Avoid multiple calls to webrtc::field_trial::FindFullName in RTPSender

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  Copyright (c) 2012 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 105 matching lines @@
       ssrc_forced_(false),
       last_rtp_timestamp_(0),
       capture_time_ms_(0),
       last_timestamp_time_ms_(0),
       media_has_been_sent_(false),
       last_packet_marker_bit_(false),
       csrcs_(),
       rtx_(kRtxOff),
       rtp_overhead_bytes_per_packet_(0),
       retransmission_rate_limiter_(retransmission_rate_limiter),
-      overhead_observer_(overhead_observer) {
+      overhead_observer_(overhead_observer),
+      send_side_bwe_with_overhead_(
+          webrtc::field_trial::FindFullName(
+              "WebRTC-SendSideBwe-WithOverhead") == "Enabled") {
   ssrc_ = ssrc_db_->CreateSSRC();
   RTC_DCHECK(ssrc_ != 0);
   ssrc_rtx_ = ssrc_db_->CreateSSRC();
   RTC_DCHECK(ssrc_rtx_ != 0);
 
   // This random initialization is not intended to be cryptographic strong.
   timestamp_offset_ = random_.Rand<uint32_t>();
   // Random start, 16 bits. Can't be 0.
   sequence_number_rtx_ = random_.Rand(1, kMaxInitRtpSeqNumber);
   sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
@@ skipping 1112 matching lines @@
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = timestamp_offset_;
 
   return state;
 }
 
 void RTPSender::AddPacketToTransportFeedback(uint16_t packet_id,
                                              const RtpPacketToSend& packet,
                                              int probe_cluster_id) {
   size_t packet_size = packet.payload_size() + packet.padding_size();
-  if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe-WithOverhead") ==
-      "Enabled") {
+  if (send_side_bwe_with_overhead_) {
     packet_size = packet.size();
   }
 
   if (transport_feedback_observer_) {
     transport_feedback_observer_->AddPacket(packet_id, packet_size,
                                             probe_cluster_id);
   }
 }
 
 void RTPSender::UpdateRtpOverhead(const RtpPacketToSend& packet) {
   if (!overhead_observer_)
     return;
   size_t overhead_bytes_per_packet;
   {
     rtc::CritScope lock(&send_critsect_);
     if (rtp_overhead_bytes_per_packet_ == packet.headers_size()) {
       return;
     }
     rtp_overhead_bytes_per_packet_ = packet.headers_size();
     overhead_bytes_per_packet = rtp_overhead_bytes_per_packet_;
   }
   overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
 }
 
 }  // namespace webrtc