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

webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h

 /*
  *  Copyright (c) 2016 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_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
 #define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
 
 #include "webrtc/base/optional.h"
 
 namespace webrtc {
 
 // An AudioNetworkAdaptor optimizes the audio experience by suggesting a
 // suitable runtime configuration (bit rate, frame length, FEC, etc.) to the
 // encoder based on network metrics.
 class AudioNetworkAdaptor {
  public:
   struct EncoderRuntimeConfig {
     EncoderRuntimeConfig();
     EncoderRuntimeConfig(const EncoderRuntimeConfig& other);
     ~EncoderRuntimeConfig();
     rtc::Optional<int> bitrate_bps;
     rtc::Optional<int> frame_length_ms;
+    // Note: This is what we tell the encoder. It doesn't have to reflect
+    // the actual NetworkMetrics; it's subject to our decision.
     rtc::Optional<float> uplink_packet_loss_fraction;
     rtc::Optional<bool> enable_fec;
     rtc::Optional<bool> enable_dtx;
 
     // Some encoders can encode fewer channels than the actual input to make
     // better use of the bandwidth. |num_channels| sets the number of channels
     // to encode.
     rtc::Optional<size_t> num_channels;
   };
 
   virtual ~AudioNetworkAdaptor() = default;
 
   virtual void SetUplinkBandwidth(int uplink_bandwidth_bps) = 0;
 
   virtual void SetUplinkPacketLossFraction(
       float uplink_packet_loss_fraction) = 0;
 
+  virtual void SetUplinkRecoverablePacketLossFraction(
+      float uplink_recoverable_packet_loss_fraction) = 0;
+
   virtual void SetRtt(int rtt_ms) = 0;
 
   virtual void SetTargetAudioBitrate(int target_audio_bitrate_bps) = 0;
 
   virtual void SetOverhead(size_t overhead_bytes_per_packet) = 0;
 
   virtual EncoderRuntimeConfig GetEncoderRuntimeConfig() = 0;
 
   virtual void StartDebugDump(FILE* file_handle) = 0;
 
   virtual void StopDebugDump() = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_