| Index: webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc
|
| diff --git a/webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc b/webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc
|
| index 034e761dcd4d30177d86982f47e5c8ce28508c68..d8f9afdd048f1bebb87bb0847927275e0e26b7e1 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc
|
| @@ -36,7 +36,11 @@ bool H264SpsParser::Parse() {
|
| // 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 &&
|
| + // Be careful about over/underflow here. byte_length_ - 3 can underflow, and
|
| + // i + 3 can overflow, but byte_length_ - i can't, because i < byte_length_
|
| + // above, and that expression will produce the number of bytes left in
|
| + // the stream including the byte at i.
|
| + if (byte_length_ - i >= 3 && sps_[i] == 0 && sps_[i + 1] == 0 &&
|
| sps_[i + 2] == 3) {
|
| // Two rbsp bytes + the emulation byte.
|
| rbsp_buffer.WriteBytes(sps_bytes + i, 2);
|
|
|