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 "libavutil/channel_layout.h" 23#include "libavutil/float_dsp.h" 24#include "libavutil/internal.h" 25#include "libavutil/mem_internal.h" 26 27#define BITSTREAM_READER_LE 28#include "avcodec.h" 29#include "celp_filters.h" 30#include "codec_internal.h" 31#include "get_bits.h" 32#include "internal.h" 33#include "lpc.h" 34#include "ra288.h" 35 36#define MAX_BACKWARD_FILTER_ORDER 36 37#define MAX_BACKWARD_FILTER_LEN 40 38#define MAX_BACKWARD_FILTER_NONREC 35 39 40#define RA288_BLOCK_SIZE 5 41#define RA288_BLOCKS_PER_FRAME 32 42 43typedef struct RA288Context { 44 void (*vector_fmul)(float *dst, const float *src0, const float *src1, 45 int len); 46 DECLARE_ALIGNED(32, float, sp_lpc)[FFALIGN(36, 16)]; ///< LPC coefficients for speech data (spec: A) 47 DECLARE_ALIGNED(32, float, gain_lpc)[FFALIGN(10, 16)]; ///< LPC coefficients for gain (spec: GB) 48 49 /** speech data history (spec: SB). 50 * Its first 70 coefficients are updated only at backward filtering. 51 */ 52 float sp_hist[111]; 53 54 /// speech part of the gain autocorrelation (spec: REXP) 55 float sp_rec[37]; 56 57 /** log-gain history (spec: SBLG). 58 * Its first 28 coefficients are updated only at backward filtering. 59 */ 60 float gain_hist[38]; 61 62 /// recursive part of the gain autocorrelation (spec: REXPLG) 63 float gain_rec[11]; 64} RA288Context; 65 66static av_cold int ra288_decode_init(AVCodecContext *avctx) 67{ 68 RA288Context *ractx = avctx->priv_data; 69 AVFloatDSPContext *fdsp; 70 71 av_channel_layout_uninit(&avctx->ch_layout); 72 avctx->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO; 73 avctx->sample_fmt = AV_SAMPLE_FMT_FLT; 74 75 if (avctx->block_align != 38) { 76 av_log(avctx, AV_LOG_ERROR, "unsupported block align\n"); 77 return AVERROR_PATCHWELCOME; 78 } 79 80 fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT); 81 if (!fdsp) 82 return AVERROR(ENOMEM); 83 ractx->vector_fmul = fdsp->vector_fmul; 84 av_free(fdsp); 85 86 return 0; 87} 88 89static void convolve(float *tgt, const float *src, int len, int n) 90{ 91 for (; n >= 0; n--) 92 tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len); 93 94} 95 96static void decode(RA288Context *ractx, float gain, int cb_coef) 97{ 98 int i; 99 double sumsum; 100 float sum, buffer[5]; 101 float *block = ractx->sp_hist + 70 + 36; // current block 102 float *gain_block = ractx->gain_hist + 28; 103 104 memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block)); 105 106 /* block 46 of G.728 spec */ 107 sum = 32.0; 108 for (i=0; i < 10; i++) 109 sum -= gain_block[9-i] * ractx->gain_lpc[i]; 110 111 /* block 47 of G.728 spec */ 112 sum = av_clipf(sum, 0, 60); 113 114 /* block 48 of G.728 spec */ 115 /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */ 116 sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23)); 117 118 for (i=0; i < 5; i++) 119 buffer[i] = codetable[cb_coef][i] * sumsum; 120 121 sum = avpriv_scalarproduct_float_c(buffer, buffer, 5); 122 123 sum = FFMAX(sum, 5.0 / (1<<24)); 124 125 /* shift and store */ 126 memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block)); 127 128 gain_block[9] = 10 * log10(sum) + (10*log10(((1<<24)/5.)) - 32); 129 130 ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36); 131} 132 133/** 134 * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification. 135 * 136 * @param order filter order 137 * @param n input length 138 * @param non_rec number of non-recursive samples 139 * @param out filter output 140 * @param hist pointer to the input history of the filter 141 * @param out pointer to the non-recursive part of the output 142 * @param out2 pointer to the recursive part of the output 143 * @param window pointer to the windowing function table 144 */ 145static void do_hybrid_window(RA288Context *ractx, 146 int order, int n, int non_rec, float *out, 147 float *hist, float *out2, const float *window) 148{ 149 int i; 150 float buffer1[MAX_BACKWARD_FILTER_ORDER + 1]; 151 float buffer2[MAX_BACKWARD_FILTER_ORDER + 1]; 152 LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER + 153 MAX_BACKWARD_FILTER_LEN + 154 MAX_BACKWARD_FILTER_NONREC, 16)]); 155 156 av_assert2(order>=0); 157 158 ractx->vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16)); 159 160 convolve(buffer1, work + order , n , order); 161 convolve(buffer2, work + order + n, non_rec, order); 162 163 for (i=0; i <= order; i++) { 164 out2[i] = out2[i] * 0.5625 + buffer1[i]; 165 out [i] = out2[i] + buffer2[i]; 166 } 167 168 /* Multiply by the white noise correcting factor (WNCF). */ 169 *out *= 257.0 / 256.0; 170} 171 172/** 173 * Backward synthesis filter, find the LPC coefficients from past speech data. 174 */ 175static void backward_filter(RA288Context *ractx, 176 float *hist, float *rec, const float *window, 177 float *lpc, const float *tab, 178 int order, int n, int non_rec, int move_size) 179{ 180 float temp[MAX_BACKWARD_FILTER_ORDER+1]; 181 182 do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window); 183 184 if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1)) 185 ractx->vector_fmul(lpc, lpc, tab, FFALIGN(order, 16)); 186 187 memmove(hist, hist + n, move_size*sizeof(*hist)); 188} 189 190static int ra288_decode_frame(AVCodecContext * avctx, AVFrame *frame, 191 int *got_frame_ptr, AVPacket *avpkt) 192{ 193 const uint8_t *buf = avpkt->data; 194 int buf_size = avpkt->size; 195 float *out; 196 int i, ret; 197 RA288Context *ractx = avctx->priv_data; 198 GetBitContext gb; 199 200 if (buf_size < avctx->block_align) { 201 av_log(avctx, AV_LOG_ERROR, 202 "Error! Input buffer is too small [%d<%d]\n", 203 buf_size, avctx->block_align); 204 return AVERROR_INVALIDDATA; 205 } 206 207 ret = init_get_bits8(&gb, buf, avctx->block_align); 208 if (ret < 0) 209 return ret; 210 211 /* get output buffer */ 212 frame->nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME; 213 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) 214 return ret; 215 out = (float *)frame->data[0]; 216 217 for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) { 218 float gain = amptable[get_bits(&gb, 3)]; 219 int cb_coef = get_bits(&gb, 6 + (i&1)); 220 221 decode(ractx, gain, cb_coef); 222 223 memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out)); 224 out += RA288_BLOCK_SIZE; 225 226 if ((i & 7) == 3) { 227 backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window, 228 ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70); 229 230 backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window, 231 ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28); 232 } 233 } 234 235 *got_frame_ptr = 1; 236 237 return avctx->block_align; 238} 239 240const FFCodec ff_ra_288_decoder = { 241 .p.name = "real_288", 242 .p.long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"), 243 .p.type = AVMEDIA_TYPE_AUDIO, 244 .p.id = AV_CODEC_ID_RA_288, 245 .priv_data_size = sizeof(RA288Context), 246 .init = ra288_decode_init, 247 FF_CODEC_DECODE_CB(ra288_decode_frame), 248 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF, 249 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, 250}; 251