Libav 0.7.1
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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 Libav. 00006 * 00007 * Libav 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 * Libav 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 Libav; 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 #define MAX_BACKWARD_FILTER_ORDER 36 00031 #define MAX_BACKWARD_FILTER_LEN 40 00032 #define MAX_BACKWARD_FILTER_NONREC 35 00033 00034 typedef struct { 00035 float sp_lpc[36]; 00036 float gain_lpc[10]; 00037 00041 float sp_hist[111]; 00042 00044 float sp_rec[37]; 00045 00049 float gain_hist[38]; 00050 00052 float gain_rec[11]; 00053 } RA288Context; 00054 00055 static av_cold int ra288_decode_init(AVCodecContext *avctx) 00056 { 00057 avctx->sample_fmt = AV_SAMPLE_FMT_FLT; 00058 return 0; 00059 } 00060 00061 static void apply_window(float *tgt, const float *m1, const float *m2, int n) 00062 { 00063 while (n--) 00064 *tgt++ = *m1++ * *m2++; 00065 } 00066 00067 static void convolve(float *tgt, const float *src, int len, int n) 00068 { 00069 for (; n >= 0; n--) 00070 tgt[n] = ff_dot_productf(src, src - n, len); 00071 00072 } 00073 00074 static void decode(RA288Context *ractx, float gain, int cb_coef) 00075 { 00076 int i; 00077 double sumsum; 00078 float sum, buffer[5]; 00079 float *block = ractx->sp_hist + 70 + 36; // current block 00080 float *gain_block = ractx->gain_hist + 28; 00081 00082 memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block)); 00083 00084 /* block 46 of G.728 spec */ 00085 sum = 32.; 00086 for (i=0; i < 10; i++) 00087 sum -= gain_block[9-i] * ractx->gain_lpc[i]; 00088 00089 /* block 47 of G.728 spec */ 00090 sum = av_clipf(sum, 0, 60); 00091 00092 /* block 48 of G.728 spec */ 00093 /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */ 00094 sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23)); 00095 00096 for (i=0; i < 5; i++) 00097 buffer[i] = codetable[cb_coef][i] * sumsum; 00098 00099 sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.); 00100 00101 sum = FFMAX(sum, 1); 00102 00103 /* shift and store */ 00104 memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block)); 00105 00106 gain_block[9] = 10 * log10(sum) - 32; 00107 00108 ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36); 00109 } 00110 00123 static void do_hybrid_window(int order, int n, int non_rec, float *out, 00124 float *hist, float *out2, const float *window) 00125 { 00126 int i; 00127 float buffer1[MAX_BACKWARD_FILTER_ORDER + 1]; 00128 float buffer2[MAX_BACKWARD_FILTER_ORDER + 1]; 00129 float work[MAX_BACKWARD_FILTER_ORDER + MAX_BACKWARD_FILTER_LEN + MAX_BACKWARD_FILTER_NONREC]; 00130 00131 apply_window(work, window, hist, order + n + non_rec); 00132 00133 convolve(buffer1, work + order , n , order); 00134 convolve(buffer2, work + order + n, non_rec, order); 00135 00136 for (i=0; i <= order; i++) { 00137 out2[i] = out2[i] * 0.5625 + buffer1[i]; 00138 out [i] = out2[i] + buffer2[i]; 00139 } 00140 00141 /* Multiply by the white noise correcting factor (WNCF). */ 00142 *out *= 257./256.; 00143 } 00144 00148 static void backward_filter(float *hist, float *rec, const float *window, 00149 float *lpc, const float *tab, 00150 int order, int n, int non_rec, int move_size) 00151 { 00152 float temp[MAX_BACKWARD_FILTER_ORDER+1]; 00153 00154 do_hybrid_window(order, n, non_rec, temp, hist, rec, window); 00155 00156 if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1)) 00157 apply_window(lpc, lpc, tab, order); 00158 00159 memmove(hist, hist + n, move_size*sizeof(*hist)); 00160 } 00161 00162 static int ra288_decode_frame(AVCodecContext * avctx, void *data, 00163 int *data_size, AVPacket *avpkt) 00164 { 00165 const uint8_t *buf = avpkt->data; 00166 int buf_size = avpkt->size; 00167 float *out = data; 00168 int i, j; 00169 RA288Context *ractx = avctx->priv_data; 00170 GetBitContext gb; 00171 00172 if (buf_size < avctx->block_align) { 00173 av_log(avctx, AV_LOG_ERROR, 00174 "Error! Input buffer is too small [%d<%d]\n", 00175 buf_size, avctx->block_align); 00176 return 0; 00177 } 00178 00179 if (*data_size < 32*5*4) 00180 return -1; 00181 00182 init_get_bits(&gb, buf, avctx->block_align * 8); 00183 00184 for (i=0; i < 32; i++) { 00185 float gain = amptable[get_bits(&gb, 3)]; 00186 int cb_coef = get_bits(&gb, 6 + (i&1)); 00187 00188 decode(ractx, gain, cb_coef); 00189 00190 for (j=0; j < 5; j++) 00191 *(out++) = ractx->sp_hist[70 + 36 + j]; 00192 00193 if ((i & 7) == 3) { 00194 backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window, 00195 ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70); 00196 00197 backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window, 00198 ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28); 00199 } 00200 } 00201 00202 *data_size = (char *)out - (char *)data; 00203 return avctx->block_align; 00204 } 00205 00206 AVCodec ff_ra_288_decoder = 00207 { 00208 "real_288", 00209 AVMEDIA_TYPE_AUDIO, 00210 CODEC_ID_RA_288, 00211 sizeof(RA288Context), 00212 ra288_decode_init, 00213 NULL, 00214 NULL, 00215 ra288_decode_frame, 00216 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"), 00217 };