Prevent size_t underflow in H264 SPS parsing.

webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
index 7bf1cf1b6f8e783d7f0c2f90aa95c460b276b711..3ad5686fe9ee4223e965dd16df9f2cbadef4605a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
@@ -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