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

webrtc/media/base/mediachannel.h

 /*
  *  Copyright (c) 2004 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 657 matching lines @@
         decoding_plc_cng(0),
         decoding_muted_output(0),
         capture_start_ntp_time_ms(-1) {}
 
   int ext_seqnum;
   int jitter_ms;
   int jitter_buffer_ms;
   int jitter_buffer_preferred_ms;
   int delay_estimate_ms;
   int audio_level;
+  // See description of "totalAudioEnergy" in the WebRTC stats spec.
+  double total_output_energy = 0.0;
+  double total_output_duration = 0.0;
   // fraction of synthesized audio inserted through expansion.
   float expand_rate;
   // fraction of synthesized speech inserted through expansion.
   float speech_expand_rate;
   // fraction of data out of secondary decoding, including FEC and RED.
   float secondary_decoded_rate;
   // Fraction of data removed through time compression.
   float accelerate_rate;
   // Fraction of data inserted through time stretching.
   float preemptive_expand_rate;
@@ skipping 527 matching lines @@
                    const char*,
                    size_t> SignalDataReceived;
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_