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..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.cc.

[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,