1/* 2 * RealAudio 2.0 (28.8K) 3 * Copyright (c) 2003 the ffmpeg project 4 * 5 * This file is part of Libav. 6 * 7 * Libav 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 * Libav 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 Libav; 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 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#include "dsputil.h" 30 31#define MAX_BACKWARD_FILTER_ORDER 36 32#define MAX_BACKWARD_FILTER_LEN 40 33#define MAX_BACKWARD_FILTER_NONREC 35 34 35#define RA288_BLOCK_SIZE 5 36#define RA288_BLOCKS_PER_FRAME 32 37 38typedef struct { 39 AVFrame frame; 40 DSPContext dsp; 41 DECLARE_ALIGNED(16, float, sp_lpc)[FFALIGN(36, 8)]; ///< LPC coefficients for speech data (spec: A) 42 DECLARE_ALIGNED(16, float, gain_lpc)[FFALIGN(10, 8)]; ///< LPC coefficients for gain (spec: GB) 43 44 /** speech data history (spec: SB). 45 * Its first 70 coefficients are updated only at backward filtering. 46 */ 47 float sp_hist[111]; 48 49 /// speech part of the gain autocorrelation (spec: REXP) 50 float sp_rec[37]; 51 52 /** log-gain history (spec: SBLG). 53 * Its first 28 coefficients are updated only at backward filtering. 54 */ 55 float gain_hist[38]; 56 57 /// recursive part of the gain autocorrelation (spec: REXPLG) 58 float gain_rec[11]; 59} RA288Context; 60 61static av_cold int ra288_decode_init(AVCodecContext *avctx) 62{ 63 RA288Context *ractx = avctx->priv_data; 64 avctx->sample_fmt = AV_SAMPLE_FMT_FLT; 65 dsputil_init(&ractx->dsp, avctx); 66 67 avcodec_get_frame_defaults(&ractx->frame); 68 avctx->coded_frame = &ractx->frame; 69 70 return 0; 71} 72 73static void convolve(float *tgt, const float *src, int len, int n) 74{ 75 for (; n >= 0; n--) 76 tgt[n] = ff_dot_productf(src, src - n, len); 77 78} 79 80static void decode(RA288Context *ractx, float gain, int cb_coef) 81{ 82 int i; 83 double sumsum; 84 float sum, buffer[5]; 85 float *block = ractx->sp_hist + 70 + 36; // current block 86 float *gain_block = ractx->gain_hist + 28; 87 88 memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block)); 89 90 /* block 46 of G.728 spec */ 91 sum = 32.; 92 for (i=0; i < 10; i++) 93 sum -= gain_block[9-i] * ractx->gain_lpc[i]; 94 95 /* block 47 of G.728 spec */ 96 sum = av_clipf(sum, 0, 60); 97 98 /* block 48 of G.728 spec */ 99 /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */ 100 sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23)); 101 102 for (i=0; i < 5; i++) 103 buffer[i] = codetable[cb_coef][i] * sumsum; 104 105 sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.); 106 107 sum = FFMAX(sum, 1); 108 109 /* shift and store */ 110 memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block)); 111 112 gain_block[9] = 10 * log10(sum) - 32; 113 114 ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36); 115} 116 117/** 118 * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification. 119 * 120 * @param order filter order 121 * @param n input length 122 * @param non_rec number of non-recursive samples 123 * @param out filter output 124 * @param hist pointer to the input history of the filter 125 * @param out pointer to the non-recursive part of the output 126 * @param out2 pointer to the recursive part of the output 127 * @param window pointer to the windowing function table 128 */ 129static void do_hybrid_window(RA288Context *ractx, 130 int order, int n, int non_rec, float *out, 131 float *hist, float *out2, const float *window) 132{ 133 int i; 134 float buffer1[MAX_BACKWARD_FILTER_ORDER + 1]; 135 float buffer2[MAX_BACKWARD_FILTER_ORDER + 1]; 136 LOCAL_ALIGNED_16(float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER + 137 MAX_BACKWARD_FILTER_LEN + 138 MAX_BACKWARD_FILTER_NONREC, 8)]); 139 140 ractx->dsp.vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 8)); 141 142 convolve(buffer1, work + order , n , order); 143 convolve(buffer2, work + order + n, non_rec, order); 144 145 for (i=0; i <= order; i++) { 146 out2[i] = out2[i] * 0.5625 + buffer1[i]; 147 out [i] = out2[i] + buffer2[i]; 148 } 149 150 /* Multiply by the white noise correcting factor (WNCF). */ 151 *out *= 257./256.; 152} 153 154/** 155 * Backward synthesis filter, find the LPC coefficients from past speech data. 156 */ 157static void backward_filter(RA288Context *ractx, 158 float *hist, float *rec, const float *window, 159 float *lpc, const float *tab, 160 int order, int n, int non_rec, int move_size) 161{ 162 float temp[MAX_BACKWARD_FILTER_ORDER+1]; 163 164 do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window); 165 166 if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1)) 167 ractx->dsp.vector_fmul(lpc, lpc, tab, FFALIGN(order, 8)); 168 169 memmove(hist, hist + n, move_size*sizeof(*hist)); 170} 171 172static int ra288_decode_frame(AVCodecContext * avctx, void *data, 173 int *got_frame_ptr, AVPacket *avpkt) 174{ 175 const uint8_t *buf = avpkt->data; 176 int buf_size = avpkt->size; 177 float *out; 178 int i, ret; 179 RA288Context *ractx = avctx->priv_data; 180 GetBitContext gb; 181 182 if (buf_size < avctx->block_align) { 183 av_log(avctx, AV_LOG_ERROR, 184 "Error! Input buffer is too small [%d<%d]\n", 185 buf_size, avctx->block_align); 186 return AVERROR_INVALIDDATA; 187 } 188 189 /* get output buffer */ 190 ractx->frame.nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME; 191 if ((ret = avctx->get_buffer(avctx, &ractx->frame)) < 0) { 192 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); 193 return ret; 194 } 195 out = (float *)ractx->frame.data[0]; 196 197 init_get_bits(&gb, buf, avctx->block_align * 8); 198 199 for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) { 200 float gain = amptable[get_bits(&gb, 3)]; 201 int cb_coef = get_bits(&gb, 6 + (i&1)); 202 203 decode(ractx, gain, cb_coef); 204 205 memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out)); 206 out += RA288_BLOCK_SIZE; 207 208 if ((i & 7) == 3) { 209 backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window, 210 ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70); 211 212 backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window, 213 ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28); 214 } 215 } 216 217 *got_frame_ptr = 1; 218 *(AVFrame *)data = ractx->frame; 219 220 return avctx->block_align; 221} 222 223AVCodec ff_ra_288_decoder = { 224 .name = "real_288", 225 .type = AVMEDIA_TYPE_AUDIO, 226 .id = CODEC_ID_RA_288, 227 .priv_data_size = sizeof(RA288Context), 228 .init = ra288_decode_init, 229 .decode = ra288_decode_frame, 230 .capabilities = CODEC_CAP_DR1, 231 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"), 232}; 233