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/api/array_view.h"
 #include "webrtc/api/optional.h"
 #include "webrtc/rtc_base/buffer.h"
 #include "webrtc/rtc_base/deprecation.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class Clock;
 class RtcEventLog;
 
+// Statistics related to Audio Network Adaptation.
+struct ANAStats {
+  ANAStats();
+  ANAStats(const ANAStats&);
+  ~ANAStats();
+  // Number of actions taken by the ANA bitrate controller since the start of
+  // the call. If this value is not set, it indicates that the bitrate
+  // controller is disabled.
+  rtc::Optional<uint32_t> bitrate_action_counter;
+  // Number of actions taken by the ANA channel controller since the start of
+  // the call. If this value is not set, it indicates that the channel
+  // controller is disabled.
+  rtc::Optional<uint32_t> channel_action_counter;
+  // Number of actions taken by the ANA DTX controller since the start of the
+  // call. If this value is not set, it indicates that the DTX controller is
+  // disabled.
+  rtc::Optional<uint32_t> dtx_action_counter;
+  // Number of actions taken by the ANA FEC controller since the start of the
+  // call. If this value is not set, it indicates that the FEC controller is
+  // disabled.
+  rtc::Optional<uint32_t> fec_action_counter;
+  // Number of actions taken by the ANA frame length controller since the start
+  // of the call. If this value is not set, it indicates that the frame length
+  // controller is disabled.
+  rtc::Optional<uint32_t> frame_length_action_counter;
+};
+
 // This is the interface class for encoders in AudioCoding module. Each codec
 // type must have an implementation of this class.
 class AudioEncoder {
  public:
   // Used for UMA logging of codec usage. The same codecs, with the
   // same values, must be listed in
   // src/tools/metrics/histograms/histograms.xml in chromium to log
   // correct values.
   enum class CodecType {
     kOther = 0,  // Codec not specified, and/or not listed in this enum
@@ skipping 156 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_