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

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 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 2875 matching lines @@
 }
 
 int Channel::GetNetworkStatistics(NetworkStatistics& stats) {
   return audio_coding_->GetNetworkStatistics(&stats);
 }
 
 void Channel::GetDecodingCallStatistics(AudioDecodingCallStats* stats) const {
   audio_coding_->GetDecodingCallStatistics(stats);
 }
 
+ANAStats Channel::GetANAStatistics() const {
+  return audio_coding_->GetANAStats();
+}
+
 uint32_t Channel::GetDelayEstimate() const {
   rtc::CritScope lock(&video_sync_lock_);
   return audio_coding_->FilteredCurrentDelayMs() + playout_delay_ms_;
 }
 
 int Channel::SetMinimumPlayoutDelay(int delayMs) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetMinimumPlayoutDelay()");
   if ((delayMs < kVoiceEngineMinMinPlayoutDelayMs) ||
       (delayMs > kVoiceEngineMaxMinPlayoutDelayMs)) {
@@ skipping 259 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc