2 * RealAudio 2.0 (28.8K)
3 * Copyright (c) 2003 the ffmpeg project
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
23 #define ALT_BITSTREAM_READER_LE
24 #include "bitstream.h"
34 float st1a[111], st1b[37], st1[37];
35 float st2a[38], st2b[11], st2[11];
40 static inline float scalar_product_float(const float * v1, const float * v2,
51 static void colmult(float *tgt, const float *m1, const float *m2, int n)
54 *(tgt++) = (*(m1++)) * (*(m2++));
57 /* Decode and produce output */
58 static void decode(RA288Context *ractx, float gain, int cb_coef)
64 memmove(ractx->sb + 5, ractx->sb, 36 * sizeof(*ractx->sb));
66 for (x=4; x >= 0; x--)
67 ractx->sb[x] = -scalar_product_float(ractx->sb + x + 1, ractx->pr1, 36);
69 /* convert log and do rms */
70 sum = 32. - scalar_product_float(ractx->pr2, ractx->lhist, 10);
72 sum = av_clipf(sum, 0, 60);
74 sumsum = exp(sum * 0.1151292546497) * gain; /* pow(10.0,sum/20)*f */
77 buffer[x] = codetable[cb_coef][x] * sumsum;
79 sum = scalar_product_float(buffer, buffer, 5) / 5;
84 memmove(ractx->lhist, ractx->lhist - 1, 10 * sizeof(*ractx->lhist));
86 *ractx->lhist = ractx->history[ractx->phase] = 10 * log10(sum) - 32;
89 for (y=x-1; y >= 0; y--)
90 buffer[x] -= ractx->pr1[x-y-1] * buffer[y];
93 for (x=0; x < 5; x++) {
94 ractx->output[ractx->phase*5+x] = ractx->sb[4-x] =
95 av_clipf(ractx->sb[4-x] + buffer[x], -4095, 4095);
100 * Converts autocorrelation coefficients to LPC coefficients using the
101 * Levinson-Durbin algorithm. See blocks 37 and 50 of the G.728 specification.
103 * @return 0 if success, -1 if fail
105 static int eval_lpc_coeffs(const float *in, float *tgt, int n)
116 in--; // To avoid a -1 subtraction in the inner loop
118 for (x=1; x <= n; x++) {
121 for (y=0; y < x - 1; y++)
122 f1 += in[x-y]*tgt[y];
124 tgt[x-1] = f2 = -f1/f0;
125 for (y=0; y < x >> 1; y++) {
126 float temp = tgt[y] + tgt[x-y-2]*f2;
127 tgt[x-y-2] += tgt[y]*f2;
130 if ((f0 += f1*f2) < 0)
137 static void prodsum(float *tgt, const float *src, int len, int n)
140 tgt[n] = scalar_product_float(src, src - n, len);
145 * Hybrid window filtering. See blocks 36 and 49 of the G.728 specification.
147 * @param order the order of the filter
148 * @param n the length of the input
149 * @param non_rec the number of non-recursive samples
150 * @param out the filter output
151 * @param in pointer to the input of the filter
152 * @param hist pointer to the input history of the filter. It is updated by
154 * @param out pointer to the non-recursive part of the output
155 * @param out2 pointer to the recursive part of the output
156 * @param window pointer to the windowing function table
158 static void do_hybrid_window(int order, int n, int non_rec, const float *in,
159 float *out, float *hist, float *out2,
168 memmove(hist , hist + n, (order + non_rec)*sizeof(*hist));
169 memcpy (hist + order + non_rec, in , n *sizeof(*hist));
171 colmult(work, window, hist, order + n + non_rec);
173 prodsum(buffer1, work + order , n , order);
174 prodsum(buffer2, work + order + n, non_rec, order);
176 for (x=0; x <= order; x++) {
177 out2[x] = out2[x] * 0.5625 + buffer1[x];
178 out [x] = out2[x] + buffer2[x];
181 /* Multiply by the white noise correcting factor (WNCF) */
186 * Backward synthesis filter. Find the LPC coefficients from past speech data.
188 static void backward_filter(RA288Context *ractx)
190 float buffer1[40], temp1[37];
191 float buffer2[8], temp2[11];
193 memcpy(buffer1 , ractx->output + 20, 20*sizeof(*buffer1));
194 memcpy(buffer1 + 20, ractx->output , 20*sizeof(*buffer1));
196 do_hybrid_window(36, 40, 35, buffer1, temp1, ractx->st1a, ractx->st1b,
199 if (!eval_lpc_coeffs(temp1, ractx->st1, 36))
200 colmult(ractx->pr1, ractx->st1, syn_bw_tab, 36);
202 memcpy(buffer2 , ractx->history + 4, 4*sizeof(*buffer2));
203 memcpy(buffer2 + 4, ractx->history , 4*sizeof(*buffer2));
205 do_hybrid_window(10, 8, 20, buffer2, temp2, ractx->st2a, ractx->st2b,
208 if (!eval_lpc_coeffs(temp2, ractx->st2, 10))
209 colmult(ractx->pr2, ractx->st2, gain_bw_tab, 10);
212 /* Decode a block (celp) */
213 static int ra288_decode_frame(AVCodecContext * avctx, void *data,
214 int *data_size, const uint8_t * buf,
219 RA288Context *ractx = avctx->priv_data;
222 if (buf_size < avctx->block_align) {
223 av_log(avctx, AV_LOG_ERROR,
224 "Error! Input buffer is too small [%d<%d]\n",
225 buf_size, avctx->block_align);
229 init_get_bits(&gb, buf, avctx->block_align * 8);
231 for (x=0; x < 32; x++) {
232 float gain = amptable[get_bits(&gb, 3)];
233 int cb_coef = get_bits(&gb, 6 + (x&1));
234 ractx->phase = x & 7;
235 decode(ractx, gain, cb_coef);
237 for (y=0; y < 5; y++)
238 *(out++) = 8 * ractx->output[ractx->phase*5 + y];
240 if (ractx->phase == 3)
241 backward_filter(ractx);
244 *data_size = (char *)out - (char *)data;
245 return avctx->block_align;
248 AVCodec ra_288_decoder =
253 sizeof(RA288Context),
258 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),