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
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
|
/*
* DSD Stream File (DSF) demuxer
* Copyright (c) 2014 Peter Ross
*
* 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/channel_layout.h"
#include "libavutil/intreadwrite.h"
#include "avformat.h"
#include "internal.h"
#include "id3v2.h"
typedef struct {
uint64_t data_end;
uint64_t audio_size;
uint64_t data_size;
} DSFContext;
static int dsf_probe(const AVProbeData *p)
{
if (p->buf_size < 12 || memcmp(p->buf, "DSD ", 4) || AV_RL64(p->buf + 4) != 28)
return 0;
return AVPROBE_SCORE_MAX;
}
static const uint64_t dsf_channel_layout[] = {
0,
AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
AV_CH_LAYOUT_SURROUND,
AV_CH_LAYOUT_QUAD,
AV_CH_LAYOUT_4POINT0,
AV_CH_LAYOUT_5POINT0_BACK,
AV_CH_LAYOUT_5POINT1_BACK,
};
static void read_id3(AVFormatContext *s, uint64_t id3pos)
{
ID3v2ExtraMeta *id3v2_extra_meta;
if (avio_seek(s->pb, id3pos, SEEK_SET) < 0)
return;
ff_id3v2_read(s, ID3v2_DEFAULT_MAGIC, &id3v2_extra_meta, 0);
if (id3v2_extra_meta) {
ff_id3v2_parse_apic(s, id3v2_extra_meta);
ff_id3v2_parse_chapters(s, id3v2_extra_meta);
}
ff_id3v2_free_extra_meta(&id3v2_extra_meta);
}
static int dsf_read_header(AVFormatContext *s)
{
DSFContext *dsf = s->priv_data;
AVIOContext *pb = s->pb;
AVStream *st;
uint64_t id3pos;
unsigned int channel_type;
avio_skip(pb, 4);
if (avio_rl64(pb) != 28)
return AVERROR_INVALIDDATA;
/* create primary stream before any id3 coverart streams */
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
avio_skip(pb, 8);
id3pos = avio_rl64(pb);
if (pb->seekable & AVIO_SEEKABLE_NORMAL) {
read_id3(s, id3pos);
avio_seek(pb, 28, SEEK_SET);
}
/* fmt chunk */
if (avio_rl32(pb) != MKTAG('f', 'm', 't', ' ') || avio_rl64(pb) != 52)
return AVERROR_INVALIDDATA;
if (avio_rl32(pb) != 1) {
avpriv_request_sample(s, "unknown format version");
return AVERROR_INVALIDDATA;
}
if (avio_rl32(pb)) {
avpriv_request_sample(s, "unknown format id");
return AVERROR_INVALIDDATA;
}
channel_type = avio_rl32(pb);
if (channel_type < FF_ARRAY_ELEMS(dsf_channel_layout))
st->codecpar->channel_layout = dsf_channel_layout[channel_type];
if (!st->codecpar->channel_layout)
avpriv_request_sample(s, "channel type %i", channel_type);
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->channels = avio_rl32(pb);
st->codecpar->sample_rate = avio_rl32(pb) / 8;
if (st->codecpar->channels <= 0)
return AVERROR_INVALIDDATA;
switch(avio_rl32(pb)) {
case 1: st->codecpar->codec_id = AV_CODEC_ID_DSD_LSBF_PLANAR; break;
case 8: st->codecpar->codec_id = AV_CODEC_ID_DSD_MSBF_PLANAR; break;
default:
avpriv_request_sample(s, "unknown most significant bit");
return AVERROR_INVALIDDATA;
}
dsf->audio_size = avio_rl64(pb) / 8 * st->codecpar->channels;
st->codecpar->block_align = avio_rl32(pb);
if (st->codecpar->block_align > INT_MAX / st->codecpar->channels || st->codecpar->block_align <= 0) {
avpriv_request_sample(s, "block_align invalid");
return AVERROR_INVALIDDATA;
}
st->codecpar->block_align *= st->codecpar->channels;
st->codecpar->bit_rate = st->codecpar->channels * 8LL * st->codecpar->sample_rate;
avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
avio_skip(pb, 4);
/* data chunk */
dsf->data_end = avio_tell(pb);
if (avio_rl32(pb) != MKTAG('d', 'a', 't', 'a'))
return AVERROR_INVALIDDATA;
dsf->data_size = avio_rl64(pb) - 12;
dsf->data_end += dsf->data_size + 12;
return 0;
}
static int dsf_read_packet(AVFormatContext *s, AVPacket *pkt)
{
FFFormatContext *const si = ffformatcontext(s);
DSFContext *dsf = s->priv_data;
AVIOContext *pb = s->pb;
AVStream *st = s->streams[0];
int64_t pos = avio_tell(pb);
int ret;
if (pos >= dsf->data_end)
return AVERROR_EOF;
if (dsf->data_size > dsf->audio_size) {
int last_packet = pos == (dsf->data_end - st->codecpar->block_align);
if (last_packet) {
int64_t data_pos = pos - si->data_offset;
int64_t packet_size = dsf->audio_size - data_pos;
int64_t skip_size = dsf->data_size - data_pos - packet_size;
uint8_t *dst;
int ch, ret;
if (packet_size <= 0 || skip_size <= 0)
return AVERROR_INVALIDDATA;
if ((ret = av_new_packet(pkt, packet_size)) < 0)
return ret;
dst = pkt->data;
for (ch = 0; ch < st->codecpar->channels; ch++) {
ret = avio_read(pb, dst, packet_size / st->codecpar->channels);
if (ret < packet_size / st->codecpar->channels)
return AVERROR_EOF;
dst += ret;
avio_skip(pb, skip_size / st->codecpar->channels);
}
pkt->pos = pos;
pkt->stream_index = 0;
pkt->pts = (pos - si->data_offset) / st->codecpar->channels;
pkt->duration = packet_size / st->codecpar->channels;
return 0;
}
}
ret = av_get_packet(pb, pkt, FFMIN(dsf->data_end - pos, st->codecpar->block_align));
if (ret < 0)
return ret;
pkt->stream_index = 0;
pkt->pts = (pos - si->data_offset) / st->codecpar->channels;
pkt->duration = st->codecpar->block_align / st->codecpar->channels;
return 0;
}
const AVInputFormat ff_dsf_demuxer = {
.name = "dsf",
.long_name = NULL_IF_CONFIG_SMALL("DSD Stream File (DSF)"),
.priv_data_size = sizeof(DSFContext),
.read_probe = dsf_probe,
.read_header = dsf_read_header,
.read_packet = dsf_read_packet,
.flags = AVFMT_GENERIC_INDEX | AVFMT_NO_BYTE_SEEK,
};
|
