Mercurial > libavcodec.hg
annotate ra288.c @ 10893:2aafcafbe1f0 libavcodec
Replace cabac checks in inline functions from h264.h with constants.
No benchmark because its just replacing variables with litteral constants
(so no risk for slowdown outside gcc silliness) and i need sleep.
author | michael |
---|---|
date | Sat, 16 Jan 2010 05:41:33 +0000 |
parents | 0dce4fe6e6f3 |
children | 8a4984c5cacc |
rev | line source |
---|---|
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 * |
3947
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diego
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5 * This file is part of FFmpeg. |
c8c591fe26f8
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6 * |
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7 * FFmpeg is free software; you can redistribute it and/or |
1304
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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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diego
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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.
diego
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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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20 */ |
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21 |
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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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22 #include "avcodec.h" |
7171 | 23 #define ALT_BITSTREAM_READER_LE |
9428 | 24 #include "get_bits.h" |
1335
b4a72edb3a71
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?
al3x
parents:
1316
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25 #include "ra288.h" |
7788
ffd4b1364b62
Avoid duplicating compute_lpc_coefs() function in both the RA288 and AAC decoders.
vitor
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26 #include "lpc.h" |
8094 | 27 #include "celp_math.h" |
9026 | 28 #include "celp_filters.h" |
2967 | 29 |
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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30 typedef struct { |
7439 | 31 float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A) |
7849 | 32 float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB) |
1304
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33 |
7853
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34 /** speech data history (spec: SB). |
21ab545aed8b
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35 * Its first 70 coefficients are updated only at backward filtering. |
21ab545aed8b
Simplify: use a single history buffer for gain and a single one for
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36 */ |
21ab545aed8b
Simplify: use a single history buffer for gain and a single one for
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37 float sp_hist[111]; |
7418
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Rename st{1,2}{a,b} context variables to more meaningful names: {sp,gain}_{rec,hist}
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38 |
7849 | 39 /// speech part of the gain autocorrelation (spec: REXP) |
7418
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40 float sp_rec[37]; |
83d005a098b5
Rename st{1,2}{a,b} context variables to more meaningful names: {sp,gain}_{rec,hist}
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41 |
7853
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42 /** log-gain history (spec: SBLG). |
21ab545aed8b
Simplify: use a single history buffer for gain and a single one for
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43 * Its first 28 coefficients are updated only at backward filtering. |
21ab545aed8b
Simplify: use a single history buffer for gain and a single one for
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44 */ |
21ab545aed8b
Simplify: use a single history buffer for gain and a single one for
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45 float gain_hist[38]; |
7418
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Rename st{1,2}{a,b} context variables to more meaningful names: {sp,gain}_{rec,hist}
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46 |
7849 | 47 /// recursive part of the gain autocorrelation (spec: REXPLG) |
7418
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48 float gain_rec[11]; |
7411
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Give the context structure of ra288 a better name: s/Real288_internal/RA288Context/
vitor
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49 } 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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50 |
7451
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51 static av_cold int ra288_decode_init(AVCodecContext *avctx) |
85ab7655ad4d
Modify all codecs to report their supported input and output sample format(s).
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52 { |
7874
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Use SAMPLE_FMT_FLT instead of doing the float->int conversion in the
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53 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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54 return 0; |
85ab7655ad4d
Modify all codecs to report their supported input and output sample format(s).
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55 } |
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Modify all codecs to report their supported input and output sample format(s).
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56 |
7848 | 57 static void apply_window(float *tgt, const float *m1, const float *m2, int n) |
7415
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Move colmult() function to the beginning of file to group DSP-related functions.
vitor
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58 { |
649496faa1d2
Move colmult() function to the beginning of file to group DSP-related functions.
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59 while (n--) |
7416 | 60 *tgt++ = *m1++ * *m2++; |
7415
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Move colmult() function to the beginning of file to group DSP-related functions.
vitor
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61 } |
649496faa1d2
Move colmult() function to the beginning of file to group DSP-related functions.
vitor
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62 |
7916
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Cosmetics: move convolve() together with the other DSP functions
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63 static void convolve(float *tgt, const float *src, int len, int n) |
2a4c42e6aecb
Cosmetics: move convolve() together with the other DSP functions
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64 { |
2a4c42e6aecb
Cosmetics: move convolve() together with the other DSP functions
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65 for (; n >= 0; n--) |
8094 | 66 tgt[n] = ff_dot_productf(src, src - n, len); |
7916
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67 |
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68 } |
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Cosmetics: move convolve() together with the other DSP functions
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69 |
7412
51da3d8200d0
Give the context variables of ra288 a better name: s/glob/ractx/
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70 static void decode(RA288Context *ractx, float gain, int cb_coef) |
1304
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71 { |
9026 | 72 int i; |
7268
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Do not declare as double a var that only stores a float
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73 double sumsum; |
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Do not declare as double a var that only stores a float
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74 float sum, buffer[5]; |
7853
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75 float *block = ractx->sp_hist + 70 + 36; // current block |
21ab545aed8b
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76 float *gain_block = ractx->gain_hist + 28; |
7167 | 77 |
7853
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78 memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block)); |
7167 | 79 |
7431
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Add a few comments pointing to the G.728 specification
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80 /* block 46 of G.728 spec */ |
7503
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vitor
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81 sum = 32.; |
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vitor
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82 for (i=0; i < 10; i++) |
7853
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83 sum -= gain_block[9-i] * ractx->gain_lpc[i]; |
7167 | 84 |
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Add a few comments pointing to the G.728 specification
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85 /* block 47 of G.728 spec */ |
7269 | 86 sum = av_clipf(sum, 0, 60); |
1304
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87 |
7431
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Add a few comments pointing to the G.728 specification
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88 /* block 48 of G.728 spec */ |
7915 | 89 /* 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
vitor
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7917
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90 sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23)); |
7167 | 91 |
7448 | 92 for (i=0; i < 5; i++) |
7915 | 93 buffer[i] = codetable[cb_coef][i] * sumsum; |
1304
e8543aab0cc9
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94 |
8094 | 95 sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.); |
7275 | 96 |
97 sum = FFMAX(sum, 1); | |
7167 | 98 |
99 /* shift and store */ | |
7853
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100 memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block)); |
7167 | 101 |
7853
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102 gain_block[9] = 10 * log10(sum) - 32; |
1304
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103 |
9026 | 104 ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36); |
1304
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105 |
7167 | 106 /* output */ |
7448 | 107 for (i=0; i < 5; i++) |
7917 | 108 block[i] = av_clipf(block[i], -4095./4096., 4095./4096.); |
1304
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109 } |
e8543aab0cc9
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110 |
7367 | 111 /** |
7847 | 112 * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification. |
7367 | 113 * |
7847 | 114 * @param order filter order |
115 * @param n input length | |
116 * @param non_rec number of non-recursive samples | |
117 * @param out filter output | |
7853
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118 * @param hist pointer to the input history of the filter |
7375 | 119 * @param out pointer to the non-recursive part of the output |
7367 | 120 * @param out2 pointer to the recursive part of the output |
121 * @param window pointer to the windowing function table | |
122 */ | |
7854 | 123 static void do_hybrid_window(int order, int n, int non_rec, float *out, |
124 float *hist, float *out2, const float *window) | |
1304
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125 { |
7449
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126 int i; |
7437
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Declare temporary buffers to be only of the necessary size
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127 float buffer1[order + 1]; |
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128 float buffer2[order + 1]; |
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129 float work[order + n + non_rec]; |
1304
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130 |
7848 | 131 apply_window(work, window, hist, order + n + non_rec); |
2967 | 132 |
7492 | 133 convolve(buffer1, work + order , n , order); |
134 convolve(buffer2, work + order + n, non_rec, order); | |
1304
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135 |
7448 | 136 for (i=0; i <= order; i++) { |
137 out2[i] = out2[i] * 0.5625 + buffer1[i]; | |
138 out [i] = out2[i] + buffer2[i]; | |
7167 | 139 } |
7365
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Rename co() function to the much more descriptive name of
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140 |
7847 | 141 /* 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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142 *out *= 257./256.; |
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143 } |
e8543aab0cc9
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144 |
7372
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Rename update() function to backward_filter() and add a doxy comment
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145 /** |
7847 | 146 * Backward synthesis filter, find the LPC coefficients from past speech data. |
7372
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Rename update() function to backward_filter() and add a doxy comment
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147 */ |
7929 | 148 static void backward_filter(float *hist, float *rec, const float *window, |
149 float *lpc, const float *tab, | |
150 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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diff
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151 { |
7929 | 152 float temp[order+1]; |
7169 | 153 |
7929 | 154 do_hybrid_window(order, n, non_rec, temp, hist, rec, window); |
7853
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155 |
7929 | 156 if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1)) |
157 apply_window(lpc, lpc, tab, order); | |
7853
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158 |
7929 | 159 memmove(hist, hist + n, move_size*sizeof(*hist)); |
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160 } |
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161 |
7174 | 162 static int ra288_decode_frame(AVCodecContext * avctx, void *data, |
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163 int *data_size, AVPacket *avpkt) |
1313 | 164 { |
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165 const uint8_t *buf = avpkt->data; |
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166 int buf_size = avpkt->size; |
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167 float *out = data; |
7448 | 168 int i, j; |
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169 RA288Context *ractx = avctx->priv_data; |
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170 GetBitContext gb; |
7167 | 171 |
7174 | 172 if (buf_size < avctx->block_align) { |
173 av_log(avctx, AV_LOG_ERROR, | |
174 "Error! Input buffer is too small [%d<%d]\n", | |
175 buf_size, avctx->block_align); | |
176 return 0; | |
177 } | |
178 | |
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179 if (*data_size < 32*5*4) |
7505 | 180 return -1; |
181 | |
7174 | 182 init_get_bits(&gb, buf, avctx->block_align * 8); |
1313 | 183 |
7448 | 184 for (i=0; i < 32; i++) { |
7192 | 185 float gain = amptable[get_bits(&gb, 3)]; |
7448 | 186 int cb_coef = get_bits(&gb, 6 + (i&1)); |
7443 | 187 |
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188 decode(ractx, gain, cb_coef); |
7167 | 189 |
7448 | 190 for (j=0; j < 5; j++) |
7917 | 191 *(out++) = ractx->sp_hist[70 + 36 + j]; |
7167 | 192 |
7929 | 193 if ((i & 7) == 3) { |
194 backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window, | |
195 ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70); | |
196 | |
197 backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window, | |
198 ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28); | |
199 } | |
7167 | 200 } |
201 | |
7174 | 202 *data_size = (char *)out - (char *)data; |
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203 return avctx->block_align; |
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204 } |
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205 |
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206 AVCodec ra_288_decoder = |
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207 { |
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208 "real_288", |
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209 CODEC_TYPE_AUDIO, |
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210 CODEC_ID_RA_288, |
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211 sizeof(RA288Context), |
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212 ra288_decode_init, |
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213 NULL, |
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214 NULL, |
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215 ra288_decode_frame, |
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216 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"), |
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217 }; |