Libav
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00001 /* 00002 * RealAudio 2.0 (28.8K) 00003 * Copyright (c) 2003 the ffmpeg project 00004 * 00005 * This file is part of FFmpeg. 00006 * 00007 * FFmpeg is free software; you can redistribute it and/or 00008 * modify it under the terms of the GNU Lesser General Public 00009 * License as published by the Free Software Foundation; either 00010 * version 2.1 of the License, or (at your option) any later version. 00011 * 00012 * FFmpeg is distributed in the hope that it will be useful, 00013 * but WITHOUT ANY WARRANTY; without even the implied warranty of 00014 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 00015 * Lesser General Public License for more details. 00016 * 00017 * You should have received a copy of the GNU Lesser General Public 00018 * License along with FFmpeg; if not, write to the Free Software 00019 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 00020 */ 00021 00022 #include "avcodec.h" 00023 #define ALT_BITSTREAM_READER_LE 00024 #include "get_bits.h" 00025 #include "ra288.h" 00026 #include "lpc.h" 00027 #include "celp_math.h" 00028 #include "celp_filters.h" 00029 00030 typedef struct { 00031 float sp_lpc[36]; 00032 float gain_lpc[10]; 00033 00037 float sp_hist[111]; 00038 00040 float sp_rec[37]; 00041 00045 float gain_hist[38]; 00046 00048 float gain_rec[11]; 00049 } RA288Context; 00050 00051 static av_cold int ra288_decode_init(AVCodecContext *avctx) 00052 { 00053 avctx->sample_fmt = SAMPLE_FMT_FLT; 00054 return 0; 00055 } 00056 00057 static void apply_window(float *tgt, const float *m1, const float *m2, int n) 00058 { 00059 while (n--) 00060 *tgt++ = *m1++ * *m2++; 00061 } 00062 00063 static void convolve(float *tgt, const float *src, int len, int n) 00064 { 00065 for (; n >= 0; n--) 00066 tgt[n] = ff_dot_productf(src, src - n, len); 00067 00068 } 00069 00070 static void decode(RA288Context *ractx, float gain, int cb_coef) 00071 { 00072 int i; 00073 double sumsum; 00074 float sum, buffer[5]; 00075 float *block = ractx->sp_hist + 70 + 36; // current block 00076 float *gain_block = ractx->gain_hist + 28; 00077 00078 memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block)); 00079 00080 /* block 46 of G.728 spec */ 00081 sum = 32.; 00082 for (i=0; i < 10; i++) 00083 sum -= gain_block[9-i] * ractx->gain_lpc[i]; 00084 00085 /* block 47 of G.728 spec */ 00086 sum = av_clipf(sum, 0, 60); 00087 00088 /* block 48 of G.728 spec */ 00089 /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */ 00090 sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23)); 00091 00092 for (i=0; i < 5; i++) 00093 buffer[i] = codetable[cb_coef][i] * sumsum; 00094 00095 sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.); 00096 00097 sum = FFMAX(sum, 1); 00098 00099 /* shift and store */ 00100 memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block)); 00101 00102 gain_block[9] = 10 * log10(sum) - 32; 00103 00104 ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36); 00105 } 00106 00119 static void do_hybrid_window(int order, int n, int non_rec, float *out, 00120 float *hist, float *out2, const float *window) 00121 { 00122 int i; 00123 float buffer1[order + 1]; 00124 float buffer2[order + 1]; 00125 float work[order + n + non_rec]; 00126 00127 apply_window(work, window, hist, order + n + non_rec); 00128 00129 convolve(buffer1, work + order , n , order); 00130 convolve(buffer2, work + order + n, non_rec, order); 00131 00132 for (i=0; i <= order; i++) { 00133 out2[i] = out2[i] * 0.5625 + buffer1[i]; 00134 out [i] = out2[i] + buffer2[i]; 00135 } 00136 00137 /* Multiply by the white noise correcting factor (WNCF). */ 00138 *out *= 257./256.; 00139 } 00140 00144 static void backward_filter(float *hist, float *rec, const float *window, 00145 float *lpc, const float *tab, 00146 int order, int n, int non_rec, int move_size) 00147 { 00148 float temp[order+1]; 00149 00150 do_hybrid_window(order, n, non_rec, temp, hist, rec, window); 00151 00152 if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1)) 00153 apply_window(lpc, lpc, tab, order); 00154 00155 memmove(hist, hist + n, move_size*sizeof(*hist)); 00156 } 00157 00158 static int ra288_decode_frame(AVCodecContext * avctx, void *data, 00159 int *data_size, AVPacket *avpkt) 00160 { 00161 const uint8_t *buf = avpkt->data; 00162 int buf_size = avpkt->size; 00163 float *out = data; 00164 int i, j; 00165 RA288Context *ractx = avctx->priv_data; 00166 GetBitContext gb; 00167 00168 if (buf_size < avctx->block_align) { 00169 av_log(avctx, AV_LOG_ERROR, 00170 "Error! Input buffer is too small [%d<%d]\n", 00171 buf_size, avctx->block_align); 00172 return 0; 00173 } 00174 00175 if (*data_size < 32*5*4) 00176 return -1; 00177 00178 init_get_bits(&gb, buf, avctx->block_align * 8); 00179 00180 for (i=0; i < 32; i++) { 00181 float gain = amptable[get_bits(&gb, 3)]; 00182 int cb_coef = get_bits(&gb, 6 + (i&1)); 00183 00184 decode(ractx, gain, cb_coef); 00185 00186 for (j=0; j < 5; j++) 00187 *(out++) = ractx->sp_hist[70 + 36 + j]; 00188 00189 if ((i & 7) == 3) { 00190 backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window, 00191 ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70); 00192 00193 backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window, 00194 ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28); 00195 } 00196 } 00197 00198 *data_size = (char *)out - (char *)data; 00199 return avctx->block_align; 00200 } 00201 00202 AVCodec ra_288_decoder = 00203 { 00204 "real_288", 00205 AVMEDIA_TYPE_AUDIO, 00206 CODEC_ID_RA_288, 00207 sizeof(RA288Context), 00208 ra288_decode_init, 00209 NULL, 00210 NULL, 00211 ra288_decode_frame, 00212 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"), 00213 };