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)
32 /** speech data history (spec: SB).
33 * Its first 70 coefficients are updated only at backward filtering.
37 /// speech part of the gain autocorrelation (spec: REXP)
40 /** log-gain history (spec: SBLG).
41 * Its first 28 coefficients are updated only at backward filtering.
45 /// recursive part of the gain autocorrelation (spec: REXPLG)
49 static av_cold int ra288_decode_init(AVCodecContext *avctx)
51 avctx->sample_fmt = SAMPLE_FMT_FLT;
55 static inline float scalar_product_float(const float * v1, const float * v2,
66 static void apply_window(float *tgt, const float *m1, const float *m2, int n)
69 *tgt++ = *m1++ * *m2++;
72 static void decode(RA288Context *ractx, float gain, int cb_coef)
77 float *block = ractx->sp_hist + 70 + 36; // current block
78 float *gain_block = ractx->gain_hist + 28;
80 memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
82 /* block 46 of G.728 spec */
84 for (i=0; i < 10; i++)
85 sum -= gain_block[9-i] * ractx->gain_lpc[i];
87 /* block 47 of G.728 spec */
88 sum = av_clipf(sum, 0, 60);
90 /* block 48 of G.728 spec */
91 sumsum = exp(sum * 0.1151292546497) * gain; /* pow(10.0,sum/20)*gain */
94 buffer[i] = codetable[cb_coef][i] * sumsum * (1./2048.);
96 sum = scalar_product_float(buffer, buffer, 5) / 5;
100 /* shift and store */
101 memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
103 gain_block[9] = 10 * log10(sum) - 32;
105 for (i=0; i < 5; i++) {
106 block[i] = buffer[i];
107 for (j=0; j < 36; j++)
108 block[i] -= block[i-1-j]*ractx->sp_lpc[j];
112 for (i=0; i < 5; i++)
113 block[i] = av_clipf(block[i], -4095, 4095);
116 static void convolve(float *tgt, const float *src, int len, int n)
119 tgt[n] = scalar_product_float(src, src - n, len);
124 * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
126 * @param order filter order
127 * @param n input length
128 * @param non_rec number of non-recursive samples
129 * @param out filter output
130 * @param hist pointer to the input history of the filter
131 * @param out pointer to the non-recursive part of the output
132 * @param out2 pointer to the recursive part of the output
133 * @param window pointer to the windowing function table
135 static void do_hybrid_window(int order, int n, int non_rec, float *out,
136 float *hist, float *out2, const float *window)
139 float buffer1[order + 1];
140 float buffer2[order + 1];
141 float work[order + n + non_rec];
143 apply_window(work, window, hist, order + n + non_rec);
145 convolve(buffer1, work + order , n , order);
146 convolve(buffer2, work + order + n, non_rec, order);
148 for (i=0; i <= order; i++) {
149 out2[i] = out2[i] * 0.5625 + buffer1[i];
150 out [i] = out2[i] + buffer2[i];
153 /* Multiply by the white noise correcting factor (WNCF). */
158 * Backward synthesis filter, find the LPC coefficients from past speech data.
160 static void backward_filter(RA288Context *ractx)
162 float temp1[37]; // RTMP in the spec
163 float temp2[11]; // GPTPMP in the spec
165 do_hybrid_window(36, 40, 35, temp1, ractx->sp_hist,
166 ractx->sp_rec, syn_window);
168 if (!compute_lpc_coefs(temp1, 36, ractx->sp_lpc, 0, 1, 1))
169 apply_window(ractx->sp_lpc, ractx->sp_lpc, syn_bw_tab, 36);
171 do_hybrid_window(10, 8, 20, temp2, ractx->gain_hist,
172 ractx->gain_rec, gain_window);
174 if (!compute_lpc_coefs(temp2, 10, ractx->gain_lpc, 0, 1, 1))
175 apply_window(ractx->gain_lpc, ractx->gain_lpc, gain_bw_tab, 10);
177 memmove(ractx->gain_hist, ractx->gain_hist + 8,
178 28*sizeof(*ractx->gain_hist));
180 memmove(ractx->sp_hist , ractx->sp_hist + 40,
181 70*sizeof(*ractx->sp_hist ));
184 static int ra288_decode_frame(AVCodecContext * avctx, void *data,
185 int *data_size, const uint8_t * buf,
190 RA288Context *ractx = avctx->priv_data;
193 if (buf_size < avctx->block_align) {
194 av_log(avctx, AV_LOG_ERROR,
195 "Error! Input buffer is too small [%d<%d]\n",
196 buf_size, avctx->block_align);
200 if (*data_size < 32*5*4)
203 init_get_bits(&gb, buf, avctx->block_align * 8);
205 for (i=0; i < 32; i++) {
206 float gain = amptable[get_bits(&gb, 3)];
207 int cb_coef = get_bits(&gb, 6 + (i&1));
209 decode(ractx, gain, cb_coef);
211 for (j=0; j < 5; j++)
212 *(out++) = (1/4096.) * ractx->sp_hist[70 + 36 + j];
215 backward_filter(ractx);
218 *data_size = (char *)out - (char *)data;
219 return avctx->block_align;
222 AVCodec ra_288_decoder =
227 sizeof(RA288Context),
232 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),