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
|
/*
* ADX ADPCM codecs
* Copyright (c) 2001,2003 BERO
*
* 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 "avcodec.h"
#include "adx.h"
#include "bytestream.h"
#include "codec_internal.h"
#include "encode.h"
#include "put_bits.h"
/**
* @file
* SEGA CRI adx codecs.
*
* Reference documents:
* http://ku-www.ss.titech.ac.jp/~yatsushi/adx.html
* adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/
*/
static void adx_encode(ADXContext *c, uint8_t *adx, const int16_t *wav,
ADXChannelState *prev, int channels)
{
PutBitContext pb;
int scale;
int i, j;
int s0, s1, s2, d;
int max = 0;
int min = 0;
s1 = prev->s1;
s2 = prev->s2;
for (i = 0, j = 0; j < 32; i += channels, j++) {
s0 = wav[i];
d = s0 + ((-c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS);
if (max < d)
max = d;
if (min > d)
min = d;
s2 = s1;
s1 = s0;
}
if (max == 0 && min == 0) {
prev->s1 = s1;
prev->s2 = s2;
memset(adx, 0, BLOCK_SIZE);
return;
}
if (max / 7 > -min / 8)
scale = max / 7;
else
scale = -min / 8;
if (scale == 0)
scale = 1;
AV_WB16(adx, scale);
init_put_bits(&pb, adx + 2, 16);
s1 = prev->s1;
s2 = prev->s2;
for (i = 0, j = 0; j < 32; i += channels, j++) {
d = wav[i] + ((-c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS);
d = av_clip_intp2(ROUNDED_DIV(d, scale), 3);
put_sbits(&pb, 4, d);
s0 = d * scale + ((c->coeff[0] * s1 + c->coeff[1] * s2) >> COEFF_BITS);
s2 = s1;
s1 = s0;
}
prev->s1 = s1;
prev->s2 = s2;
flush_put_bits(&pb);
}
#define HEADER_SIZE 36
static int adx_encode_header(AVCodecContext *avctx, uint8_t *buf, int bufsize)
{
ADXContext *c = avctx->priv_data;
bytestream_put_be16(&buf, 0x8000); /* header signature */
bytestream_put_be16(&buf, HEADER_SIZE - 4); /* copyright offset */
bytestream_put_byte(&buf, 3); /* encoding */
bytestream_put_byte(&buf, BLOCK_SIZE); /* block size */
bytestream_put_byte(&buf, 4); /* sample size */
bytestream_put_byte(&buf, avctx->ch_layout.nb_channels); /* channels */
bytestream_put_be32(&buf, avctx->sample_rate); /* sample rate */
bytestream_put_be32(&buf, 0); /* total sample count */
bytestream_put_be16(&buf, c->cutoff); /* cutoff frequency */
bytestream_put_byte(&buf, 3); /* version */
bytestream_put_byte(&buf, 0); /* flags */
bytestream_put_be32(&buf, 0); /* unknown */
bytestream_put_be32(&buf, 0); /* loop enabled */
bytestream_put_be16(&buf, 0); /* padding */
bytestream_put_buffer(&buf, "(c)CRI", 6); /* copyright signature */
return HEADER_SIZE;
}
static av_cold int adx_encode_init(AVCodecContext *avctx)
{
ADXContext *c = avctx->priv_data;
if (avctx->ch_layout.nb_channels > 2) {
av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
return AVERROR(EINVAL);
}
avctx->frame_size = BLOCK_SAMPLES;
/* the cutoff can be adjusted, but this seems to work pretty well */
c->cutoff = 500;
ff_adx_calculate_coeffs(c->cutoff, avctx->sample_rate, COEFF_BITS, c->coeff);
return 0;
}
static int adx_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
ADXContext *c = avctx->priv_data;
const int16_t *samples = frame ? (const int16_t *)frame->data[0] : NULL;
uint8_t *dst;
int channels = avctx->ch_layout.nb_channels;
int ch, out_size, ret;
if (!samples) {
if (c->eof)
return 0;
if ((ret = ff_get_encode_buffer(avctx, avpkt, 18, 0)) < 0)
return ret;
c->eof = 1;
dst = avpkt->data;
bytestream_put_be16(&dst, 0x8001);
bytestream_put_be16(&dst, 0x000E);
bytestream_put_be64(&dst, 0x0);
bytestream_put_be32(&dst, 0x0);
bytestream_put_be16(&dst, 0x0);
*got_packet_ptr = 1;
return 0;
}
out_size = BLOCK_SIZE * channels + !c->header_parsed * HEADER_SIZE;
if ((ret = ff_get_encode_buffer(avctx, avpkt, out_size, 0)) < 0)
return ret;
dst = avpkt->data;
if (!c->header_parsed) {
int hdrsize;
if ((hdrsize = adx_encode_header(avctx, dst, avpkt->size)) < 0) {
av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
}
dst += hdrsize;
c->header_parsed = 1;
}
for (ch = 0; ch < channels; ch++) {
adx_encode(c, dst, samples + ch, &c->prev[ch], channels);
dst += BLOCK_SIZE;
}
avpkt->pts = frame->pts;
avpkt->duration = frame->nb_samples;
*got_packet_ptr = 1;
return 0;
}
const FFCodec ff_adpcm_adx_encoder = {
.p.name = "adpcm_adx",
CODEC_LONG_NAME("SEGA CRI ADX ADPCM"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_ADPCM_ADX,
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY,
.priv_data_size = sizeof(ADXContext),
.init = adx_encode_init,
FF_CODEC_ENCODE_CB(adx_encode_frame),
.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
};
|
