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

webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 470 matching lines @@
     // End offset is actually start offset for next unit, excluding length field
     // so remove that from this units length.
     size_t end_offset = nalu_start_offsets[i + 1] - kLengthFieldSize;
     if (end_offset - start_offset < H264::kNaluTypeSize) {
       LOG(LS_ERROR) << "STAP-A packet too short";
       return false;
     }
 
     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;
 
     switch (nalu.type) {
       case H264::NaluType::kSps: {
         // Check if VUI is present in SPS and if it needs to be modified to
         // avoid
         // excessive decoder latency.
 
@@ skipping 167 matching lines @@
   h264->packetization_type = kH264FuA;
   h264->nalu_type = original_nal_type;
   if (first_fragment) {
     h264->nalus[h264->nalus_length] = nalu;
     h264->nalus_length = 1;
   }
   return true;
 }
 
 }  // namespace webrtc