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"
27 #include "celp_math.h"
28 #include "celp_filters.h"
31 float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A)
32 float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB)
34 /** speech data history (spec: SB).
35 * Its first 70 coefficients are updated only at backward filtering.
39 /// speech part of the gain autocorrelation (spec: REXP)
42 /** log-gain history (spec: SBLG).
43 * Its first 28 coefficients are updated only at backward filtering.
47 /// recursive part of the gain autocorrelation (spec: REXPLG)
51 static av_cold int ra288_decode_init(AVCodecContext *avctx)
53 avctx->sample_fmt = SAMPLE_FMT_FLT;
57 static void apply_window(float *tgt, const float *m1, const float *m2, int n)
60 *tgt++ = *m1++ * *m2++;
63 static void convolve(float *tgt, const float *src, int len, int n)
66 tgt[n] = ff_dot_productf(src, src - n, len);
70 static void decode(RA288Context *ractx, float gain, int cb_coef)
75 float *block = ractx->sp_hist + 70 + 36; // current block
76 float *gain_block = ractx->gain_hist + 28;
78 memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
80 /* block 46 of G.728 spec */
82 for (i=0; i < 10; i++)
83 sum -= gain_block[9-i] * ractx->gain_lpc[i];
85 /* block 47 of G.728 spec */
86 sum = av_clipf(sum, 0, 60);
88 /* block 48 of G.728 spec */
89 /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
90 sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
93 buffer[i] = codetable[cb_coef][i] * sumsum;
95 sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.);
100 memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
102 gain_block[9] = 10 * log10(sum) - 32;
104 ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
107 for (i=0; i < 5; i++)
108 block[i] = av_clipf(block[i], -4095./4096., 4095./4096.);
112 * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
114 * @param order filter order
115 * @param n input length
116 * @param non_rec number of non-recursive samples
117 * @param out filter output
118 * @param hist pointer to the input history of the filter
119 * @param out pointer to the non-recursive part of the output
120 * @param out2 pointer to the recursive part of the output
121 * @param window pointer to the windowing function table
123 static void do_hybrid_window(int order, int n, int non_rec, float *out,
124 float *hist, float *out2, const float *window)
127 float buffer1[order + 1];
128 float buffer2[order + 1];
129 float work[order + n + non_rec];
131 apply_window(work, window, hist, order + n + non_rec);
133 convolve(buffer1, work + order , n , order);
134 convolve(buffer2, work + order + n, non_rec, order);
136 for (i=0; i <= order; i++) {
137 out2[i] = out2[i] * 0.5625 + buffer1[i];
138 out [i] = out2[i] + buffer2[i];
141 /* Multiply by the white noise correcting factor (WNCF). */
146 * Backward synthesis filter, find the LPC coefficients from past speech data.
148 static void backward_filter(float *hist, float *rec, const float *window,
149 float *lpc, const float *tab,
150 int order, int n, int non_rec, int move_size)
154 do_hybrid_window(order, n, non_rec, temp, hist, rec, window);
156 if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
157 apply_window(lpc, lpc, tab, order);
159 memmove(hist, hist + n, move_size*sizeof(*hist));
162 static int ra288_decode_frame(AVCodecContext * avctx, void *data,
163 int *data_size, const uint8_t * buf,
168 RA288Context *ractx = avctx->priv_data;
171 if (buf_size < avctx->block_align) {
172 av_log(avctx, AV_LOG_ERROR,
173 "Error! Input buffer is too small [%d<%d]\n",
174 buf_size, avctx->block_align);
178 if (*data_size < 32*5*4)
181 init_get_bits(&gb, buf, avctx->block_align * 8);
183 for (i=0; i < 32; i++) {
184 float gain = amptable[get_bits(&gb, 3)];
185 int cb_coef = get_bits(&gb, 6 + (i&1));
187 decode(ractx, gain, cb_coef);
189 for (j=0; j < 5; j++)
190 *(out++) = ractx->sp_hist[70 + 36 + j];
193 backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window,
194 ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
196 backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window,
197 ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
201 *data_size = (char *)out - (char *)data;
202 return avctx->block_align;
205 AVCodec ra_288_decoder =
210 sizeof(RA288Context),
215 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),