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); |
} |
} |