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

webrtc/modules/video_coding/codecs/h264/include/h264_globals.h

 /*
  *  Copyright (c) 2012 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 39 matching lines @@
       stream << "SingleNalUnit";
       break;
   }
   return stream;
 }
 
 struct NaluInfo {
   uint8_t type;
   int sps_id;
   int pps_id;
-
-  // Offset and size are only valid for non-FuA packets.
-  size_t offset;
-  size_t size;
 };
 
 const size_t kMaxNalusPerPacket = 10;
 
 struct RTPVideoHeaderH264 {
   // The NAL unit type. If this is a header for a
   // fragmented packet, it's the NAL unit type of
   // the original data. If this is the header for an
   // aggregated packet, it's the NAL unit type of
   // the first NAL unit in the packet.
   uint8_t nalu_type;
   // The packetization type of this buffer - single, aggregated or fragmented.
   H264PacketizationTypes packetization_type;
   NaluInfo nalus[kMaxNalusPerPacket];
   size_t nalus_length;
   // The packetization mode of this transport. Packetization mode
   // determines which packetization types are allowed when packetizing.
   H264PacketizationMode packetization_mode;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_