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

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.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 275 matching lines @@
   audio_network_adaptor_ =
       audio_network_adaptor_creator_(config_string, event_log, clock);
   return audio_network_adaptor_.get() != nullptr;
 }
 
 void AudioEncoderOpus::DisableAudioNetworkAdaptor() {
   audio_network_adaptor_.reset(nullptr);
 }
 
 void AudioEncoderOpus::OnReceivedUplinkPacketLossFraction(
-    float uplink_packet_loss_fraction) {
+    const rtc::Optional<float>& uplink_packet_loss_fraction) {
   if (!audio_network_adaptor_) {
-    packet_loss_fraction_smoother_->AddSample(uplink_packet_loss_fraction);
-    float average_fraction_loss = packet_loss_fraction_smoother_->GetAverage();
-    return SetProjectedPacketLossRate(average_fraction_loss);
+    if (uplink_packet_loss_fraction) {
+      packet_loss_fraction_smoother_->AddSample(*uplink_packet_loss_fraction);
+      float average_fraction_loss =
+          packet_loss_fraction_smoother_->GetAverage();
+      SetProjectedPacketLossRate(average_fraction_loss);
+    }
+  } else {
+    audio_network_adaptor_->SetUplinkPacketLossFraction(
+        uplink_packet_loss_fraction);
+    ApplyAudioNetworkAdaptor();
   }
-  audio_network_adaptor_->SetUplinkPacketLossFraction(
-      uplink_packet_loss_fraction);
-  ApplyAudioNetworkAdaptor();
 }
 
 void AudioEncoderOpus::OnReceivedUplinkBandwidth(
     int target_audio_bitrate_bps,
     rtc::Optional<int64_t> probing_interval_ms) {
   if (audio_network_adaptor_) {
     audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps);
     // We give smoothed bitrate allocation to audio network adaptor as
     // the uplink bandwidth.
     // The probing spikes should not affect the bitrate smoother more than 25%.
@@ skipping 246 matching lines @@
             config_.uplink_bandwidth_update_interval_ms) {
       rtc::Optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
       if (smoothed_bitrate)
         audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
       bitrate_smoother_last_update_time_ = rtc::Optional<int64_t>(now_ms);
     }
   }
 }
 
 }  // namespace webrtc