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..ea9c66fbd7a4a6e7a96025145b68553eef2404e3 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_to_send->HasExtension<VideoTimingExtension>()) |
+ packet_to_send->set_pacer_exit_time_ms(now_ms); |
+ |
PacketOptions options; |
if (UpdateTransportSequenceNumber(packet_to_send, &options.packet_id)) { |
AddPacketToTransportFeedback(options.packet_id, *packet_to_send, |
@@ -827,6 +830,8 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet, |
if (packet->capture_time_ms() > 0) { |
packet->SetExtension<TransmissionOffset>( |
kTimestampTicksPerMs * (now_ms - packet->capture_time_ms())); |
+ if (packet->HasExtension<VideoTimingExtension>()) |
+ packet->set_pacer_exit_time_ms(now_ms); |
åsapersson
2017/06/14 08:03:50
check indentation
ilnik
2017/06/14 10:17:10
Done.
|
} |
packet->SetExtension<AbsoluteSendTime>(now_ms); |