Moving overhead counting to bitrate estimators.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 6fdac6f63fb08507ec68781c4bff1e59b6ca693e..1025f18fb6b821c0a42e09da8121365623ed384a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -29,7 +29,6 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "webrtc/modules/rtp_rtcp/source/time_util.h"
-#include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
@@ -122,9 +121,7 @@ RTPSender::RTPSender(
       rtx_(kRtxOff),
       rtp_overhead_bytes_per_packet_(0),
       retransmission_rate_limiter_(retransmission_rate_limiter),
-      overhead_observer_(overhead_observer),
-      send_side_bwe_with_overhead_(
-          webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")) {
+      overhead_observer_(overhead_observer) {
   // This random initialization is not intended to be cryptographic strong.
   timestamp_offset_ = random_.Rand<uint32_t>();
   // Random start, 16 bits. Can't be 0.
@@ -1237,14 +1234,10 @@ void RTPSender::AddPacketToTransportFeedback(
     uint16_t packet_id,
     const RtpPacketToSend& packet,
     const PacedPacketInfo& pacing_info) {
-  size_t packet_size = packet.payload_size() + packet.padding_size();
-  if (send_side_bwe_with_overhead_) {
-    packet_size = packet.size();
-  }
-
   if (transport_feedback_observer_) {
-    transport_feedback_observer_->AddPacket(SSRC(), packet_id, packet_size,
-                                            pacing_info);
+    transport_feedback_observer_->AddPacket(
+        SSRC(), packet_id, packet.payload_size() + packet.padding_size(),
+        packet.headers_size(), pacing_info);
   }
 }