Chromium Code Reviews| Index: webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc |
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc |
| index 7fcc6578fba356fa35964f613548bf8a9db720bd..849ed78eadc07a94ad988683cef58c330701f89c 100644 |
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc |
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc |
| @@ -198,11 +198,13 @@ void RTPSenderVideo::SendVideoPacketWithFlexfec( |
| std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets = |
| flexfec_sender_->GetFecPackets(); |
| for (auto& fec_packet : fec_packets) { |
| + size_t packet_length = fec_packet->size(); |
| uint32_t timestamp = fec_packet->Timestamp(); |
| uint16_t seq_num = fec_packet->SequenceNumber(); |
| if (rtp_sender_->SendToNetwork(std::move(fec_packet), kDontRetransmit, |
| RtpPacketSender::kLowPriority)) { |
| - // TODO(brandtr): Wire up stats here. |
| + rtc::CritScope cs(&stats_crit_); |
| + fec_bitrate_.Update(packet_length, clock_->TimeInMilliseconds()); |
|
danilchap
2017/01/23 12:42:39
add/extend a unittest for this.
brandtr
2017/01/24 09:42:27
Done.
|
| TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), |
| "Video::PacketFlexfec", "timestamp", timestamp, |
| "seqnum", seq_num); |