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
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
|
/*
* MS ADPCM decoder
*
* This file is responsible for decoding Microsoft ADPCM data.
* Details about the data format can be found here:
* http://www.pcisys.net/~melanson/codecs/
*
* Copyright (c) 2002 Mike Melanson
*
* This file is part of MPlayer.
*
* MPlayer is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* MPlayer 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include "config.h"
#include "libavutil/common.h"
#include "libavutil/intreadwrite.h"
#include "mpbswap.h"
#include "ad_internal.h"
static const ad_info_t info =
{
"MS ADPCM audio decoder",
"msadpcm",
"Nick Kurshev",
"Mike Melanson",
""
};
LIBAD_EXTERN(msadpcm)
static const int ms_adapt_table[] =
{
230, 230, 230, 230, 307, 409, 512, 614,
768, 614, 512, 409, 307, 230, 230, 230
};
static const uint8_t ms_adapt_coeff1[] =
{
64, 128, 0, 48, 60, 115, 98
};
static const int8_t ms_adapt_coeff2[] =
{
0, -64, 0, 16, 0, -52, -58
};
#define MS_ADPCM_PREAMBLE_SIZE 6
#define LE_16(x) ((int16_t)AV_RL16(x))
// clamp a number between 0 and 88
#define CLAMP_0_TO_88(x) x = av_clip(x, 0, 88);
// clamp a number within a signed 16-bit range
#define CLAMP_S16(x) x = av_clip_int16(x);
// clamp a number above 16
#define CLAMP_ABOVE_16(x) if (x < 16) x = 16;
// sign extend a 4-bit value
#define SE_4BIT(x) if (x & 0x8) x -= 0x10;
static int preinit(sh_audio_t *sh_audio)
{
sh_audio->audio_out_minsize = sh_audio->wf->nBlockAlign * 4;
sh_audio->ds->ss_div =
(sh_audio->wf->nBlockAlign - MS_ADPCM_PREAMBLE_SIZE) * 2;
sh_audio->audio_in_minsize =
sh_audio->ds->ss_mul = sh_audio->wf->nBlockAlign;
return 1;
}
static int init(sh_audio_t *sh_audio)
{
sh_audio->channels=sh_audio->wf->nChannels;
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
sh_audio->i_bps = sh_audio->wf->nBlockAlign *
(sh_audio->channels*sh_audio->samplerate) / sh_audio->ds->ss_div;
sh_audio->samplesize=2;
return 1;
}
static void uninit(sh_audio_t *sh_audio)
{
}
static int control(sh_audio_t *sh_audio,int cmd,void* arg, ...)
{
if(cmd==ADCTRL_SKIP_FRAME){
demux_read_data(sh_audio->ds, sh_audio->a_in_buffer,sh_audio->ds->ss_mul);
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}
static inline int check_coeff(uint8_t c) {
if (c > 6) {
mp_msg(MSGT_DECAUDIO, MSGL_WARN,
"MS ADPCM: coefficient (%d) out of range (should be [0..6])\n",
c);
c = 6;
}
return c;
}
static int ms_adpcm_decode_block(unsigned short *output, unsigned char *input,
int channels, int block_size)
{
int current_channel = 0;
int coeff_idx;
int idelta[2];
int sample1[2];
int sample2[2];
int coeff1[2];
int coeff2[2];
int stream_ptr = 0;
int out_ptr = 0;
int upper_nibble = 1;
int nibble;
int snibble; // signed nibble
int predictor;
if (channels != 1) channels = 2;
if (block_size < 7 * channels)
return -1;
// fetch the header information, in stereo if both channels are present
coeff_idx = check_coeff(input[stream_ptr]);
coeff1[0] = ms_adapt_coeff1[coeff_idx];
coeff2[0] = ms_adapt_coeff2[coeff_idx];
stream_ptr++;
if (channels == 2)
{
coeff_idx = check_coeff(input[stream_ptr]);
coeff1[1] = ms_adapt_coeff1[coeff_idx];
coeff2[1] = ms_adapt_coeff2[coeff_idx];
stream_ptr++;
}
idelta[0] = LE_16(&input[stream_ptr]);
stream_ptr += 2;
if (channels == 2)
{
idelta[1] = LE_16(&input[stream_ptr]);
stream_ptr += 2;
}
sample1[0] = LE_16(&input[stream_ptr]);
stream_ptr += 2;
if (channels == 2)
{
sample1[1] = LE_16(&input[stream_ptr]);
stream_ptr += 2;
}
sample2[0] = LE_16(&input[stream_ptr]);
stream_ptr += 2;
if (channels == 2)
{
sample2[1] = LE_16(&input[stream_ptr]);
stream_ptr += 2;
}
if (channels == 1)
{
output[out_ptr++] = sample2[0];
output[out_ptr++] = sample1[0];
} else {
output[out_ptr++] = sample2[0];
output[out_ptr++] = sample2[1];
output[out_ptr++] = sample1[0];
output[out_ptr++] = sample1[1];
}
while (stream_ptr < block_size)
{
// get the next nibble
if (upper_nibble)
nibble = snibble = input[stream_ptr] >> 4;
else
nibble = snibble = input[stream_ptr++] & 0x0F;
upper_nibble ^= 1;
SE_4BIT(snibble);
// should this really be a division and not a shift?
// coefficients were originally scaled by for, which might have
// been an optimization for 8-bit CPUs _if_ a shift is correct
predictor = (
((sample1[current_channel] * coeff1[current_channel]) +
(sample2[current_channel] * coeff2[current_channel])) / 64) +
(snibble * idelta[current_channel]);
CLAMP_S16(predictor);
sample2[current_channel] = sample1[current_channel];
sample1[current_channel] = predictor;
output[out_ptr++] = predictor;
// compute the next adaptive scale factor (a.k.a. the variable idelta)
idelta[current_channel] =
(ms_adapt_table[nibble] * idelta[current_channel]) / 256;
CLAMP_ABOVE_16(idelta[current_channel]);
// toggle the channel
current_channel ^= channels - 1;
}
return (block_size - (MS_ADPCM_PREAMBLE_SIZE * channels)) * 2;
}
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
int res;
if (demux_read_data(sh_audio->ds, sh_audio->a_in_buffer,
sh_audio->ds->ss_mul) !=
sh_audio->ds->ss_mul)
return -1; /* EOF */
res = ms_adpcm_decode_block(
(unsigned short*)buf, sh_audio->a_in_buffer,
sh_audio->wf->nChannels, sh_audio->wf->nBlockAlign);
return res < 0 ? res : 2 * res;
}
|
