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

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 1208 matching lines @@
       const std::string& c_name,
       const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec,
       const std::vector<webrtc::RtpExtension>& extensions,
       int max_send_bitrate_bps,
       const rtc::Optional<std::string>& audio_network_adaptor_config,
       webrtc::Call* call,
       webrtc::Transport* send_transport)
       : voe_audio_transport_(voe_audio_transport),
         call_(call),
         config_(send_transport),
+        send_side_bwe_with_overhead_(webrtc::field_trial::FindFullName(
+            "WebRTC-SendSideBwe-WithOverhead") == "Enabled"),
         max_send_bitrate_bps_(max_send_bitrate_bps),
         rtp_parameters_(CreateRtpParametersWithOneEncoding()) {
     RTC_DCHECK_GE(ch, 0);
     // TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore:
     // RTC_DCHECK(voe_audio_transport);
     RTC_DCHECK(call);
     config_.rtp.ssrc = ssrc;
     config_.rtp.c_name = c_name;
     config_.voe_channel_id = ch;
     config_.rtp.extensions = extensions;
@@ skipping 212 matching lines @@
       call_->DestroyAudioSendStream(stream_);
       stream_ = nullptr;
     }
     RTC_DCHECK(!stream_);
     if (webrtc::field_trial::FindFullName("WebRTC-Audio-SendSideBwe") ==
         "Enabled") {
       config_.min_bitrate_bps = kOpusMinBitrateBps;
       config_.max_bitrate_bps = kOpusBitrateFbBps;
       // TODO(mflodman): Keep testing this and set proper values.
       // Note: This is an early experiment currently only supported by Opus.
-      if (webrtc::field_trial::FindFullName(
-              "WebRTC-SendSideBwe-WithOverhead") == "Enabled") {
+      if (send_side_bwe_with_overhead_) {
         auto packet_sizes_ms = WebRtcVoiceCodecs::GetPacketSizesMs(
             config_.send_codec_spec.codec_inst);
         if (!packet_sizes_ms.empty()) {
           int max_packet_size_ms =
               *std::max_element(packet_sizes_ms.begin(), packet_sizes_ms.end());
           int min_packet_size_ms =
               *std::min_element(packet_sizes_ms.begin(), packet_sizes_ms.end());
 
           // Audio network adaptor will just use 20ms and 60ms frame lengths.
           // The adaptor will only be active for the Opus encoder.
@@ skipping 20 matching lines @@
     stream_ = call_->CreateAudioSendStream(config_);
     RTC_CHECK(stream_);
     UpdateSendState();
   }
 
   rtc::ThreadChecker worker_thread_checker_;
   rtc::RaceChecker audio_capture_race_checker_;
   webrtc::AudioTransport* const voe_audio_transport_ = nullptr;
   webrtc::Call* call_ = nullptr;
   webrtc::AudioSendStream::Config config_;
+  const bool send_side_bwe_with_overhead_;
   // The stream is owned by WebRtcAudioSendStream and may be reallocated if
   // configuration changes.
   webrtc::AudioSendStream* stream_ = nullptr;
 
   // Raw pointer to AudioSource owned by LocalAudioTrackHandler.
   // PeerConnection will make sure invalidating the pointer before the object
   // goes away.
   AudioSource* source_ = nullptr;
   bool send_ = false;
   bool muted_ = false;
@@ skipping 1190 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE