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 /* Decode and produce output */
52 static void decode(Real288_internal *glob, float gain, int cb_coef)
58 memmove(glob->sb + 5, glob->sb, 36 * sizeof(*glob->sb));
60 for (x=4; x >= 0; x--)
61 glob->sb[x] = -scalar_product_float(glob->sb + x + 1, glob->pr1, 36);
63 /* convert log and do rms */
64 sum = 32. - scalar_product_float(glob->pr2, glob->lhist, 10);
66 sum = av_clipf(sum, 0, 60);
68 sumsum = exp(sum * 0.1151292546497) * gain; /* pow(10.0,sum/20)*f */
71 buffer[x] = codetable[cb_coef][x] * sumsum;
73 sum = scalar_product_float(buffer, buffer, 5) / 5;
78 memmove(glob->lhist, glob->lhist - 1, 10 * sizeof(*glob->lhist));
80 *glob->lhist = glob->history[glob->phase] = 10 * log10(sum) - 32;
83 for (y=x-1; y >= 0; y--)
84 buffer[x] -= glob->pr1[x-y-1] * buffer[y];
87 for (x=0; x < 5; x++) {
88 glob->output[glob->phase*5+x] = glob->sb[4-x] =
89 av_clipf(glob->sb[4-x] + buffer[x], -4095, 4095);
94 static void colmult(float *tgt, const float *m1, const float *m2, int n)
97 *(tgt++) = (*(m1++)) * (*(m2++));
101 * Converts autocorrelation coefficients to LPC coefficients using the
102 * Levinson-Durbin algorithm. See blocks 37 and 50 of the G.728 specification.
104 * @return 0 if success, -1 if fail
106 static int eval_lpc_coeffs(const float *in, float *tgt, int n)
117 in--; // To avoid a -1 subtraction in the inner loop
119 for (x=1; x <= n; x++) {
122 for (y=0; y < x - 1; y++)
123 f1 += in[x-y]*tgt[y];
125 tgt[x-1] = f2 = -f1/f0;
126 for (y=0; y < x >> 1; y++) {
127 float temp = tgt[y] + tgt[x-y-2]*f2;
128 tgt[x-y-2] += tgt[y]*f2;
131 if ((f0 += f1*f2) < 0)
138 /* product sum (lsf) */
139 static void prodsum(float *tgt, const float *src, int len, int n)
142 tgt[n] = scalar_product_float(src, src - n, len);
147 * Hybrid window filtering. See blocks 36 and 49 of the G.728 specification.
149 * @param order the order of the filter
150 * @param n the length of the input
151 * @param non_rec the number of non-recursive samples
152 * @param out the filter output
153 * @param in pointer to the input of the filter
154 * @param hist pointer to the input history of the filter. It is updated by
156 * @param out pointer to the non-recursive part of the output
157 * @param out2 pointer to the recursive part of the output
158 * @param window pointer to the windowing function table
160 static void do_hybrid_window(int order, int n, int non_rec, const float *in,
161 float *out, float *hist, float *out2,
170 memmove(hist , hist + n, (order + non_rec)*sizeof(*hist));
171 memcpy (hist + order + non_rec, in , n *sizeof(*hist));
173 colmult(work, window, hist, order + n + non_rec);
175 prodsum(buffer1, work + order , n , order);
176 prodsum(buffer2, work + order + n, non_rec, order);
178 for (x=0; x <= order; x++) {
179 out2[x] = out2[x] * 0.5625 + buffer1[x];
180 out [x] = out2[x] + buffer2[x];
183 /* Multiply by the white noise correcting factor (WNCF) */
188 * Backward synthesis filter. Find the LPC coefficients from past speech data.
190 static void backward_filter(Real288_internal *glob)
192 float buffer1[40], temp1[37];
193 float buffer2[8], temp2[11];
195 memcpy(buffer1 , glob->output + 20, 20*sizeof(*buffer1));
196 memcpy(buffer1 + 20, glob->output , 20*sizeof(*buffer1));
198 do_hybrid_window(36, 40, 35, buffer1, temp1, glob->st1a, glob->st1b,
201 if (!eval_lpc_coeffs(temp1, glob->st1, 36))
202 colmult(glob->pr1, glob->st1, syn_bw_tab, 36);
204 memcpy(buffer2 , glob->history + 4, 4*sizeof(*buffer2));
205 memcpy(buffer2 + 4, glob->history , 4*sizeof(*buffer2));
207 do_hybrid_window(10, 8, 20, buffer2, temp2, glob->st2a, glob->st2b,
210 if (!eval_lpc_coeffs(temp2, glob->st2, 10))
211 colmult(glob->pr2, glob->st2, gain_bw_tab, 10);
214 /* Decode a block (celp) */
215 static int ra288_decode_frame(AVCodecContext * avctx, void *data,
216 int *data_size, const uint8_t * buf,
221 Real288_internal *glob = avctx->priv_data;
224 if (buf_size < avctx->block_align) {
225 av_log(avctx, AV_LOG_ERROR,
226 "Error! Input buffer is too small [%d<%d]\n",
227 buf_size, avctx->block_align);
231 init_get_bits(&gb, buf, avctx->block_align * 8);
233 for (x=0; x < 32; x++) {
234 float gain = amptable[get_bits(&gb, 3)];
235 int cb_coef = get_bits(&gb, 6 + (x&1));
237 decode(glob, gain, cb_coef);
239 for (y=0; y < 5; y++)
240 *(out++) = 8 * glob->output[glob->phase*5 + y];
242 if (glob->phase == 3)
243 backward_filter(glob);
246 *data_size = (char *)out - (char *)data;
247 return avctx->block_align;
250 AVCodec ra_288_decoder =
255 sizeof(Real288_internal),
260 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),