Mercurial > libavcodec.hg
annotate ra288.c @ 8917:422a9dbebb53 libavcodec
Almost cosmetics.
Move some assignments after the checks.
author | iive |
---|---|
date | Sun, 15 Feb 2009 10:14:16 +0000 |
parents | 36671e6689c7 |
children | 19bbd1457372 |
rev | line source |
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1304
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1 /* |
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2 * RealAudio 2.0 (28.8K) |
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3 * Copyright (c) 2003 the ffmpeg project |
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4 * |
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5 * This file is part of FFmpeg. |
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6 * |
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7 * FFmpeg is free software; you can redistribute it and/or |
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8 * modify it under the terms of the GNU Lesser General Public |
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9 * License as published by the Free Software Foundation; either |
3947
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10 * version 2.1 of the License, or (at your option) any later version. |
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11 * |
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12 * FFmpeg is distributed in the hope that it will be useful, |
1304
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13 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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15 * Lesser General Public License for more details. |
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16 * |
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17 * You should have received a copy of the GNU Lesser General Public |
3947
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18 * License along with FFmpeg; if not, write to the Free Software |
3036
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Update licensing information: The FSF changed postal address.
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
1304
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20 */ |
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21 |
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22 #include "avcodec.h" |
7171 | 23 #define ALT_BITSTREAM_READER_LE |
24 #include "bitstream.h" | |
1335
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moved the tables into header files (and applied the 'static' patch). Nick: why do you like mergeing tables and code into one file, so making it unusable big?
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25 #include "ra288.h" |
7788
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Avoid duplicating compute_lpc_coefs() function in both the RA288 and AAC decoders.
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26 #include "lpc.h" |
8094 | 27 #include "celp_math.h" |
2967 | 28 |
1304
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29 typedef struct { |
7439 | 30 float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A) |
7849 | 31 float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB) |
1304
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32 |
7853
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33 /** speech data history (spec: SB). |
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34 * Its first 70 coefficients are updated only at backward filtering. |
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35 */ |
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36 float sp_hist[111]; |
7418
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37 |
7849 | 38 /// speech part of the gain autocorrelation (spec: REXP) |
7418
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39 float sp_rec[37]; |
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40 |
7853
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41 /** log-gain history (spec: SBLG). |
21ab545aed8b
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42 * Its first 28 coefficients are updated only at backward filtering. |
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43 */ |
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44 float gain_hist[38]; |
7418
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45 |
7849 | 46 /// recursive part of the gain autocorrelation (spec: REXPLG) |
7418
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47 float gain_rec[11]; |
7411
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Give the context structure of ra288 a better name: s/Real288_internal/RA288Context/
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48 } RA288Context; |
1304
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RealAudio 1.0 (14_4) and 2.0 (28_8) native decoders. Author unknown, code from a mplayerhq (originally from public domain player for Amiga
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49 |
7451
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Modify all codecs to report their supported input and output sample format(s).
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50 static av_cold int ra288_decode_init(AVCodecContext *avctx) |
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51 { |
7874
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Use SAMPLE_FMT_FLT instead of doing the float->int conversion in the
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52 avctx->sample_fmt = SAMPLE_FMT_FLT; |
7451
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Modify all codecs to report their supported input and output sample format(s).
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53 return 0; |
85ab7655ad4d
Modify all codecs to report their supported input and output sample format(s).
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54 } |
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Modify all codecs to report their supported input and output sample format(s).
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55 |
7848 | 56 static void apply_window(float *tgt, const float *m1, const float *m2, int n) |
7415
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57 { |
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58 while (n--) |
7416 | 59 *tgt++ = *m1++ * *m2++; |
7415
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Move colmult() function to the beginning of file to group DSP-related functions.
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60 } |
649496faa1d2
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61 |
7916
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Cosmetics: move convolve() together with the other DSP functions
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62 static void convolve(float *tgt, const float *src, int len, int n) |
2a4c42e6aecb
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63 { |
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64 for (; n >= 0; n--) |
8094 | 65 tgt[n] = ff_dot_productf(src, src - n, len); |
7916
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66 |
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67 } |
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68 |
7412
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Give the context variables of ra288 a better name: s/glob/ractx/
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69 static void decode(RA288Context *ractx, float gain, int cb_coef) |
1304
e8543aab0cc9
RealAudio 1.0 (14_4) and 2.0 (28_8) native decoders. Author unknown, code from a mplayerhq (originally from public domain player for Amiga
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70 { |
7448 | 71 int i, j; |
7268
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Do not declare as double a var that only stores a float
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72 double sumsum; |
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73 float sum, buffer[5]; |
7853
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74 float *block = ractx->sp_hist + 70 + 36; // current block |
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75 float *gain_block = ractx->gain_hist + 28; |
7167 | 76 |
7853
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77 memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block)); |
7167 | 78 |
7431
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79 /* block 46 of G.728 spec */ |
7503
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80 sum = 32.; |
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81 for (i=0; i < 10; i++) |
7853
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82 sum -= gain_block[9-i] * ractx->gain_lpc[i]; |
7167 | 83 |
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84 /* block 47 of G.728 spec */ |
7269 | 85 sum = av_clipf(sum, 0, 60); |
1304
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86 |
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87 /* block 48 of G.728 spec */ |
7915 | 88 /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */ |
7925
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Do instead of divising a float by a constant, multiply by its inverse
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89 sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23)); |
7167 | 90 |
7448 | 91 for (i=0; i < 5; i++) |
7915 | 92 buffer[i] = codetable[cb_coef][i] * sumsum; |
1304
e8543aab0cc9
RealAudio 1.0 (14_4) and 2.0 (28_8) native decoders. Author unknown, code from a mplayerhq (originally from public domain player for Amiga
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93 |
8094 | 94 sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.); |
7275 | 95 |
96 sum = FFMAX(sum, 1); | |
7167 | 97 |
98 /* shift and store */ | |
7853
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99 memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block)); |
7167 | 100 |
7853
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101 gain_block[9] = 10 * log10(sum) - 32; |
1304
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102 |
7912 | 103 for (i=0; i < 5; i++) { |
7913 | 104 block[i] = buffer[i]; |
7912 | 105 for (j=0; j < 36; j++) |
106 block[i] -= block[i-1-j]*ractx->sp_lpc[j]; | |
107 } | |
1304
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108 |
7167 | 109 /* output */ |
7448 | 110 for (i=0; i < 5; i++) |
7917 | 111 block[i] = av_clipf(block[i], -4095./4096., 4095./4096.); |
1304
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RealAudio 1.0 (14_4) and 2.0 (28_8) native decoders. Author unknown, code from a mplayerhq (originally from public domain player for Amiga
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112 } |
e8543aab0cc9
RealAudio 1.0 (14_4) and 2.0 (28_8) native decoders. Author unknown, code from a mplayerhq (originally from public domain player for Amiga
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113 |
7367 | 114 /** |
7847 | 115 * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification. |
7367 | 116 * |
7847 | 117 * @param order filter order |
118 * @param n input length | |
119 * @param non_rec number of non-recursive samples | |
120 * @param out filter output | |
7853
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121 * @param hist pointer to the input history of the filter |
7375 | 122 * @param out pointer to the non-recursive part of the output |
7367 | 123 * @param out2 pointer to the recursive part of the output |
124 * @param window pointer to the windowing function table | |
125 */ | |
7854 | 126 static void do_hybrid_window(int order, int n, int non_rec, float *out, |
127 float *hist, float *out2, const float *window) | |
1304
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128 { |
7449
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129 int i; |
7437
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Declare temporary buffers to be only of the necessary size
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130 float buffer1[order + 1]; |
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131 float buffer2[order + 1]; |
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132 float work[order + n + non_rec]; |
1304
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133 |
7848 | 134 apply_window(work, window, hist, order + n + non_rec); |
2967 | 135 |
7492 | 136 convolve(buffer1, work + order , n , order); |
137 convolve(buffer2, work + order + n, non_rec, order); | |
1304
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138 |
7448 | 139 for (i=0; i <= order; i++) { |
140 out2[i] = out2[i] * 0.5625 + buffer1[i]; | |
141 out [i] = out2[i] + buffer2[i]; | |
7167 | 142 } |
7365
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Rename co() function to the much more descriptive name of
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143 |
7847 | 144 /* Multiply by the white noise correcting factor (WNCF). */ |
7365
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Rename co() function to the much more descriptive name of
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145 *out *= 257./256.; |
1304
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146 } |
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147 |
7372
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Rename update() function to backward_filter() and add a doxy comment
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148 /** |
7847 | 149 * Backward synthesis filter, find the LPC coefficients from past speech data. |
7372
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150 */ |
7929 | 151 static void backward_filter(float *hist, float *rec, const float *window, |
152 float *lpc, const float *tab, | |
153 int order, int n, int non_rec, int move_size) | |
1304
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RealAudio 1.0 (14_4) and 2.0 (28_8) native decoders. Author unknown, code from a mplayerhq (originally from public domain player for Amiga
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154 { |
7929 | 155 float temp[order+1]; |
7169 | 156 |
7929 | 157 do_hybrid_window(order, n, non_rec, temp, hist, rec, window); |
7853
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158 |
7929 | 159 if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1)) |
160 apply_window(lpc, lpc, tab, order); | |
7853
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161 |
7929 | 162 memmove(hist, hist + n, move_size*sizeof(*hist)); |
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163 } |
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164 |
7174 | 165 static int ra288_decode_frame(AVCodecContext * avctx, void *data, |
166 int *data_size, const uint8_t * buf, | |
167 int buf_size) | |
1313 | 168 { |
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169 float *out = data; |
7448 | 170 int i, j; |
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171 RA288Context *ractx = avctx->priv_data; |
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172 GetBitContext gb; |
7167 | 173 |
7174 | 174 if (buf_size < avctx->block_align) { |
175 av_log(avctx, AV_LOG_ERROR, | |
176 "Error! Input buffer is too small [%d<%d]\n", | |
177 buf_size, avctx->block_align); | |
178 return 0; | |
179 } | |
180 | |
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181 if (*data_size < 32*5*4) |
7505 | 182 return -1; |
183 | |
7174 | 184 init_get_bits(&gb, buf, avctx->block_align * 8); |
1313 | 185 |
7448 | 186 for (i=0; i < 32; i++) { |
7192 | 187 float gain = amptable[get_bits(&gb, 3)]; |
7448 | 188 int cb_coef = get_bits(&gb, 6 + (i&1)); |
7443 | 189 |
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190 decode(ractx, gain, cb_coef); |
7167 | 191 |
7448 | 192 for (j=0; j < 5; j++) |
7917 | 193 *(out++) = ractx->sp_hist[70 + 36 + j]; |
7167 | 194 |
7929 | 195 if ((i & 7) == 3) { |
196 backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window, | |
197 ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70); | |
198 | |
199 backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window, | |
200 ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28); | |
201 } | |
7167 | 202 } |
203 | |
7174 | 204 *data_size = (char *)out - (char *)data; |
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205 return avctx->block_align; |
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206 } |
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207 |
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208 AVCodec ra_288_decoder = |
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209 { |
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210 "real_288", |
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211 CODEC_TYPE_AUDIO, |
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212 CODEC_ID_RA_288, |
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213 sizeof(RA288Context), |
7451
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214 ra288_decode_init, |
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215 NULL, |
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216 NULL, |
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217 ra288_decode_frame, |
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218 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"), |
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219 }; |