mirror of
https://github.com/xenia-project/FFmpeg.git
synced 2024-12-05 01:56:41 +00:00
8b1cd25ca7
This fixes a division by 0. Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
184 lines
5.4 KiB
C
184 lines
5.4 KiB
C
/*
|
|
* PMP demuxer.
|
|
* Copyright (c) 2011 Reimar Döffinger
|
|
*
|
|
* This file is part of FFmpeg.
|
|
*
|
|
* FFmpeg is free software; you can redistribute it and/or
|
|
* modify it under the terms of the GNU Lesser General Public
|
|
* License as published by the Free Software Foundation; either
|
|
* version 2.1 of the License, or (at your option) any later version.
|
|
*
|
|
* FFmpeg is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
* Lesser General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU Lesser General Public
|
|
* License along with FFmpeg; if not, write to the Free Software
|
|
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
|
*/
|
|
|
|
#include "libavutil/intreadwrite.h"
|
|
#include "avformat.h"
|
|
#include "internal.h"
|
|
|
|
typedef struct {
|
|
int cur_stream;
|
|
int num_streams;
|
|
int audio_packets;
|
|
int current_packet;
|
|
uint32_t *packet_sizes;
|
|
int packet_sizes_alloc;
|
|
} PMPContext;
|
|
|
|
static int pmp_probe(AVProbeData *p) {
|
|
if (AV_RN32(p->buf) == AV_RN32("pmpm") &&
|
|
AV_RL32(p->buf + 4) == 1)
|
|
return AVPROBE_SCORE_MAX;
|
|
return 0;
|
|
}
|
|
|
|
static int pmp_header(AVFormatContext *s)
|
|
{
|
|
PMPContext *pmp = s->priv_data;
|
|
AVIOContext *pb = s->pb;
|
|
int tb_num, tb_den;
|
|
int index_cnt;
|
|
int audio_codec_id = CODEC_ID_NONE;
|
|
int srate, channels;
|
|
int i;
|
|
uint64_t pos;
|
|
AVStream *vst = avformat_new_stream(s, NULL);
|
|
if (!vst)
|
|
return AVERROR(ENOMEM);
|
|
vst->codec->codec_type = AVMEDIA_TYPE_VIDEO;
|
|
avio_skip(pb, 8);
|
|
switch (avio_rl32(pb)) {
|
|
case 0:
|
|
vst->codec->codec_id = CODEC_ID_MPEG4;
|
|
break;
|
|
case 1:
|
|
vst->codec->codec_id = CODEC_ID_H264;
|
|
break;
|
|
default:
|
|
av_log(s, AV_LOG_ERROR, "Unsupported video format\n");
|
|
break;
|
|
}
|
|
index_cnt = avio_rl32(pb);
|
|
vst->codec->width = avio_rl32(pb);
|
|
vst->codec->height = avio_rl32(pb);
|
|
|
|
tb_num = avio_rl32(pb);
|
|
tb_den = avio_rl32(pb);
|
|
avpriv_set_pts_info(vst, 32, tb_num, tb_den);
|
|
vst->nb_frames = index_cnt;
|
|
vst->duration = index_cnt;
|
|
|
|
switch (avio_rl32(pb)) {
|
|
case 0:
|
|
audio_codec_id = CODEC_ID_MP3;
|
|
break;
|
|
case 1:
|
|
av_log(s, AV_LOG_ERROR, "AAC not yet correctly supported\n");
|
|
audio_codec_id = CODEC_ID_AAC;
|
|
break;
|
|
default:
|
|
av_log(s, AV_LOG_ERROR, "Unsupported audio format\n");
|
|
break;
|
|
}
|
|
pmp->num_streams = avio_rl16(pb) + 1;
|
|
avio_skip(pb, 10);
|
|
srate = avio_rl32(pb);
|
|
channels = avio_rl32(pb) + 1;
|
|
for (i = 1; i < pmp->num_streams; i++) {
|
|
AVStream *ast = avformat_new_stream(s, NULL);
|
|
if (!ast)
|
|
return AVERROR(ENOMEM);
|
|
ast->codec->codec_type = AVMEDIA_TYPE_AUDIO;
|
|
ast->codec->codec_id = audio_codec_id;
|
|
ast->codec->channels = channels;
|
|
ast->codec->sample_rate = srate;
|
|
avpriv_set_pts_info(ast, 32, 1, srate);
|
|
}
|
|
pos = avio_tell(pb) + 4*index_cnt;
|
|
for (i = 0; i < index_cnt; i++) {
|
|
int size = avio_rl32(pb);
|
|
int flags = size & 1 ? AVINDEX_KEYFRAME : 0;
|
|
size >>= 1;
|
|
av_add_index_entry(vst, pos, i, size, 0, flags);
|
|
pos += size;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static int pmp_packet(AVFormatContext *s, AVPacket *pkt)
|
|
{
|
|
PMPContext *pmp = s->priv_data;
|
|
AVIOContext *pb = s->pb;
|
|
int ret = 0;
|
|
int i;
|
|
|
|
if (url_feof(pb))
|
|
return AVERROR_EOF;
|
|
if (pmp->cur_stream == 0) {
|
|
int num_packets;
|
|
pmp->audio_packets = avio_r8(pb);
|
|
if (!pmp->audio_packets) {
|
|
av_log_ask_for_sample(s, "0 audio packets\n");
|
|
return AVERROR_PATCHWELCOME;
|
|
}
|
|
num_packets = (pmp->num_streams - 1) * pmp->audio_packets + 1;
|
|
avio_skip(pb, 8);
|
|
pmp->current_packet = 0;
|
|
av_fast_malloc(&pmp->packet_sizes,
|
|
&pmp->packet_sizes_alloc,
|
|
num_packets * sizeof(*pmp->packet_sizes));
|
|
if (!pmp->packet_sizes_alloc) {
|
|
av_log(s, AV_LOG_ERROR, "Cannot (re)allocate packet buffer\n");
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
for (i = 0; i < num_packets; i++)
|
|
pmp->packet_sizes[i] = avio_rl32(pb);
|
|
}
|
|
ret = av_get_packet(pb, pkt, pmp->packet_sizes[pmp->current_packet]);
|
|
if (ret >= 0) {
|
|
ret = 0;
|
|
// FIXME: this is a hack that should be removed once
|
|
// compute_pkt_fields() can handle timestamps properly
|
|
if (pmp->cur_stream == 0)
|
|
pkt->dts = s->streams[0]->cur_dts++;
|
|
pkt->stream_index = pmp->cur_stream;
|
|
}
|
|
if (pmp->current_packet % pmp->audio_packets == 0)
|
|
pmp->cur_stream = (pmp->cur_stream + 1) % pmp->num_streams;
|
|
pmp->current_packet++;
|
|
return ret;
|
|
}
|
|
|
|
static int pmp_seek(AVFormatContext *s, int stream_index, int64_t ts, int flags)
|
|
{
|
|
PMPContext *pmp = s->priv_data;
|
|
pmp->cur_stream = 0;
|
|
// fallback to default seek now
|
|
return -1;
|
|
}
|
|
|
|
static int pmp_close(AVFormatContext *s)
|
|
{
|
|
PMPContext *pmp = s->priv_data;
|
|
av_freep(&pmp->packet_sizes);
|
|
return 0;
|
|
}
|
|
|
|
AVInputFormat ff_pmp_demuxer = {
|
|
.name = "pmp",
|
|
.long_name = NULL_IF_CONFIG_SMALL("Playstation Portable PMP format"),
|
|
.priv_data_size = sizeof(PMPContext),
|
|
.read_probe = pmp_probe,
|
|
.read_header = pmp_header,
|
|
.read_packet = pmp_packet,
|
|
.read_seek = pmp_seek,
|
|
.read_close = pmp_close,
|
|
};
|