1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
|
/*
* 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 = AV_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 = 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;
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"),
.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,
};
|
