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

webrtc/audio/audio_receive_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 169 matching lines @@
   if (codec_inst.pltype != -1) {
     stats.codec_name = codec_inst.plname;
     stats.codec_payload_type = rtc::Optional<int>(codec_inst.pltype);
   }
   stats.ext_seqnum = call_stats.extendedMax;
   if (codec_inst.plfreq / 1000 > 0) {
     stats.jitter_ms = call_stats.jitterSamples / (codec_inst.plfreq / 1000);
   }
   stats.delay_estimate_ms = channel_proxy_->GetDelayEstimate();
   stats.audio_level = channel_proxy_->GetSpeechOutputLevelFullRange();
+  stats.total_output_energy = channel_proxy_->GetTotalOutputEnergy();
+  stats.total_output_duration = channel_proxy_->GetTotalOutputDuration();
 
   // Get jitter buffer and total delay (alg + jitter + playout) stats.
   auto ns = channel_proxy_->GetNetworkStatistics();
   stats.jitter_buffer_ms = ns.currentBufferSize;
   stats.jitter_buffer_preferred_ms = ns.preferredBufferSize;
   stats.expand_rate = Q14ToFloat(ns.currentExpandRate);
   stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate);
   stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate);
   stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate);
   stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate);
@@ skipping 139 matching lines @@
 int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) {
   ScopedVoEInterface<VoEBase> base(voice_engine());
   if (playout) {
     return base->StartPlayout(config_.voe_channel_id);
   } else {
     return base->StopPlayout(config_.voe_channel_id);
   }
 }
 }  // namespace internal
 }  // namespace webrtc