| Index: webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
|
| index ebd46b02e0e04a9381693ef1a25f981bdf3d0a94..a5b42ab3e0e2c27661fe6ba16bed9ae7a7983df2 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
|
| @@ -10,6 +10,7 @@
|
|
|
| #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"
|
| @@ -316,6 +317,11 @@ 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) {
|
|
|