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Prevent size_t underflow in H264 SPS parsing.

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BUG=webrtc:4771
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1219493004

Cr-Commit-Position: refs/heads/master@{#9523}
This commit is contained in:
pbos 2015-07-01 03:00:13 -07:00 committed by Commit bot
parent 2f1509395b
commit 4daa90eed7
2 changed files with 8 additions and 2 deletions
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4
webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc
View File

@ -36,8 +36,8 @@ 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 < byte_length_ - 3 &&
sps_[i] == 0 && sps_[i + 1] == 0 && sps_[i + 2] == 3) {
if (i + 3 < byte_length_ && sps_[i] == 0 && sps_[i + 1] == 0 &&
sps_[i + 2] == 3) {
// Two rbsp bytes + the emulation byte.
rbsp_buffer.WriteBytes(sps_bytes + i, 2);
i += 3;

6
webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
View File

@ -563,4 +563,10 @@ TEST_F(RtpDepacketizerH264Test, TestTruncationJustAfterSingleStapANalu) {
EXPECT_FALSE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
}
TEST_F(RtpDepacketizerH264Test, TestShortSpsPacket) {
const uint8_t kPayload[] = {0x27, 0x80, 0x00};
RtpDepacketizer::ParsedPayload payload;
EXPECT_TRUE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
}
} // namespace webrtc