Allow ANA to receive RPLR (recoverable packet loss rate) indications

webrtc/modules/audio_coding/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 157 matching lines @@
                                          const Clock* clock);
 
   // Disables audio network adaptor.
   virtual void DisableAudioNetworkAdaptor();
 
   // Provides uplink packet loss fraction to this encoder to allow it to adapt.
   // |uplink_packet_loss_fraction| is in the range [0.0, 1.0].
   virtual void OnReceivedUplinkPacketLossFraction(
       float uplink_packet_loss_fraction);
 
+  // Provides 1st-order-FEC-recoverable uplink packet loss rate to this encoder
+  // to allow it to adapt.
+  // |uplink_recoverable_packet_loss_fraction| is in the range [0.0, 1.0].
+  virtual void OnReceivedUplinkRecoverablePacketLossFraction(
+      float uplink_recoverable_packet_loss_fraction);
+
   // Provides target audio bitrate to this encoder to allow it to adapt.
   virtual void OnReceivedTargetAudioBitrate(int target_bps);
 
   // Provides target audio bitrate and corresponding probing interval of
   // the bandwidth estimator to this encoder to allow it to adapt.
   virtual void OnReceivedUplinkBandwidth(
       int target_audio_bitrate_bps,
       rtc::Optional<int64_t> probing_interval_ms);
 
   // Provides RTT to this encoder to allow it to adapt.
@@ skipping 10 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_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_