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mp3demux: Rewrite xing TOC based seeking 

The libav code depends on mp3 startcodes only occuring at the start of
frames. But there is nothing in mp3 that prevents them occuring elsewhere
by chance. Thus the code would fail randomly, the new code searches for 3
consecutive valid frames. If this turns out to be still insufficient the
number can be raised further, or additional checks added.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2012-10-21 02:42:16 +02:00
parent 7696a392e2
commit b6267901c4
1 changed files with 34 additions and 10 deletions
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44
libavformat/mp3dec.c
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@ -253,6 +253,21 @@ static int mp3_read_packet(AVFormatContext *s, AVPacket *pkt)
return ret;
}
static int check(AVFormatContext *s, int64_t pos)
{
int64_t ret = avio_seek(s->pb, pos, SEEK_SET);
unsigned header;
MPADecodeHeader sd;
if (ret < 0)
return ret;
header = avio_rb32(s->pb);
if (ff_mpa_check_header(header) < 0)
return -1;
if (avpriv_mpegaudio_decode_header(&sd, header) == 1)
return -1;
return sd.frame_size;
}
static int mp3_seek(AVFormatContext *s, int stream_index, int64_t timestamp,
int flags)
{
@ -261,6 +276,7 @@ static int mp3_seek(AVFormatContext *s, int stream_index, int64_t timestamp,
AVStream *st = s->streams[0];
int64_t ret = av_index_search_timestamp(st, timestamp, flags);
uint32_t header = 0;
int i, j;
if (!mp3->xing_toc) {
st->skip_samples = timestamp <= 0 ? mp3->start_pad + 528 + 1 : 0;
@ -276,19 +292,27 @@ static int mp3_seek(AVFormatContext *s, int stream_index, int64_t timestamp,
if (ret < 0)
return ret;
while (!s->pb->eof_reached) {
header = (header << 8) + avio_r8(s->pb);
if (ff_mpa_check_header(header) >= 0) {
ff_update_cur_dts(s, st, ie->timestamp);
ret = avio_seek(s->pb, -4, SEEK_CUR);
st->skip_samples = ie->timestamp <= 0 ? mp3->start_pad + 528 + 1 : 0;
return (ret >= 0) ? 0 : ret;
#define MIN_VALID 3
for(i=0; i<4096; i++) {
int64_t pos = ie->pos + i;
for(j=0; j<MIN_VALID; j++) {
ret = check(s, pos);
if(ret < 0)
break;
pos += ret;
}
if(j==MIN_VALID)
break;
}
if(j!=MIN_VALID)
i=0;
return AVERROR_EOF;
ret = avio_seek(s->pb, ie->pos + i, SEEK_SET);
if (ret < 0)
return ret;
ff_update_cur_dts(s, st, ie->timestamp);
st->skip_samples = ie->timestamp <= 0 ? mp3->start_pad + 528 + 1 : 0;
return 0;
}
AVInputFormat ff_mp3_demuxer = {