Don't update the jitter estimate with frames containing retransmitted packets.

webrtc/modules/video_coding/frame_object.cc

Index: webrtc/modules/video_coding/frame_object.cc
diff --git a/webrtc/modules/video_coding/frame_object.cc b/webrtc/modules/video_coding/frame_object.cc
index d593d4bef4ca4956f65646c0e694fbc26ebe95ef..70b0a02868ab0511b23675a7538dde2f85ab2d00 100644
--- a/webrtc/modules/video_coding/frame_object.cc
+++ b/webrtc/modules/video_coding/frame_object.cc
@@ -121,6 +121,10 @@ int64_t RtpFrameObject::RenderTime() const {
   return _renderTimeMs;
 }
 
+bool RtpFrameObject::delayed_by_retransmission() const {
+  return times_nacked() > 0;
+}
+
 rtc::Optional<RTPVideoTypeHeader> RtpFrameObject::GetCodecHeader() const {
   rtc::CritScope lock(&packet_buffer_->crit_);
   VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);