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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Issue 2911193002: Implement timing frames. (Closed)
Patch Set: Fix capture timestamp issues which cause capture time from the future Created 3 years, 7 months ago
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« webrtc/api/video/video_timing.h ('K') | « webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc » ('j') | webrtc/test/frame_generator_capturer.cc » ('J')
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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>())
+ packet->set_pacer_exit_time_ms(clock_->TimeInMilliseconds());
+
PacketOptions options;
if (UpdateTransportSequenceNumber(packet_to_send, &options.packet_id)) {
AddPacketToTransportFeedback(options.packet_id, *packet_to_send,
« webrtc/api/video/video_timing.h ('K') | « webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc » ('j') | webrtc/test/frame_generator_capturer.cc » ('J')

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