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 596 matching lines @@
   rtc::Optional<int> codec_payload_type;
   std::vector<SsrcReceiverInfo> local_stats;
   std::vector<SsrcSenderInfo> remote_stats;
 };
 
 struct VoiceSenderInfo : public MediaSenderInfo {
   VoiceSenderInfo()
       : ext_seqnum(0),
         jitter_ms(0),
         audio_level(0),
+        total_input_energy(0.0),
+        total_input_duration(0.0),
         aec_quality_min(0.0),
         echo_delay_median_ms(0),
         echo_delay_std_ms(0),
         echo_return_loss(0),
         echo_return_loss_enhancement(0),
         residual_echo_likelihood(0.0f),
         residual_echo_likelihood_recent_max(0.0f),
         typing_noise_detected(false) {}
 
   int ext_seqnum;
   int jitter_ms;
   int audio_level;
+  // See description of "totalAudioEnergy" in the WebRTC stats spec:
+  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
+  double total_input_energy;
+  double total_input_duration;
   float aec_quality_min;
   int echo_delay_median_ms;
   int echo_delay_std_ms;
   int echo_return_loss;
   int echo_return_loss_enhancement;
   float residual_echo_likelihood;
   float residual_echo_likelihood_recent_max;
   bool typing_noise_detected;
 };
 
 struct VoiceReceiverInfo : public MediaReceiverInfo {
   VoiceReceiverInfo()
       : ext_seqnum(0),
         jitter_ms(0),
         jitter_buffer_ms(0),
         jitter_buffer_preferred_ms(0),
         delay_estimate_ms(0),
         audio_level(0),
+        total_output_energy(0.0),
+        total_output_duration(0.0),
         expand_rate(0),
         speech_expand_rate(0),
         secondary_decoded_rate(0),
         accelerate_rate(0),
         preemptive_expand_rate(0),
         decoding_calls_to_silence_generator(0),
         decoding_calls_to_neteq(0),
         decoding_normal(0),
         decoding_plc(0),
         decoding_cng(0),
         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:
+  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
+  double total_output_energy;
+  double total_output_duration;
   // 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 529 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_