return_loss = -100;
     int32_t echo_return_loss_enhancement = -100;
     float residual_echo_likelihood = -1.0f;
     float residual_echo_likelihood_recent_max = -1.0f;
     bool typing_noise_detected = false;
   };
 
@@ skipping 82 matching lines @@
   virtual void SetMuted(bool muted) = 0;
 
   virtual Stats GetStats() const = 0;
 
  protected:
   virtual ~AudioSendStream() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_AUDIO_SEND_STREAM_H_