Avoid overflow in checking for emulation bytes in rbsp.

webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc

 /*
  *  Copyright (c) 2015 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 18 matching lines @@
   // General note: this is based off the 02/2014 version of the H.264 standard.
   // You can find it on this page:
   // http://www.itu.int/rec/T-REC-H.264
 
   const char* sps_bytes = reinterpret_cast<const char*>(sps_);
   // First, parse out rbsp, which is basically the source buffer minus emulation
   // bytes (the last byte of a 0x00 0x00 0x03 sequence). RBSP is defined in
   // section 7.3.1 of the H.264 standard.
   rtc::ByteBuffer rbsp_buffer;
   for (size_t i = 0; i < byte_length_;) {
-    if (i + 3 < byte_length_ && sps_[i] == 0 && sps_[i + 1] == 0 &&
+    if (byte_length_ - i >= 3 && sps_[i] == 0 && sps_[i + 1] == 0 &&

[pbos-webrtc, 2015/07/09 09:20:07]

Can you put a comment here saying that we check (byte_length_ - i) since i <
byte_length_ there's no risk of overflow? Since we're doing specific things here
to avoid overflow it'd be nice to state them so that it doesn't look like it's
just written in a different style.

         sps_[i + 2] == 3) {
       // Two rbsp bytes + the emulation byte.
       rbsp_buffer.WriteBytes(sps_bytes + i, 2);
       i += 3;
     } else {
       // Single rbsp byte.
       rbsp_buffer.WriteBytes(sps_bytes + i, 1);
       i++;
     }
   }
@@ skipping 169 matching lines @@
   // Subtract the crop for each dimension.
   width -= (frame_crop_left_offset + frame_crop_right_offset);
   height -= (frame_crop_top_offset + frame_crop_bottom_offset);
 
   width_ = width;
   height_ = height;
   return true;
 }
 
 }  // namespace webrtc