Only append SPS/PPS to bitstream if supplied out of bound.

webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
index ce3f340084569c523f1d1084c2ef414be2b12c82..68e54f14328e486a530fccc88d1313bbdbc4f876 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -488,8 +488,6 @@ bool RtpDepacketizerH264::ProcessStapAOrSingleNalu(
 
     NaluInfo nalu;
     nalu.type = payload_data[start_offset] & kTypeMask;
-    nalu.offset = start_offset;
-    nalu.size = end_offset - start_offset;
     nalu.sps_id = -1;
     nalu.pps_id = -1;
     start_offset += H264::kNaluTypeSize;