| Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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| index 04c742651f789fb6f14f9adca6f88d9c9f74c46e..fcdbe26bbf9a1f86268e1309529a1a6ebf525f45 100644
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| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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| @@ -29,6 +29,7 @@
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|  #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
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|  #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
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|  #include "webrtc/modules/rtp_rtcp/source/time_util.h"
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| +#include "webrtc/system_wrappers/include/field_trial.h"
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|  
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|  namespace webrtc {
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|  
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| @@ -1298,10 +1299,16 @@ void RTPSender::SetTransportOverhead(int transport_overhead) {
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|  void RTPSender::AddPacketToTransportFeedback(uint16_t packet_id,
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|                                               const RtpPacketToSend& packet,
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|                                               int probe_cluster_id) {
 | 
| +  size_t packet_size = packet.payload_size() + packet.padding_size();
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| +  if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe-WithOverhead") ==
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| +      "Enabled") {
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| +    rtc::CritScope lock(&send_critsect_);
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| +    packet_size = packet.size() + transport_overhead_bytes_per_packet_;
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| +  }
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| +
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|    if (transport_feedback_observer_) {
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| -    transport_feedback_observer_->AddPacket(
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| -        packet_id, packet.payload_size() + packet.padding_size(),
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| -        probe_cluster_id);
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| +    transport_feedback_observer_->AddPacket(packet_id, packet_size,
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| +                                            probe_cluster_id);
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|    }
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|  }
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|  
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| 
 |