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"
28 float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A)
29 float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB)
32 float sp_hist[111]; ///< Speech data history (spec: SB)
34 /** Speech part of the gain autocorrelation (spec: REXP) */
37 float gain_hist[38]; ///< Log-gain history (spec: SBLG)
39 /** Recursive part of the gain autocorrelation (spec: REXPLG) */
46 static inline float scalar_product_float(const float * v1, const float * v2,
57 static void colmult(float *tgt, const float *m1, const float *m2, int n)
60 *tgt++ = *m1++ * *m2++;
63 static void decode(RA288Context *ractx, float gain, int cb_coef)
69 memmove(ractx->sb + 5, ractx->sb, 36 * sizeof(*ractx->sb));
71 for (x=4; x >= 0; x--)
72 ractx->sb[x] = -scalar_product_float(ractx->sb + x + 1,
75 /* block 46 of G.728 spec */
76 sum = 32. - scalar_product_float(ractx->gain_lpc, ractx->lhist, 10);
78 /* block 47 of G.728 spec */
79 sum = av_clipf(sum, 0, 60);
81 /* block 48 of G.728 spec */
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->sb[4-x] = av_clipf(ractx->sb[4-x] + buffer[x], -4095, 4095);
106 * Converts autocorrelation coefficients to LPC coefficients using the
107 * Levinson-Durbin algorithm. See blocks 37 and 50 of the G.728 specification.
109 * @return 0 if success, -1 if fail
111 static int eval_lpc_coeffs(const float *in, float *tgt, int n)
122 in--; // To avoid a -1 subtraction in the inner loop
124 for (x=1; x <= n; x++) {
127 for (y=0; y < x - 1; y++)
128 f1 += in[x-y]*tgt[y];
130 tgt[x-1] = f2 = -f1/f0;
131 for (y=0; y < x >> 1; y++) {
132 float temp = tgt[y] + tgt[x-y-2]*f2;
133 tgt[x-y-2] += tgt[y]*f2;
136 if ((f0 += f1*f2) < 0)
143 static void prodsum(float *tgt, const float *src, int len, int n)
146 tgt[n] = scalar_product_float(src, src - n, len);
151 * Hybrid window filtering. See blocks 36 and 49 of the G.728 specification.
153 * @note This function is slightly different from that described in the spec.
154 * It expects in[0] to be the newest sample and in[n-1] to be the oldest
155 * one stored. The spec has in the more ordinary way (in[0] the oldest
156 * and in[n-1] the newest).
158 * @param order the order of the filter
159 * @param n the length of the input
160 * @param non_rec the number of non-recursive samples
161 * @param out the filter output
162 * @param in pointer to the input of the filter
163 * @param hist pointer to the input history of the filter. It is updated by
165 * @param out pointer to the non-recursive part of the output
166 * @param out2 pointer to the recursive part of the output
167 * @param window pointer to the windowing function table
169 static void do_hybrid_window(int order, int n, int non_rec, const float *in,
170 float *out, float *hist, float *out2,
174 float buffer1[order + 1];
175 float buffer2[order + 1];
176 float work[order + n + non_rec];
179 memmove(hist, hist + n, (order + non_rec)*sizeof(*hist));
181 for (x=0; x < n; x++)
182 hist[order + non_rec + x] = in[n-x-1];
184 colmult(work, window, hist, order + n + non_rec);
186 prodsum(buffer1, work + order , n , order);
187 prodsum(buffer2, work + order + n, non_rec, order);
189 for (x=0; x <= order; x++) {
190 out2[x] = out2[x] * 0.5625 + buffer1[x];
191 out [x] = out2[x] + buffer2[x];
194 /* Multiply by the white noise correcting factor (WNCF) */
199 * Backward synthesis filter. Find the LPC coefficients from past speech data.
201 static void backward_filter(RA288Context *ractx)
203 float temp1[37]; // RTMP in the spec
204 float temp2[11]; // GPTPMP in the spec
206 do_hybrid_window(36, 40, 35, ractx->sb, 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, ractx->lhist, 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 static int ra288_decode_frame(AVCodecContext * avctx, void *data,
220 int *data_size, const uint8_t * buf,
225 RA288Context *ractx = avctx->priv_data;
228 if (buf_size < avctx->block_align) {
229 av_log(avctx, AV_LOG_ERROR,
230 "Error! Input buffer is too small [%d<%d]\n",
231 buf_size, avctx->block_align);
235 init_get_bits(&gb, buf, avctx->block_align * 8);
237 for (x=0; x < 32; x++) {
238 float gain = amptable[get_bits(&gb, 3)];
239 int cb_coef = get_bits(&gb, 6 + (x&1));
240 ractx->phase = (x + 4) & 7;
241 decode(ractx, gain, cb_coef);
243 for (y=0; y < 5; y++)
244 *(out++) = 8 * ractx->sb[4 - y];
246 if (ractx->phase == 7)
247 backward_filter(ractx);
250 *data_size = (char *)out - (char *)data;
251 return avctx->block_align;
254 AVCodec ra_288_decoder =
259 sizeof(RA288Context),
264 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),