Add new ANA stats to GetStats() to count the number of actions taken by each controller.

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 315 matching lines @@
   stats.echo_return_loss_enhancement =
       audio_processing_stats.echo_return_loss_enhancement.instant();
   stats.residual_echo_likelihood =
       audio_processing_stats.residual_echo_likelihood;
   stats.residual_echo_likelihood_recent_max =
       audio_processing_stats.residual_echo_likelihood_recent_max;
 
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   stats.typing_noise_detected = audio_state->typing_noise_detected();
+  stats.ana_statistics = channel_proxy_->GetANAStatistics();
 
   return stats;
 }
 
 void AudioSendStream::SignalNetworkState(NetworkState state) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
 }
 
 bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   // TODO(solenberg): Tests call this function on a network thread, libjingle
@@ skipping 303 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