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
|
/*
* Real Audio 1.0 (14.4K)
*
* Copyright (c) 2008 Vitor Sessak
* Copyright (c) 2003 Nick Kurshev
* Based on public domain decoder at http://www.honeypot.net/audio
*
* 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/intmath.h"
#include "avcodec.h"
#include "get_bits.h"
#include "ra144.h"
static av_cold int ra144_decode_init(AVCodecContext * avctx)
{
RA144Context *ractx = avctx->priv_data;
ractx->avctx = avctx;
ractx->lpc_coef[0] = ractx->lpc_tables[0];
ractx->lpc_coef[1] = ractx->lpc_tables[1];
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}
static void do_output_subblock(RA144Context *ractx, const uint16_t *lpc_coefs,
int gval, GetBitContext *gb)
{
int cba_idx = get_bits(gb, 7); // index of the adaptive CB, 0 if none
int gain = get_bits(gb, 8);
int cb1_idx = get_bits(gb, 7);
int cb2_idx = get_bits(gb, 7);
ff_subblock_synthesis(ractx, lpc_coefs, cba_idx, cb1_idx, cb2_idx, gval,
gain);
}
/** Uncompress one block (20 bytes -> 160*2 bytes). */
static int ra144_decode_frame(AVCodecContext * avctx, void *vdata,
int *data_size, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
static const uint8_t sizes[10] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
unsigned int refl_rms[4]; // RMS of the reflection coefficients
uint16_t block_coefs[4][10]; // LPC coefficients of each sub-block
unsigned int lpc_refl[10]; // LPC reflection coefficients of the frame
int i, j;
int16_t *data = vdata;
unsigned int energy;
RA144Context *ractx = avctx->priv_data;
GetBitContext gb;
if (*data_size < 2*160)
return -1;
if(buf_size < 20) {
av_log(avctx, AV_LOG_ERROR,
"Frame too small (%d bytes). Truncated file?\n", buf_size);
*data_size = 0;
return buf_size;
}
init_get_bits(&gb, buf, 20 * 8);
for (i=0; i<10; i++)
lpc_refl[i] = ff_lpc_refl_cb[i][get_bits(&gb, sizes[i])];
ff_eval_coefs(ractx->lpc_coef[0], lpc_refl);
ractx->lpc_refl_rms[0] = ff_rms(lpc_refl);
energy = ff_energy_tab[get_bits(&gb, 5)];
refl_rms[0] = ff_interp(ractx, block_coefs[0], 1, 1, ractx->old_energy);
refl_rms[1] = ff_interp(ractx, block_coefs[1], 2,
energy <= ractx->old_energy,
ff_t_sqrt(energy*ractx->old_energy) >> 12);
refl_rms[2] = ff_interp(ractx, block_coefs[2], 3, 0, energy);
refl_rms[3] = ff_rescale_rms(ractx->lpc_refl_rms[0], energy);
ff_int_to_int16(block_coefs[3], ractx->lpc_coef[0]);
for (i=0; i < 4; i++) {
do_output_subblock(ractx, block_coefs[i], refl_rms[i], &gb);
for (j=0; j < BLOCKSIZE; j++)
*data++ = av_clip_int16(ractx->curr_sblock[j + 10] << 2);
}
ractx->old_energy = energy;
ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];
FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);
*data_size = 2*160;
return 20;
}
AVCodec ra_144_decoder =
{
"real_144",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_RA_144,
sizeof(RA144Context),
ra144_decode_init,
NULL,
NULL,
ra144_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
};
|
