OnReceivedUplinkPacketLossFraction() receives [const rtc::Optional<float>&]

webrtc/modules/audio_coding/codecs/audio_encoder.cc

 /*
  *  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 56 matching lines @@
 
 bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string,
                                              RtcEventLog* event_log,
                                              const Clock* clock) {
   return false;
 }
 
 void AudioEncoder::DisableAudioNetworkAdaptor() {}
 
 void AudioEncoder::OnReceivedUplinkPacketLossFraction(
-    float uplink_packet_loss_fraction) {}
+    const rtc::Optional<float>& uplink_packet_loss_fraction) {}
 
 void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
   OnReceivedUplinkBandwidth(target_audio_bitrate_bps, rtc::Optional<int64_t>());
 }
 
 void AudioEncoder::OnReceivedUplinkBandwidth(
     int target_audio_bitrate_bps,
     rtc::Optional<int64_t> probing_interval_ms) {}
 
 void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
 
 void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {}
 
 void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms,
                                                int max_frame_length_ms) {}
 
 }  // namespace webrtc