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
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
|
/*
* RealAudio 2.0 (28.8K)
* Copyright (c) 2003 the ffmpeg project
*
* 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"
#define ALT_BITSTREAM_READER_LE
#include "bitstream.h"
#include "ra288.h"
typedef struct {
float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A)
float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB)
float sp_hist[111]; ///< Speech data history (spec: SB)
/** Speech part of the gain autocorrelation (spec: REXP) */
float sp_rec[37];
float gain_hist[38]; ///< Log-gain history (spec: SBLG)
/** Recursive part of the gain autocorrelation (spec: REXPLG) */
float gain_rec[11];
float sp_block[41]; ///< Speech data of four blocks (spec: STTMP)
float gain_block[10]; ///< Gain data of four blocks (spec: GSTATE)
} RA288Context;
static inline float scalar_product_float(const float * v1, const float * v2,
int size)
{
float res = 0.;
while (size--)
res += *v1++ * *v2++;
return res;
}
static void colmult(float *tgt, const float *m1, const float *m2, int n)
{
while (n--)
*tgt++ = *m1++ * *m2++;
}
static void decode(RA288Context *ractx, float gain, int cb_coef)
{
int x, y;
double sumsum;
float sum, buffer[5];
memmove(ractx->sp_block + 5, ractx->sp_block, 36*sizeof(*ractx->sp_block));
for (x=4; x >= 0; x--)
ractx->sp_block[x] = -scalar_product_float(ractx->sp_block + x + 1,
ractx->sp_lpc, 36);
/* block 46 of G.728 spec */
sum = 32. - scalar_product_float(ractx->gain_lpc, ractx->gain_block, 10);
/* block 47 of G.728 spec */
sum = av_clipf(sum, 0, 60);
/* block 48 of G.728 spec */
sumsum = exp(sum * 0.1151292546497) * gain; /* pow(10.0,sum/20)*gain */
for (x=0; x < 5; x++)
buffer[x] = codetable[cb_coef][x] * sumsum;
sum = scalar_product_float(buffer, buffer, 5) / 5;
sum = FFMAX(sum, 1);
/* shift and store */
memmove(ractx->gain_block, ractx->gain_block - 1,
10 * sizeof(*ractx->gain_block));
*ractx->gain_block = 10 * log10(sum) - 32;
for (x=1; x < 5; x++)
for (y=x-1; y >= 0; y--)
buffer[x] -= ractx->sp_lpc[x-y-1] * buffer[y];
/* output */
for (x=0; x < 5; x++)
ractx->sp_block[4-x] =
av_clipf(ractx->sp_block[4-x] + buffer[x], -4095, 4095);
}
/**
* Converts autocorrelation coefficients to LPC coefficients using the
* Levinson-Durbin algorithm. See blocks 37 and 50 of the G.728 specification.
*
* @return 0 if success, -1 if fail
*/
static int eval_lpc_coeffs(const float *in, float *tgt, int n)
{
int x, y;
double f0, f1, f2;
if (in[n] == 0)
return -1;
if ((f0 = *in) <= 0)
return -1;
in--; // To avoid a -1 subtraction in the inner loop
for (x=1; x <= n; x++) {
f1 = in[x+1];
for (y=0; y < x - 1; y++)
f1 += in[x-y]*tgt[y];
tgt[x-1] = f2 = -f1/f0;
for (y=0; y < x >> 1; y++) {
float temp = tgt[y] + tgt[x-y-2]*f2;
tgt[x-y-2] += tgt[y]*f2;
tgt[y] = temp;
}
if ((f0 += f1*f2) < 0)
return -1;
}
return 0;
}
static void prodsum(float *tgt, const float *src, int len, int n)
{
for (; n >= 0; n--)
tgt[n] = scalar_product_float(src, src - n, len);
}
/**
* Hybrid window filtering. See blocks 36 and 49 of the G.728 specification.
*
* @note This function is slightly different from that described in the spec.
* It expects in[0] to be the newest sample and in[n-1] to be the oldest
* one stored. The spec has in the more ordinary way (in[0] the oldest
* and in[n-1] the newest).
*
* @param order the order of the filter
* @param n the length of the input
* @param non_rec the number of non-recursive samples
* @param out the filter output
* @param in pointer to the input of the filter
* @param hist pointer to the input history of the filter. It is updated by
* this function.
* @param out pointer to the non-recursive part of the output
* @param out2 pointer to the recursive part of the output
* @param window pointer to the windowing function table
*/
static void do_hybrid_window(int order, int n, int non_rec, const float *in,
float *out, float *hist, float *out2,
const float *window)
{
unsigned int x;
float buffer1[order + 1];
float buffer2[order + 1];
float work[order + n + non_rec];
/* update history */
memmove(hist, hist + n, (order + non_rec)*sizeof(*hist));
for (x=0; x < n; x++)
hist[order + non_rec + x] = in[n-x-1];
colmult(work, window, hist, order + n + non_rec);
prodsum(buffer1, work + order , n , order);
prodsum(buffer2, work + order + n, non_rec, order);
for (x=0; x <= order; x++) {
out2[x] = out2[x] * 0.5625 + buffer1[x];
out [x] = out2[x] + buffer2[x];
}
/* Multiply by the white noise correcting factor (WNCF) */
*out *= 257./256.;
}
/**
* Backward synthesis filter. Find the LPC coefficients from past speech data.
*/
static void backward_filter(RA288Context *ractx)
{
float temp1[37]; // RTMP in the spec
float temp2[11]; // GPTPMP in the spec
do_hybrid_window(36, 40, 35, ractx->sp_block, temp1, ractx->sp_hist,
ractx->sp_rec, syn_window);
if (!eval_lpc_coeffs(temp1, ractx->sp_lpc, 36))
colmult(ractx->sp_lpc, ractx->sp_lpc, syn_bw_tab, 36);
do_hybrid_window(10, 8, 20, ractx->gain_block, temp2, ractx->gain_hist,
ractx->gain_rec, gain_window);
if (!eval_lpc_coeffs(temp2, ractx->gain_lpc, 10))
colmult(ractx->gain_lpc, ractx->gain_lpc, gain_bw_tab, 10);
}
static int ra288_decode_frame(AVCodecContext * avctx, void *data,
int *data_size, const uint8_t * buf,
int buf_size)
{
int16_t *out = data;
int x, y;
RA288Context *ractx = avctx->priv_data;
GetBitContext gb;
if (buf_size < avctx->block_align) {
av_log(avctx, AV_LOG_ERROR,
"Error! Input buffer is too small [%d<%d]\n",
buf_size, avctx->block_align);
return 0;
}
init_get_bits(&gb, buf, avctx->block_align * 8);
for (x=0; x < 32; x++) {
float gain = amptable[get_bits(&gb, 3)];
int cb_coef = get_bits(&gb, 6 + (x&1));
decode(ractx, gain, cb_coef);
for (y=0; y < 5; y++)
*(out++) = 8 * ractx->sp_block[4 - y];
if ((x & 7) == 3)
backward_filter(ractx);
}
*data_size = (char *)out - (char *)data;
return avctx->block_align;
}
AVCodec ra_288_decoder =
{
"real_288",
CODEC_TYPE_AUDIO,
CODEC_ID_RA_288,
sizeof(RA288Context),
NULL,
NULL,
NULL,
ra288_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
};
|