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

webrtc/call/audio_send_stream.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_CALL_AUDIO_SEND_STREAM_H_
 #define WEBRTC_CALL_AUDIO_SEND_STREAM_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
+#include "webrtc/api/audio_codecs/audio_encoder.h"
 #include "webrtc/api/audio_codecs/audio_encoder_factory.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/api/call/transport.h"
 #include "webrtc/api/optional.h"
 #include "webrtc/api/rtpparameters.h"
 #include "webrtc/call/rtp_config.h"
 #include "webrtc/rtc_base/scoped_ref_ptr.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ skipping 26 matching lines @@
     double total_input_energy = 0.0;
     double total_input_duration = 0.0;
     float aec_quality_min = -1.0f;
     int32_t echo_delay_median_ms = -1;
     int32_t echo_delay_std_ms = -1;
     int32_t echo_return_loss = -100;
     int32_t echo_return_loss_enhancement = -100;
     float residual_echo_likelihood = -1.0f;
     float residual_echo_likelihood_recent_max = -1.0f;
     bool typing_noise_detected = false;
+    ANAStats ana_statistics;
   };
 
   struct Config {
     Config() = delete;
     explicit Config(Transport* send_transport);
     ~Config();
     std::string ToString() const;
 
     // Send-stream specific RTP settings.
     struct Rtp {
@@ skipping 75 matching lines @@
   virtual bool SendTelephoneEvent(int payload_type, int payload_frequency,
                                   int event, int duration_ms) = 0;
 
   virtual void SetMuted(bool muted) = 0;
 
   virtual Stats GetStats() const = 0;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_AUDIO_SEND_STREAM_H_