Prevent size_t underflow in H264 SPS parsing.

webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc

 /*
  *  Copyright (c) 2014 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 545 matching lines @@
   RtpDepacketizer::ParsedPayload payload;
   EXPECT_FALSE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
 }
 
 TEST_F(RtpDepacketizerH264Test, TestTruncationJustAfterSingleStapANalu) {
   const uint8_t kPayload[] = {0x38, 0x27, 0x27};
   RtpDepacketizer::ParsedPayload payload;
   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