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

webrtc/voice_engine/channel.h

 /*
  *  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 250 matching lines @@
   int GetSpeechOutputLevel() const;
   int GetSpeechOutputLevelFullRange() const;
   // See description of "totalAudioEnergy" in the WebRTC stats spec:
   // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
   double GetTotalOutputEnergy() const;
   double GetTotalOutputDuration() const;
 
   // Stats.
   int GetNetworkStatistics(NetworkStatistics& stats);
   void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
+  ANAStats GetANAStatistics() const;
 
   // Audio+Video Sync.
   uint32_t GetDelayEstimate() const;
   int SetMinimumPlayoutDelay(int delayMs);
   int GetPlayoutTimestamp(unsigned int& timestamp);
   int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
 
   // DTMF.
   int SendTelephoneEventOutband(int event, int duration_ms);
   int SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency);
@@ skipping 276 matching lines @@
 
   bool encoder_queue_is_active_ GUARDED_BY(encoder_queue_lock_) = false;
 
   rtc::TaskQueue* encoder_queue_ = nullptr;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_