Chromium Code Reviews| 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 5a0dc13a6d7025e95723482d566e5477b35b6c83..4341b7b0a750d1264d5c9ef2360d98b35048dda5 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" |
| @@ -904,6 +905,15 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet, |
| } |
| packet->SetExtension<AbsoluteSendTime>(now_ms); |
| + BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "SentBitrate[Kbps]", now_ms, \ |
| + ActualSendBitrateKbit(), packet->Ssrc()); |
| + BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "FecBitrate[Kbps]", now_ms, \ |
| + FecOverheadRate()/1000, packet->Ssrc()); |
| + BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "NackBitrate[Kbps]", now_ms, \ |
| + NackOverheadRate()/1000, packet->Ssrc()); |
| + BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoBitrate[bps]", now_ms, \ |
| + VideoBitrateSent()/1000, packet->Ssrc()); |
|
stefan-webrtc
2016/09/02 11:13:04
Same here, remove '\' and add spaces around '/'.
|
| + |
| if (paced_sender_) { |
| uint16_t seq_no = packet->SequenceNumber(); |
| uint32_t ssrc = packet->Ssrc(); |