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

webrtc/api/audio_codecs/audio_encoder.h

 /*
  *  Copyright (c) 2014 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.
  */
 
 #ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_H_
 #define WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_H_
 
 #include <algorithm>
 #include <memory>
 #include <string>
 #include <vector>
 
+#include "webrtc/common_types.h"
 #include "webrtc/rtc_base/array_view.h"
 #include "webrtc/rtc_base/buffer.h"
 #include "webrtc/rtc_base/deprecation.h"
 #include "webrtc/rtc_base/optional.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class Clock;
 class RtcEventLog;
@@ skipping 167 matching lines @@
 
   // Provides overhead to this encoder to adapt. The overhead is the number of
   // bytes that will be added to each packet the encoder generates.
   virtual void OnReceivedOverhead(size_t overhead_bytes_per_packet);
 
   // To allow encoder to adapt its frame length, it must be provided the frame
   // length range that receivers can accept.
   virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
                                            int max_frame_length_ms);
 
+  // Get statistics related to audio network adaptation.
+  virtual ANAStats GetANAStats() const;
+
  protected:
   // Subclasses implement this to perform the actual encoding. Called by
   // Encode().
   virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                                  rtc::ArrayView<const int16_t> audio,
                                  rtc::Buffer* encoded) = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_H_