Add a placeholder stat for logging the estimated residual echo likelihood.

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 218 matching lines @@
     int erl = -100;
     int erle = -100;
     int dummy1 = 0;
     int dummy2 = 0;
     error = processing->GetEchoMetrics(erl, erle, dummy1, dummy2);
     RTC_DCHECK_EQ(0, error);
     stats.echo_return_loss = erl;
     stats.echo_return_loss_enhancement = erle;
   }
 
+  // TODO(ivoc): Hook this up to the residual echo detector.
+  stats.residual_echo_likelihood = 0.0f;
+
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   stats.typing_noise_detected = audio_state->typing_noise_detected();
 
   return stats;
 }
 
 void AudioSendStream::SignalNetworkState(NetworkState state) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
@@ skipping 31 matching lines @@
 
 VoiceEngine* AudioSendStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc