Mercurial > libavcodec.hg
annotate ra144.c @ 8831:27616b326d57 libavcodec
Rename xvmc.c --> mpegvideo_xvmc.c as suggested by Ivan.
author | diego |
---|---|
date | Sat, 14 Feb 2009 15:30:40 +0000 |
parents | 611a21e4b01b |
children | ad7fd7a40717 |
rev | line source |
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1304
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1 /* |
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2 * Real Audio 1.0 (14.4K) |
7163 | 3 * |
4 * Copyright (c) 2008 Vitor Sessak | |
5 * Copyright (c) 2003 Nick Kurshev | |
6 * Based on public domain decoder at http://www.honeypot.net/audio | |
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7 * |
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8 * This file is part of FFmpeg. |
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Change license headers to say 'FFmpeg' instead of 'this program/this library'
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9 * |
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10 * FFmpeg is free software; you can redistribute it and/or |
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11 * modify it under the terms of the GNU Lesser General Public |
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12 * License as published by the Free Software Foundation; either |
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13 * version 2.1 of the License, or (at your option) any later version. |
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14 * |
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15 * FFmpeg is distributed in the hope that it will be useful, |
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16 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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18 * Lesser General Public License for more details. |
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19 * |
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20 * You should have received a copy of the GNU Lesser General Public |
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21 * 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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22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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23 */ |
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24 |
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25 #include "avcodec.h" |
6768
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Rewrite unpack_input() completely, patch by Vitor Sessak, vitor1001 gmail com.
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26 #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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27 #include "ra144.h" |
8049
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Split off celp_filters.[ch] from acelp_filters.[ch] for the QCELP decoder.
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28 #include "celp_filters.h" |
1304
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29 |
7107 | 30 #define NBLOCKS 4 ///< number of subblocks within a block |
31 #define BLOCKSIZE 40 ///< subblock size in 16-bit words | |
32 #define BUFFERSIZE 146 ///< the size of the adaptive codebook | |
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33 |
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34 |
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35 typedef struct { |
6910 | 36 unsigned int old_energy; ///< previous frame energy |
1304
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37 |
7005 | 38 unsigned int lpc_tables[2][10]; |
7128
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39 |
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40 /** LPC coefficients: lpc_coef[0] is the coefficients of the current frame |
7847 | 41 * and lpc_coef[1] of the previous one. */ |
7128
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42 unsigned int *lpc_coef[2]; |
614bc53e67a8
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43 |
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44 unsigned int lpc_refl_rms[2]; |
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45 |
7847 | 46 /** The current subblock padded by the last 10 values of the previous one. */ |
7096
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Avoid a memcpy. Code is also simpler now in my opinion.
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47 int16_t curr_sblock[50]; |
1ad555e54621
Avoid a memcpy. Code is also simpler now in my opinion.
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48 |
7847 | 49 /** Adaptive codebook, its size is two units bigger to avoid a |
50 * buffer overflow. */ | |
7827
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51 uint16_t adapt_cb[146+2]; |
6894
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Rename context struct to be more consistent with the rest of ffmpeg
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52 } RA144Context; |
1304
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53 |
7828 | 54 static av_cold int ra144_decode_init(AVCodecContext * avctx) |
1304
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55 { |
6895
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56 RA144Context *ractx = avctx->priv_data; |
1304
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57 |
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58 ractx->lpc_coef[0] = ractx->lpc_tables[0]; |
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59 ractx->lpc_coef[1] = ractx->lpc_tables[1]; |
1304
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60 |
7451
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Modify all codecs to report their supported input and output sample format(s).
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61 avctx->sample_fmt = SAMPLE_FMT_S16; |
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62 return 0; |
1304
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63 } |
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64 |
6899 | 65 /** |
6902 | 66 * Evaluate sqrt(x << 24). x must fit in 20 bits. This value is evaluated in an |
6899 | 67 * odd way to make the output identical to the binary decoder. |
68 */ | |
1304
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69 static int t_sqrt(unsigned int x) |
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70 { |
7104 | 71 int s = 2; |
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72 while (x > 0xfff) { |
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73 s++; |
7507 | 74 x >>= 2; |
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75 } |
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76 |
7104 | 77 return ff_sqrt(x << 20) << s; |
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78 } |
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79 |
6952 | 80 /** |
81 * Evaluate the LPC filter coefficients from the reflection coefficients. | |
6965 | 82 * Does the inverse of the eval_refl() function. |
6952 | 83 */ |
7120
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84 static void eval_coefs(int *coefs, const int *refl) |
1304
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85 { |
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86 int buffer[10]; |
6782 | 87 int *b1 = buffer; |
6964 | 88 int *b2 = coefs; |
7447 | 89 int i, j; |
2967 | 90 |
7447 | 91 for (i=0; i < 10; i++) { |
92 b1[i] = refl[i] << 4; | |
6782 | 93 |
7447 | 94 for (j=0; j < i; j++) |
95 b1[j] = ((refl[i] * b2[i-j-1]) >> 12) + b2[j]; | |
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96 |
6781 | 97 FFSWAP(int *, b1, b2); |
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98 } |
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99 |
7447 | 100 for (i=0; i < 10; i++) |
101 coefs[i] >>= 4; | |
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102 } |
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103 |
7105
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Rename rotate_buffer() to copy_and_dup() and doxy it
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104 /** |
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105 * Copy the last offset values of *source to *target. If those values are not |
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106 * enough to fill the target buffer, fill it with another copy of those values. |
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107 */ |
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108 static void copy_and_dup(int16_t *target, const int16_t *source, int offset) |
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109 { |
6831 | 110 source += BUFFERSIZE - offset; |
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111 |
7829 | 112 memcpy(target, source, FFMIN(BLOCKSIZE, offset)*sizeof(*target)); |
113 if (offset < BLOCKSIZE) | |
7100 | 114 memcpy(target + offset, source, (BLOCKSIZE - offset)*sizeof(*target)); |
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115 } |
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116 |
7110 | 117 /** inverse root mean square */ |
7112 | 118 static int irms(const int16_t *data) |
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119 { |
6838 | 120 unsigned int i, sum = 0; |
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121 |
6838 | 122 for (i=0; i < BLOCKSIZE; i++) |
123 sum += data[i] * data[i]; | |
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124 |
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125 if (sum == 0) |
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126 return 0; /* OOPS - division by zero */ |
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127 |
7112 | 128 return 0x20000000 / (t_sqrt(sum) >> 8); |
1304
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129 } |
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130 |
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131 static void add_wav(int16_t *dest, int n, int skip_first, int *m, |
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132 const int16_t *s1, const int8_t *s2, const int8_t *s3) |
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133 { |
6873
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Make add_wav() receive a vector instead of three integers
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134 int i; |
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135 int v[3]; |
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136 |
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137 v[0] = 0; |
6880 | 138 for (i=!skip_first; i<3; i++) |
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Normalize triplets in gain_val_tab[][] so gain_exp_tab[] can be just an
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139 v[i] = (gain_val_tab[n][i] * m[i]) >> gain_exp_tab[n]; |
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140 |
6839 | 141 for (i=0; i < BLOCKSIZE; i++) |
7117 | 142 dest[i] = (s1[i]*v[0] + s2[i]*v[1] + s3[i]*v[2]) >> 12; |
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143 } |
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144 |
7113 | 145 static unsigned int rescale_rms(unsigned int rms, unsigned int energy) |
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146 { |
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147 return (rms * energy) >> 10; |
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148 } |
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149 |
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150 static unsigned int rms(const int *data) |
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151 { |
7447 | 152 int i; |
6850 | 153 unsigned int res = 0x10000; |
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154 int b = 10; |
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155 |
7447 | 156 for (i=0; i < 10; i++) { |
157 res = (((0x1000000 - data[i]*data[i]) >> 12) * res) >> 12; | |
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158 |
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159 if (res == 0) |
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160 return 0; |
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161 |
6850 | 162 while (res <= 0x3fff) { |
163 b++; | |
164 res <<= 2; | |
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165 } |
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166 } |
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167 |
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168 return t_sqrt(res) >> b; |
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169 } |
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170 |
7133 | 171 static void do_output_subblock(RA144Context *ractx, const uint16_t *lpc_coefs, |
172 int gval, GetBitContext *gb) | |
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173 { |
6887 | 174 uint16_t buffer_a[40]; |
175 uint16_t *block; | |
6907 | 176 int cba_idx = get_bits(gb, 7); // index of the adaptive CB, 0 if none |
6900 | 177 int gain = get_bits(gb, 8); |
178 int cb1_idx = get_bits(gb, 7); | |
179 int cb2_idx = get_bits(gb, 7); | |
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180 int m[3]; |
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181 |
6900 | 182 if (cba_idx) { |
7106 | 183 cba_idx += BLOCKSIZE/2 - 1; |
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184 copy_and_dup(buffer_a, ractx->adapt_cb, cba_idx); |
7112 | 185 m[0] = (irms(buffer_a) * gval) >> 12; |
6888 | 186 } else { |
187 m[0] = 0; | |
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188 } |
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189 |
7121 | 190 m[1] = (cb1_base[cb1_idx] * gval) >> 8; |
191 m[2] = (cb2_base[cb2_idx] * gval) >> 8; | |
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192 |
6896 | 193 memmove(ractx->adapt_cb, ractx->adapt_cb + BLOCKSIZE, |
7119 | 194 (BUFFERSIZE - BLOCKSIZE) * sizeof(*ractx->adapt_cb)); |
6882 | 195 |
6896 | 196 block = ractx->adapt_cb + BUFFERSIZE - BLOCKSIZE; |
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197 |
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198 add_wav(block, gain, cba_idx, m, buffer_a, |
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199 cb1_vects[cb1_idx], cb2_vects[cb2_idx]); |
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200 |
7101
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201 memcpy(ractx->curr_sblock, ractx->curr_sblock + 40, |
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202 10*sizeof(*ractx->curr_sblock)); |
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203 |
8049
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204 if (ff_celp_lp_synthesis_filter(ractx->curr_sblock + 10, lpc_coefs, |
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205 block, BLOCKSIZE, 10, 1, 0xfff)) |
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206 memset(ractx->curr_sblock, 0, 50*sizeof(*ractx->curr_sblock)); |
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207 } |
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208 |
7077 | 209 static void int_to_int16(int16_t *out, const int *inp) |
6934 | 210 { |
211 int i; | |
212 | |
7447 | 213 for (i=0; i < 30; i++) |
7800 | 214 *out++ = *inp++; |
6934 | 215 } |
216 | |
6952 | 217 /** |
218 * Evaluate the reflection coefficients from the filter coefficients. | |
6965 | 219 * Does the inverse of the eval_coefs() function. |
6952 | 220 * |
7847 | 221 * @return 1 if one of the reflection coefficients is greater than |
6952 | 222 * 4095, 0 if not. |
223 */ | |
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224 static int eval_refl(int *refl, const int16_t *coefs, RA144Context *ractx) |
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225 { |
7516 | 226 int b, i, j; |
6806 | 227 int buffer1[10]; |
228 int buffer2[10]; | |
6852 | 229 int *bp1 = buffer1; |
230 int *bp2 = buffer2; | |
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231 |
6852 | 232 for (i=0; i < 10; i++) |
6964 | 233 buffer2[i] = coefs[i]; |
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234 |
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235 refl[9] = bp2[9]; |
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236 |
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237 if ((unsigned) bp2[9] + 0x1000 > 0x1fff) { |
6973 | 238 av_log(ractx, AV_LOG_ERROR, "Overflow. Broken sample?\n"); |
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239 return 1; |
6973 | 240 } |
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241 |
7516 | 242 for (i=8; i >= 0; i--) { |
243 b = 0x1000-((bp2[i+1] * bp2[i+1]) >> 12); | |
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244 |
7514 | 245 if (!b) |
7513 | 246 b = -2; |
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247 |
7516 | 248 for (j=0; j <= i; j++) |
249 bp1[j] = ((bp2[j] - ((refl[i+1] * bp2[i-j]) >> 12)) * (0x1000000 / b)) >> 12; | |
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250 |
7516 | 251 if ((unsigned) bp1[i] + 0x1000 > 0x1fff) |
7509
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252 return 1; |
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253 |
7801 | 254 refl[i] = bp1[i]; |
255 | |
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Use correct type in FFSWAP (thanks to Benoit Fouet for pointing it out)
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256 FFSWAP(int *, bp1, bp2); |
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257 } |
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258 return 0; |
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259 } |
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260 |
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261 static int interp(RA144Context *ractx, int16_t *out, int a, |
7134 | 262 int copyold, int energy) |
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263 { |
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264 int work[10]; |
6934 | 265 int b = NBLOCKS - a; |
7447 | 266 int i; |
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267 |
7847 | 268 // Interpolate block coefficients from the this frame's forth block and |
269 // last frame's forth block. | |
7447 | 270 for (i=0; i<30; i++) |
271 out[i] = (a * ractx->lpc_coef[0][i] + b * ractx->lpc_coef[1][i])>> 2; | |
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272 |
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273 if (eval_refl(work, out, ractx)) { |
6921 | 274 // The interpolated coefficients are unstable, copy either new or old |
7847 | 275 // coefficients. |
7134 | 276 int_to_int16(out, ractx->lpc_coef[copyold]); |
277 return rescale_rms(ractx->lpc_refl_rms[copyold], energy); | |
6921 | 278 } else { |
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279 return rescale_rms(rms(work), energy); |
6921 | 280 } |
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281 } |
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282 |
7847 | 283 /** Uncompress one block (20 bytes -> 160*2 bytes). */ |
7132 | 284 static int ra144_decode_frame(AVCodecContext * avctx, void *vdata, |
285 int *data_size, const uint8_t *buf, int buf_size) | |
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286 { |
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287 static const uint8_t sizes[10] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2}; |
7084 | 288 unsigned int refl_rms[4]; // RMS of the reflection coefficients |
6966 | 289 uint16_t block_coefs[4][30]; // LPC coefficients of each sub-block |
7005 | 290 unsigned int lpc_refl[10]; // LPC reflection coefficients of the frame |
7447 | 291 int i, j; |
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292 int16_t *data = vdata; |
6898 | 293 unsigned int energy; |
6806 | 294 |
6934 | 295 RA144Context *ractx = avctx->priv_data; |
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296 GetBitContext gb; |
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297 |
7506
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Check available size before writing in decode_frame()
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298 if (*data_size < 2*160) |
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299 return -1; |
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300 |
6854 | 301 if(buf_size < 20) { |
302 av_log(avctx, AV_LOG_ERROR, | |
303 "Frame too small (%d bytes). Truncated file?\n", buf_size); | |
7085
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Do not write junk in the end of truncated files. FATE
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304 *data_size = 0; |
6854 | 305 return buf_size; |
306 } | |
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307 init_get_bits(&gb, buf, 20 * 8); |
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308 |
6805
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309 for (i=0; i<10; i++) |
7127 | 310 lpc_refl[i] = lpc_refl_cb[i][get_bits(&gb, sizes[i])]; |
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311 |
7128
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Use lpc_coef[2] instead of lpc_coef and lpc_coef_old
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312 eval_coefs(ractx->lpc_coef[0], lpc_refl); |
614bc53e67a8
Use lpc_coef[2] instead of lpc_coef and lpc_coef_old
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313 ractx->lpc_refl_rms[0] = rms(lpc_refl); |
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314 |
7126 | 315 energy = energy_tab[get_bits(&gb, 5)]; |
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316 |
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317 refl_rms[0] = interp(ractx, block_coefs[0], 1, 1, ractx->old_energy); |
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318 refl_rms[1] = interp(ractx, block_coefs[1], 2, energy <= ractx->old_energy, |
6947 | 319 t_sqrt(energy*ractx->old_energy) >> 12); |
7830
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320 refl_rms[2] = interp(ractx, block_coefs[2], 3, 0, energy); |
7128
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Use lpc_coef[2] instead of lpc_coef and lpc_coef_old
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7127
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321 refl_rms[3] = rescale_rms(ractx->lpc_refl_rms[0], energy); |
6968
50dd3a733f43
The function dec1() do two completly unrelated things. Split the rms calculation out of it.
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322 |
7128
614bc53e67a8
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323 int_to_int16(block_coefs[3], ractx->lpc_coef[0]); |
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|
324 |
7447 | 325 for (i=0; i < 4; i++) { |
326 do_output_subblock(ractx, block_coefs[i], refl_rms[i], &gb); | |
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327 |
7447 | 328 for (j=0; j < BLOCKSIZE; j++) |
329 *data++ = av_clip_int16(ractx->curr_sblock[j + 10] << 2); | |
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330 } |
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
nickols_k
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diff
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331 |
6898 | 332 ractx->old_energy = energy; |
7128
614bc53e67a8
Use lpc_coef[2] instead of lpc_coef and lpc_coef_old
vitor
parents:
7127
diff
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333 ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0]; |
6781 | 334 |
7128
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Use lpc_coef[2] instead of lpc_coef and lpc_coef_old
vitor
parents:
7127
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335 FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]); |
6781 | 336 |
6828 | 337 *data_size = 2*160; |
6767
f95c5091b802
cosmetics: Reindent file, patch by Vitor Sessak, vitor1001 gmail com,
diego
parents:
6712
diff
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338 return 20; |
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
nickols_k
parents:
diff
changeset
|
339 } |
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
nickols_k
parents:
diff
changeset
|
340 |
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
nickols_k
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diff
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341 AVCodec ra_144_decoder = |
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
nickols_k
parents:
diff
changeset
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342 { |
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
nickols_k
parents:
diff
changeset
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343 "real_144", |
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
nickols_k
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diff
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344 CODEC_TYPE_AUDIO, |
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
nickols_k
parents:
diff
changeset
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345 CODEC_ID_RA_144, |
6894
eb3c16b99f27
Rename context struct to be more consistent with the rest of ffmpeg
vitor
parents:
6888
diff
changeset
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346 sizeof(RA144Context), |
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
nickols_k
parents:
diff
changeset
|
347 ra144_decode_init, |
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
nickols_k
parents:
diff
changeset
|
348 NULL, |
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
nickols_k
parents:
diff
changeset
|
349 NULL, |
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
nickols_k
parents:
diff
changeset
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350 ra144_decode_frame, |
7040
e943e1409077
Make AVCodec long_names definition conditional depending on CONFIG_SMALL.
stefano
parents:
7005
diff
changeset
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351 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"), |
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
nickols_k
parents:
diff
changeset
|
352 }; |