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| author | Kostya Shishkov <kostya.shishkov@gmail.com> | 2009-12-13 09:08:58 +0000 |
|---|---|---|
| committer | Kostya Shishkov <kostya.shishkov@gmail.com> | 2009-12-13 09:08:58 +0000 |
| commit | 119c61a30faf564216524f953e4f1582fa800409 (patch) | |
| tree | 75c0c0e459c9f59ea544369b95fa5605ce268dec | |
| parent | 461ef741163c6aba4d8e738073b9a54b085fd2a4 (diff) | |
| download | ffmpeg-119c61a30faf564216524f953e4f1582fa800409.tar.gz | |
RM audio stream version should be 16-bit followed by header size or reserved
word, so treat it this way instead of extracting different parts from 32-bit
value.
Originally committed as revision 20820 to svn://svn.ffmpeg.org/ffmpeg/trunk
| -rw-r--r-- | libavformat/rmdec.c | 20 |
1 files changed, 11 insertions, 9 deletions
diff --git a/libavformat/rmdec.c b/libavformat/rmdec.c index 026d132e6e..d53918fc70 100644 --- a/libavformat/rmdec.c +++ b/libavformat/rmdec.c @@ -130,19 +130,20 @@ static int rm_read_audio_stream_info(AVFormatContext *s, ByteIOContext *pb, uint32_t version; /* ra type header */ - version = get_be32(pb); /* version */ - if (((version >> 16) & 0xff) == 3) { + version = get_be16(pb); /* version */ + if (version == 3) { + int header_size = get_be16(pb); int64_t startpos = url_ftell(pb); url_fskip(pb, 14); rm_read_metadata(s, 0); - if ((startpos + (version & 0xffff)) >= url_ftell(pb) + 2) { + if ((startpos + header_size) >= url_ftell(pb) + 2) { // fourcc (should always be "lpcJ") get_byte(pb); get_str8(pb, buf, sizeof(buf)); } // Skip extra header crap (this should never happen) - if ((startpos + (version & 0xffff)) > url_ftell(pb)) - url_fskip(pb, (version & 0xffff) + startpos - url_ftell(pb)); + if ((startpos + header_size) > url_ftell(pb)) + url_fskip(pb, header_size + startpos - url_ftell(pb)); st->codec->sample_rate = 8000; st->codec->channels = 1; st->codec->codec_type = CODEC_TYPE_AUDIO; @@ -151,6 +152,7 @@ static int rm_read_audio_stream_info(AVFormatContext *s, ByteIOContext *pb, int flavor, sub_packet_h, coded_framesize, sub_packet_size; int codecdata_length; /* old version (4) */ + url_fskip(pb, 2); /* unused */ get_be32(pb); /* .ra4 */ get_be32(pb); /* data size */ get_be16(pb); /* version2 */ @@ -164,13 +166,13 @@ static int rm_read_audio_stream_info(AVFormatContext *s, ByteIOContext *pb, st->codec->block_align= get_be16(pb); /* frame size */ ast->sub_packet_size = sub_packet_size = get_be16(pb); /* sub packet size */ get_be16(pb); /* ??? */ - if (((version >> 16) & 0xff) == 5) { + if (version == 5) { get_be16(pb); get_be16(pb); get_be16(pb); } st->codec->sample_rate = get_be16(pb); get_be32(pb); st->codec->channels = get_be16(pb); - if (((version >> 16) & 0xff) == 5) { + if (version == 5) { get_be32(pb); get_buffer(pb, buf, 4); buf[4] = 0; @@ -201,7 +203,7 @@ static int rm_read_audio_stream_info(AVFormatContext *s, ByteIOContext *pb, case CODEC_ID_ATRAC3: case CODEC_ID_SIPR: get_be16(pb); get_byte(pb); - if (((version >> 16) & 0xff) == 5) + if (version == 5) get_byte(pb); codecdata_length = get_be32(pb); if(codecdata_length + FF_INPUT_BUFFER_PADDING_SIZE <= (unsigned)codecdata_length){ @@ -241,7 +243,7 @@ static int rm_read_audio_stream_info(AVFormatContext *s, ByteIOContext *pb, break; case CODEC_ID_AAC: get_be16(pb); get_byte(pb); - if (((version >> 16) & 0xff) == 5) + if (version == 5) get_byte(pb); st->codec->codec_id = CODEC_ID_AAC; codecdata_length = get_be32(pb); |