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
|
/*
* Copyright (c) 2012 Laurent Aimar
*
* 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 "avcodec.h"
#include "internal.h"
#include "dvaudio.h"
typedef struct DVAudioContext {
int block_size;
int is_12bit;
int is_pal;
int16_t shuffle[2000];
} DVAudioContext;
static av_cold int decode_init(AVCodecContext *avctx)
{
DVAudioContext *s = avctx->priv_data;
int i;
if (avctx->channels != 2) {
av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
return AVERROR(EINVAL);
}
if (avctx->codec_tag == 0x0215) {
s->block_size = 7200;
} else if (avctx->codec_tag == 0x0216) {
s->block_size = 8640;
} else if (avctx->block_align == 7200 ||
avctx->block_align == 8640) {
s->block_size = avctx->block_align;
} else {
return AVERROR(EINVAL);
}
s->is_pal = s->block_size == 8640;
s->is_12bit = avctx->bits_per_coded_sample == 12;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->channel_layout = AV_CH_LAYOUT_STEREO;
for (i = 0; i < FF_ARRAY_ELEMS(s->shuffle); i++) {
const unsigned a = s->is_pal ? 18 : 15;
const unsigned b = 3 * a;
s->shuffle[i] = 80 * ((21 * (i % 3) + 9 * (i / 3) + ((i / a) % 3)) % b) +
(2 + s->is_12bit) * (i / b) + 8;
}
return 0;
}
static inline uint16_t dv_audio_12to16(uint16_t sample)
{
uint16_t shift, result;
sample = (sample < 0x800) ? sample : sample | 0xf000;
shift = (sample & 0xf00) >> 8;
if (shift < 0x2 || shift > 0xd) {
result = sample;
} else if (shift < 0x8) {
shift--;
result = (sample - (256 * shift)) << shift;
} else {
shift = 0xe - shift;
result = ((sample + ((256 * shift) + 1)) << shift) - 1;
}
return result;
}
static int decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *pkt)
{
DVAudioContext *s = avctx->priv_data;
AVFrame *frame = data;
const uint8_t *src = pkt->data;
int16_t *dst;
int ret, i;
if (pkt->size < s->block_size)
return AVERROR_INVALIDDATA;
frame->nb_samples = dv_get_audio_sample_count(pkt->data + 244, s->is_pal);
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
dst = (int16_t *)frame->data[0];
for (i = 0; i < frame->nb_samples; i++) {
const uint8_t *v = &src[s->shuffle[i]];
if (s->is_12bit) {
*dst++ = dv_audio_12to16((v[0] << 4) | ((v[2] >> 4) & 0x0f));
*dst++ = dv_audio_12to16((v[1] << 4) | ((v[2] >> 0) & 0x0f));
} else {
*dst++ = AV_RB16(&v[0]);
*dst++ = AV_RB16(&v[s->is_pal ? 4320 : 3600]);
}
}
*got_frame_ptr = 1;
return s->block_size;
}
const AVCodec ff_dvaudio_decoder = {
.name = "dvaudio",
.long_name = NULL_IF_CONFIG_SMALL("Ulead DV Audio"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_DVAUDIO,
.init = decode_init,
.decode = decode_frame,
.capabilities = AV_CODEC_CAP_DR1,
.priv_data_size = sizeof(DVAudioContext),
};
|
