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 04c742651f789fb6f14f9adca6f88d9c9f74c46e..fcdbe26bbf9a1f86268e1309529a1a6ebf525f45 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
@@ -29,6 +29,7 @@ |
#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 { |
@@ -1298,10 +1299,16 @@ void RTPSender::SetTransportOverhead(int transport_overhead) { |
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") { |
+ rtc::CritScope lock(&send_critsect_); |
+ packet_size = packet.size() + transport_overhead_bytes_per_packet_; |
+ } |
+ |
if (transport_feedback_observer_) { |
- transport_feedback_observer_->AddPacket( |
- packet_id, packet.payload_size() + packet.padding_size(), |
- probe_cluster_id); |
+ transport_feedback_observer_->AddPacket(packet_id, packet_size, |
+ probe_cluster_id); |
} |
} |