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

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.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.
  */
 
@@ skipping 98 matching lines @@
       float uplink_packet_loss_fraction) override;
   void OnReceivedUplinkRecoverablePacketLossFraction(
       float uplink_recoverable_packet_loss_fraction) override;
   void OnReceivedUplinkBandwidth(
       int target_audio_bitrate_bps,
       rtc::Optional<int64_t> bwe_period_ms) override;
   void OnReceivedRtt(int rtt_ms) override;
   void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
   void SetReceiverFrameLengthRange(int min_frame_length_ms,
                                    int max_frame_length_ms) override;
+  ANAStats GetANAStats() const override;
   rtc::ArrayView<const int> supported_frame_lengths_ms() const {
     return config_.supported_frame_lengths_ms;
   }
 
   // Getters for testing.
   float packet_loss_rate() const { return packet_loss_rate_; }
   AudioEncoderOpusConfig::ApplicationMode application() const {
     return config_.application;
   }
   bool fec_enabled() const { return config_.fec_enabled; }
@@ skipping 43 matching lines @@
   rtc::Optional<size_t> overhead_bytes_per_packet_;
   const std::unique_ptr<SmoothingFilter> bitrate_smoother_;
   rtc::Optional<int64_t> bitrate_smoother_last_update_time_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_