| 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();
|
|
|