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"
29 float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A)
30 float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB)
33 float sp_hist[111]; ///< Speech data history (spec: SB)
35 /** Speech part of the gain autocorrelation (spec: REXP) */
38 float gain_hist[38]; ///< Log-gain history (spec: SBLG)
40 /** Recursive part of the gain autocorrelation (spec: REXPLG) */
47 static inline float scalar_product_float(const float * v1, const float * v2,
58 static void colmult(float *tgt, const float *m1, const float *m2, int n)
61 *tgt++ = *m1++ * *m2++;
64 /* Decode and produce output */
65 static void decode(RA288Context *ractx, float gain, int cb_coef)
71 memmove(ractx->sb + 5, ractx->sb, 36 * sizeof(*ractx->sb));
73 for (x=4; x >= 0; x--)
74 ractx->sb[x] = -scalar_product_float(ractx->sb + x + 1,
77 /* convert log and do rms */
78 sum = 32. - scalar_product_float(ractx->gain_lpc, ractx->lhist, 10);
80 sum = av_clipf(sum, 0, 60);
82 sumsum = exp(sum * 0.1151292546497) * gain; /* pow(10.0,sum/20)*f */
85 buffer[x] = codetable[cb_coef][x] * sumsum;
87 sum = scalar_product_float(buffer, buffer, 5) / 5;
92 memmove(ractx->lhist, ractx->lhist - 1, 10 * sizeof(*ractx->lhist));
94 *ractx->lhist = 10 * log10(sum) - 32;
97 for (y=x-1; y >= 0; y--)
98 buffer[x] -= ractx->sp_lpc[x-y-1] * buffer[y];
101 for (x=0; x < 5; x++) {
102 ractx->output[ractx->phase*5+x] = ractx->sb[4-x] =
103 av_clipf(ractx->sb[4-x] + buffer[x], -4095, 4095);
108 * Converts autocorrelation coefficients to LPC coefficients using the
109 * Levinson-Durbin algorithm. See blocks 37 and 50 of the G.728 specification.
111 * @return 0 if success, -1 if fail
113 static int eval_lpc_coeffs(const float *in, float *tgt, int n)
124 in--; // To avoid a -1 subtraction in the inner loop
126 for (x=1; x <= n; x++) {
129 for (y=0; y < x - 1; y++)
130 f1 += in[x-y]*tgt[y];
132 tgt[x-1] = f2 = -f1/f0;
133 for (y=0; y < x >> 1; y++) {
134 float temp = tgt[y] + tgt[x-y-2]*f2;
135 tgt[x-y-2] += tgt[y]*f2;
138 if ((f0 += f1*f2) < 0)
145 static void prodsum(float *tgt, const float *src, int len, int n)
148 tgt[n] = scalar_product_float(src, src - n, len);
153 * Hybrid window filtering. See blocks 36 and 49 of the G.728 specification.
155 * @param order the order of the filter
156 * @param n the length of the input
157 * @param non_rec the number of non-recursive samples
158 * @param out the filter output
159 * @param in pointer to the input of the filter
160 * @param hist pointer to the input history of the filter. It is updated by
162 * @param out pointer to the non-recursive part of the output
163 * @param out2 pointer to the recursive part of the output
164 * @param window pointer to the windowing function table
166 static void do_hybrid_window(int order, int n, int non_rec, const float *in,
167 float *out, float *hist, float *out2,
176 memmove(hist , hist + n, (order + non_rec)*sizeof(*hist));
177 memcpy (hist + order + non_rec, in , n *sizeof(*hist));
179 colmult(work, window, hist, order + n + non_rec);
181 prodsum(buffer1, work + order , n , order);
182 prodsum(buffer2, work + order + n, non_rec, order);
184 for (x=0; x <= order; x++) {
185 out2[x] = out2[x] * 0.5625 + buffer1[x];
186 out [x] = out2[x] + buffer2[x];
189 /* Multiply by the white noise correcting factor (WNCF) */
194 * Backward synthesis filter. Find the LPC coefficients from past speech data.
196 static void backward_filter(RA288Context *ractx)
198 float temp1[37]; // RTMP in the spec
199 float temp2[11]; // GPTPMP in the spec
203 for (i=0 ; i < 8; i++)
204 history[i] = ractx->lhist[7-i];
206 do_hybrid_window(36, 40, 35, ractx->output, temp1, ractx->sp_hist,
207 ractx->sp_rec, syn_window);
209 if (!eval_lpc_coeffs(temp1, ractx->sp_lpc, 36))
210 colmult(ractx->sp_lpc, ractx->sp_lpc, syn_bw_tab, 36);
212 do_hybrid_window(10, 8, 20, history, temp2, ractx->gain_hist,
213 ractx->gain_rec, gain_window);
215 if (!eval_lpc_coeffs(temp2, ractx->gain_lpc, 10))
216 colmult(ractx->gain_lpc, ractx->gain_lpc, gain_bw_tab, 10);
219 /* Decode a block (celp) */
220 static int ra288_decode_frame(AVCodecContext * avctx, void *data,
221 int *data_size, const uint8_t * buf,
226 RA288Context *ractx = avctx->priv_data;
229 if (buf_size < avctx->block_align) {
230 av_log(avctx, AV_LOG_ERROR,
231 "Error! Input buffer is too small [%d<%d]\n",
232 buf_size, avctx->block_align);
236 init_get_bits(&gb, buf, avctx->block_align * 8);
238 for (x=0; x < 32; x++) {
239 float gain = amptable[get_bits(&gb, 3)];
240 int cb_coef = get_bits(&gb, 6 + (x&1));
241 ractx->phase = (x + 4) & 7;
242 decode(ractx, gain, cb_coef);
244 for (y=0; y < 5; y++)
245 *(out++) = 8 * ractx->output[ractx->phase*5 + y];
247 if (ractx->phase == 7)
248 backward_filter(ractx);
251 *data_size = (char *)out - (char *)data;
252 return avctx->block_align;
255 AVCodec ra_288_decoder =
260 sizeof(RA288Context),
265 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),