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 b77155123384229d12ecf544947d51e6ec486099..84f1fae216accc60a8f18b78bbe5e6f1ebc387ce 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
@@ -20,6 +20,7 @@ |
#include "webrtc/base/timeutils.h" |
#include "webrtc/call.h" |
#include "webrtc/call/rtc_event_log.h" |
+#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h" |
#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" |
#include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
#include "webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h" |
@@ -905,6 +906,20 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet, |
} |
packet->SetExtension<AbsoluteSendTime>(now_ms); |
+ if (video_) { |
+ BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate[kbps]", now_ms, |
+ ActualSendBitrateKbit(), packet->Ssrc()); |
+ BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoFecBitrate[Kbps]", now_ms, |
+ FecOverheadRate() / 1000, packet->Ssrc()); |
+ BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoNackBitrate[Kbps]", now_ms, |
+ NackOverheadRate() / 1000, packet->Ssrc()); |
+ } else { |
+ BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate[kbps]", now_ms, |
+ ActualSendBitrateKbit(), packet->Ssrc()); |
+ BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioNackBitrate[Kbps]", now_ms, |
+ NackOverheadRate() / 1000, packet->Ssrc()); |
+ } |
+ |
if (paced_sender_) { |
uint16_t seq_no = packet->SequenceNumber(); |
uint32_t ssrc = packet->Ssrc(); |