Move more calls to webrtc::field_trial::FindFullName into ctor

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 156 matching lines @@
   }
   return bitrate_bps <= complexity_threshold_bps
              ? rtc::Optional<int>(low_rate_complexity)
              : rtc::Optional<int>(complexity);
 }
 
 AudioEncoderOpus::AudioEncoderOpus(
     const Config& config,
     AudioNetworkAdaptorCreator&& audio_network_adaptor_creator,
     std::unique_ptr<SmoothingFilter> bitrate_smoother)
-    : packet_loss_rate_(0.0),
+    : send_side_bwe_with_overhead_(webrtc::field_trial::FindFullName(
+          "WebRTC-SendSideBwe-WithOverhead") == "Enabled"),
+      packet_loss_rate_(0.0),
       inst_(nullptr),
       packet_loss_fraction_smoother_(new PacketLossFractionSmoother(
           config.clock)),
       audio_network_adaptor_creator_(
           audio_network_adaptor_creator
               ? std::move(audio_network_adaptor_creator)
               : [this](const std::string& config_string,
                        RtcEventLog* event_log,
                        const Clock* clock) {
                   return DefaultAudioNetworkAdaptorCreator(config_string,
@@ skipping 119 matching lines @@
     // In order to limit the affect of a BWE spike within 25% of its value
     // before
     // the next probing, we would choose a time constant that fulfills
     // 1 - e^(-probing_interval_ms / time_constant) < 0.25
     // Then 4 * probing_interval_ms is a good choice.
     if (probing_interval_ms)
       bitrate_smoother_->SetTimeConstantMs(*probing_interval_ms * 4);
     bitrate_smoother_->AddSample(target_audio_bitrate_bps);
 
     ApplyAudioNetworkAdaptor();
-  } else if (webrtc::field_trial::FindFullName(
-                 "WebRTC-SendSideBwe-WithOverhead") == "Enabled") {
+  } else if (send_side_bwe_with_overhead_) {
     if (!overhead_bytes_per_packet_) {
       LOG(LS_INFO)
           << "AudioEncoderOpus: Overhead unknown, target audio bitrate "
           << target_audio_bitrate_bps << " bps is ignored.";
       return;
     }
     const int overhead_bps = static_cast<int>(
         *overhead_bytes_per_packet_ * 8 * 100 / Num10MsFramesInNextPacket());
     SetTargetBitrate(std::min(
         kMaxBitrateBps,
@@ skipping 221 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