Implement timing frames.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

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);