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 3cac4195b46c132d58e1c968c22283fe0ae826b5..54994bce29c3b0eb4c0431461cff0c07179c1a01 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
@@ -740,6 +740,9 @@ bool RTPSender::PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet, |
diff_ms); |
packet_to_send->SetExtension<AbsoluteSendTime>(now_ms); |
+ if (packet->HasExtension<VideoTimingExtension>()) |
åsapersson
2017/06/12 14:33:42
packet_to_send?
ilnik
2017/06/13 08:43:13
Done. Thank's for catching that.
|
+ packet->set_pacer_exit_time_ms(clock_->TimeInMilliseconds()); |
åsapersson
2017/06/12 14:33:42
maybe use now_ms
ilnik
2017/06/13 08:43:13
Done.
|
+ |
åsapersson
2017/06/12 14:33:42
Maybe add a unit test, e.g. in rtp_sender_unittest
ilnik
2017/06/13 08:43:13
Done. Please check rtp_sender_unittest.cc
|
PacketOptions options; |
if (UpdateTransportSequenceNumber(packet_to_send, &options.packet_id)) { |
AddPacketToTransportFeedback(options.packet_id, *packet_to_send, |