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/*
* 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;
uint32_t index_cnt;
int audio_codec_id = AV_CODEC_ID_NONE;
int srate, channels;
int i;
uint64_t pos;
int64_t fsize = avio_size(pb);
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 = AV_CODEC_ID_MPEG4;
break;
case 1:
vst->codec->codec_id = AV_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 = AV_CODEC_ID_MP3;
break;
case 1:
av_log(s, AV_LOG_ERROR, "AAC not yet correctly supported\n");
audio_codec_id = AV_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;
pos = avio_tell(pb) + 4*index_cnt;
for (i = 0; i < index_cnt; i++) {
uint32_t size = avio_rl32(pb);
int flags = size & 1 ? AVINDEX_KEYFRAME : 0;
if (url_feof(pb)) {
av_log(s, AV_LOG_FATAL, "Encountered EOF while reading index.\n");
return AVERROR_INVALIDDATA;
}
size >>= 1;
if (size < 9 + 4*pmp->num_streams) {
av_log(s, AV_LOG_ERROR, "Packet too small\n");
return AVERROR_INVALIDDATA;
}
av_add_index_entry(vst, pos, i, size, 0, flags);
pos += size;
if (fsize > 0 && i == 0 && pos > fsize) {
av_log(s, AV_LOG_ERROR, "File ends before first packet\n");
return AVERROR_INVALIDDATA;
}
}
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);
}
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"),
.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,
};
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