Implement timing frames.

webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
index 3f4c401a19d5c43ff9f64c6a9e3bd14d60ff9fb6..41af62b9faa3c6622d7d00086371b9a8c4774e33 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -304,6 +304,7 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
   auto last_packet = rtc::MakeUnique<RtpPacketToSend>(*rtp_header);
 
   size_t fec_packet_overhead;
+  bool is_timing_frame = false;
   bool red_enabled;
   int32_t retransmission_settings;
   {
@@ -332,6 +333,11 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
         last_packet->SetExtension<VideoContentTypeExtension>(
             video_header->content_type);
       }
+      if (video_header->video_timing.is_timing_frame) {
+        last_packet->SetExtension<VideoTimingExtension>(
+            video_header->video_timing);
+        is_timing_frame = true;
+      }
     }
 
     // FEC settings.
@@ -388,6 +394,11 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
     if (!rtp_sender_->AssignSequenceNumber(packet.get()))
       return false;
 
+    // Put packetization finish timestamp into extension.
+    if (last && is_timing_frame) {
+      packet->set_packetization_finish_time_ms(clock_->TimeInMilliseconds());
+    }
+
     const bool protect_packet =
         (packetizer->GetProtectionType() == kProtectedPacket);
     if (flexfec_enabled()) {