| 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);
|
| }
|
| }
|
|
|
|
|