Commit 6a2d3fc3 authored by Ronald S. Bultje's avatar Ronald S. Bultje

Merge code for packet reading in "old" (.ra, audio-only) Realmedia files and

the newer (.rm, audio/video) files. See "[PATCH] rmdec.c: merge old/new
packet reading code" thread on mailinglist.

Originally committed as revision 18005 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent 2816ce4c
......@@ -704,71 +704,48 @@ ff_rm_retrieve_cache (AVFormatContext *s, ByteIOContext *pb,
static int rm_read_packet(AVFormatContext *s, AVPacket *pkt)
{
RMDemuxContext *rm = s->priv_data;
ByteIOContext *pb = s->pb;
AVStream *st;
int i, len, res;
int i, len, res, seq = 1;
int64_t timestamp, pos;
int old_flags, flags;
for (;;) {
if (rm->audio_pkt_cnt) {
// If there are queued audio packet return them first
st = s->streams[rm->audio_stream_num];
ff_rm_retrieve_cache(s, s->pb, st, st->priv_data, pkt);
} else if (rm->old_format) {
} else {
if (rm->old_format) {
RMStream *ast;
st = s->streams[0];
ast = st->priv_data;
if (st->codec->codec_id == CODEC_ID_RA_288) {
int x, y;
for (y = 0; y < ast->sub_packet_h; y++)
for (x = 0; x < ast->sub_packet_h/2; x++)
if (get_buffer(pb, ast->pkt.data+x*2*ast->audio_framesize+y*ast->coded_framesize, ast->coded_framesize) <= 0)
return AVERROR(EIO);
rm->audio_stream_num = 0;
rm->audio_pkt_cnt = ast->sub_packet_h * ast->audio_framesize / st->codec->block_align - 1;
// Release first audio packet
av_new_packet(pkt, st->codec->block_align);
memcpy(pkt->data, ast->pkt.data, st->codec->block_align); //FIXME avoid this
pkt->flags |= PKT_FLAG_KEY; // Mark first packet as keyframe
pkt->stream_index = 0;
} else {
/* just read raw bytes */
len = RAW_PACKET_SIZE;
len= av_get_packet(pb, pkt, len);
pkt->stream_index = 0;
if (len <= 0) {
return AVERROR(EIO);
}
pkt->size = len;
}
rm_ac3_swap_bytes(st, pkt);
timestamp = AV_NOPTS_VALUE;
len = !ast->audio_framesize ? RAW_PACKET_SIZE :
ast->coded_framesize * ast->sub_packet_h / 2;
flags = (seq++ == 1) ? 2 : 0;
} else {
int seq=1;
resync:
len=sync(s, &timestamp, &flags, &i, &pos);
if(len<0)
return AVERROR(EIO);
st = s->streams[i];
}
if(len<0 || url_feof(s->pb))
return AVERROR(EIO);
old_flags = flags;
res = ff_rm_parse_packet (s, s->pb, st, st->priv_data, len, pkt,
&seq, &flags, &timestamp);
if((old_flags&2) && (seq&0x7F) == 1)
av_add_index_entry(st, pos, timestamp, 0, 0, AVINDEX_KEYFRAME);
if (res < 0)
goto resync;
if (res)
continue;
}
if( (st->discard >= AVDISCARD_NONKEY && !(flags&2))
|| st->discard >= AVDISCARD_ALL){
av_free_packet(pkt);
while (rm->audio_pkt_cnt > 0) {
ff_rm_retrieve_cache(s, s->pb, st, st->priv_data, pkt);
av_free_packet(pkt);
}
goto resync;
}
} else
break;
}
return 0;
......
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