1/* 2 * RealAudio 2.0 (28.8K) 3 * Copyright (c) 2003 the ffmpeg project 4 * 5 * This file is part of FFmpeg. 6 * 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. 11 * 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. 16 * 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 20 */ 21 22#include "avcodec.h" 23#define ALT_BITSTREAM_READER_LE 24#include "get_bits.h" 25#include "ra288.h" 26#include "lpc.h" 27#include "celp_math.h" 28#include "celp_filters.h" 29 30typedef struct { 31 float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A) 32 float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB) 33 34 /** speech data history (spec: SB). 35 * Its first 70 coefficients are updated only at backward filtering. 36 */ 37 float sp_hist[111]; 38 39 /// speech part of the gain autocorrelation (spec: REXP) 40 float sp_rec[37]; 41 42 /** log-gain history (spec: SBLG). 43 * Its first 28 coefficients are updated only at backward filtering. 44 */ 45 float gain_hist[38]; 46 47 /// recursive part of the gain autocorrelation (spec: REXPLG) 48 float gain_rec[11]; 49} RA288Context; 50 51static av_cold int ra288_decode_init(AVCodecContext *avctx) 52{ 53 avctx->sample_fmt = SAMPLE_FMT_FLT; 54 return 0; 55} 56 57static void apply_window(float *tgt, const float *m1, const float *m2, int n) 58{ 59 while (n--) 60 *tgt++ = *m1++ * *m2++; 61} 62 63static void convolve(float *tgt, const float *src, int len, int n) 64{ 65 for (; n >= 0; n--) 66 tgt[n] = ff_dot_productf(src, src - n, len); 67 68} 69 70static void decode(RA288Context *ractx, float gain, int cb_coef) 71{ 72 int i; 73 double sumsum; 74 float sum, buffer[5]; 75 float *block = ractx->sp_hist + 70 + 36; // current block 76 float *gain_block = ractx->gain_hist + 28; 77 78 memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block)); 79 80 /* block 46 of G.728 spec */ 81 sum = 32.; 82 for (i=0; i < 10; i++) 83 sum -= gain_block[9-i] * ractx->gain_lpc[i]; 84 85 /* block 47 of G.728 spec */ 86 sum = av_clipf(sum, 0, 60); 87 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)); 91 92 for (i=0; i < 5; i++) 93 buffer[i] = codetable[cb_coef][i] * sumsum; 94 95 sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.); 96 97 sum = FFMAX(sum, 1); 98 99 /* shift and store */ 100 memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block)); 101 102 gain_block[9] = 10 * log10(sum) - 32; 103 104 ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36); 105} 106 107/** 108 * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification. 109 * 110 * @param order filter order 111 * @param n input length 112 * @param non_rec number of non-recursive samples 113 * @param out filter output 114 * @param hist pointer to the input history of the filter 115 * @param out pointer to the non-recursive part of the output 116 * @param out2 pointer to the recursive part of the output 117 * @param window pointer to the windowing function table 118 */ 119static void do_hybrid_window(int order, int n, int non_rec, float *out, 120 float *hist, float *out2, const float *window) 121{ 122 int i; 123 float buffer1[order + 1]; 124 float buffer2[order + 1]; 125 float work[order + n + non_rec]; 126 127 apply_window(work, window, hist, order + n + non_rec); 128 129 convolve(buffer1, work + order , n , order); 130 convolve(buffer2, work + order + n, non_rec, order); 131 132 for (i=0; i <= order; i++) { 133 out2[i] = out2[i] * 0.5625 + buffer1[i]; 134 out [i] = out2[i] + buffer2[i]; 135 } 136 137 /* Multiply by the white noise correcting factor (WNCF). */ 138 *out *= 257./256.; 139} 140 141/** 142 * Backward synthesis filter, find the LPC coefficients from past speech data. 143 */ 144static void backward_filter(float *hist, float *rec, const float *window, 145 float *lpc, const float *tab, 146 int order, int n, int non_rec, int move_size) 147{ 148 float temp[order+1]; 149 150 do_hybrid_window(order, n, non_rec, temp, hist, rec, window); 151 152 if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1)) 153 apply_window(lpc, lpc, tab, order); 154 155 memmove(hist, hist + n, move_size*sizeof(*hist)); 156} 157 158static int ra288_decode_frame(AVCodecContext * avctx, void *data, 159 int *data_size, AVPacket *avpkt) 160{ 161 const uint8_t *buf = avpkt->data; 162 int buf_size = avpkt->size; 163 float *out = data; 164 int i, j; 165 RA288Context *ractx = avctx->priv_data; 166 GetBitContext gb; 167 168 if (buf_size < avctx->block_align) { 169 av_log(avctx, AV_LOG_ERROR, 170 "Error! Input buffer is too small [%d<%d]\n", 171 buf_size, avctx->block_align); 172 return 0; 173 } 174 175 if (*data_size < 32*5*4) 176 return -1; 177 178 init_get_bits(&gb, buf, avctx->block_align * 8); 179 180 for (i=0; i < 32; i++) { 181 float gain = amptable[get_bits(&gb, 3)]; 182 int cb_coef = get_bits(&gb, 6 + (i&1)); 183 184 decode(ractx, gain, cb_coef); 185 186 for (j=0; j < 5; j++) 187 *(out++) = ractx->sp_hist[70 + 36 + j]; 188 189 if ((i & 7) == 3) { 190 backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window, 191 ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70); 192 193 backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window, 194 ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28); 195 } 196 } 197 198 *data_size = (char *)out - (char *)data; 199 return avctx->block_align; 200} 201 202AVCodec ra_288_decoder = 203{ 204 "real_288", 205 AVMEDIA_TYPE_AUDIO, 206 CODEC_ID_RA_288, 207 sizeof(RA288Context), 208 ra288_decode_init, 209 NULL, 210 NULL, 211 ra288_decode_frame, 212 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"), 213}; 214