Prevent OOB reads for zero-length H264 payloads.

webrtc/modules/rtp_rtcp/source/rtp_format_h264.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.
  */
 
 #include <string.h>
 
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/h264_sps_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
 
 namespace webrtc {
 namespace {
 
 enum Nalu {
   kSlice = 1,
@@ skipping 286 matching lines @@
 }
 
 std::string RtpPacketizerH264::ToString() {
   return "RtpPacketizerH264";
 }
 
 bool RtpDepacketizerH264::Parse(ParsedPayload* parsed_payload,
                                 const uint8_t* payload_data,
                                 size_t payload_data_length) {
   assert(parsed_payload != NULL);
+  if (payload_data_length == 0) {
+    LOG(LS_ERROR) << "Empty payload.";
+    return false;
+  }
+
   uint8_t nal_type = payload_data[0] & kTypeMask;
   size_t offset = 0;
   if (nal_type == kFuA) {
     // Fragmented NAL units (FU-A).
     ParseFuaNalu(parsed_payload, payload_data, payload_data_length, &offset);
   } else {
     // We handle STAP-A and single NALU's the same way here. The jitter buffer
     // will depacketize the STAP-A into NAL units later.
     ParseSingleNalu(parsed_payload, payload_data, payload_data_length);
   }
 
   parsed_payload->payload = payload_data + offset;
   parsed_payload->payload_length = payload_data_length - offset;
   return true;
 }
 }  // namespace webrtc