Add and modify a few ANA stats.

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.
  */
 
@@ skipping 30 matching lines @@
   // 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;
+  // Number of times the ANA frame length controller decided to increase the
+  // frame length 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_increase_counter;
+  // Number of times the ANA frame length controller decided to decrease the
+  // frame length 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_decrease_counter;
+  // The uplink packet loss fractions as set by the ANA FEC controller. If this
+  // value is not set, it indicates that the ANA FEC controller is not active.
+  rtc::Optional<float> uplink_packet_loss_fraction;
 };
 
 // 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.
@@ skipping 170 matching lines @@
 
  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_