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

webrtc/audio/audio_send_stream.cc

 /*
  *  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 261 matching lines @@
         break;
       }
     }
   }
 
   ScopedVoEInterface<VoEBase> base(voice_engine());
   RTC_DCHECK(base->transmit_mixer());
   stats.audio_level = base->transmit_mixer()->AudioLevelFullRange();
   RTC_DCHECK_LE(0, stats.audio_level);
 
+  stats.total_input_energy = base->transmit_mixer()->GetTotalInputEnergy();
+  stats.total_input_duration = base->transmit_mixer()->GetTotalInputDuration();
+
   RTC_DCHECK(audio_state_->audio_processing());
   auto audio_processing_stats =
       audio_state_->audio_processing()->GetStatistics();
   stats.echo_delay_median_ms = audio_processing_stats.delay_median;
   stats.echo_delay_std_ms = audio_processing_stats.delay_standard_deviation;
   stats.echo_return_loss = audio_processing_stats.echo_return_loss.instant();
   stats.echo_return_loss_enhancement =
       audio_processing_stats.echo_return_loss_enhancement.instant();
   stats.residual_echo_likelihood =
       audio_processing_stats.residual_echo_likelihood;
@@ skipping 315 matching lines @@
     if (rtp_rtcp_module_->RegisterSendPayload(codec) != 0) {
       LOG(LS_ERROR) << "RegisterCngPayloadType() failed to register CN to "
                        "RTP/RTCP module";
     }
   }
 }
 
 
 }  // namespace internal
 }  // namespace webrtc