Mercurial > libavcodec.hg
annotate ra144.c @ 11300:1f512abdf680 libavcodec
Simplify code to set cbp_*
this seems 1 cpu cycle slower even though we practically just remove code.
Speed loss seems caused by the merge of if(left_type), iam commiting this
anyway as i cant imagine this to be anything but compiler messup.
author | michael |
---|---|
date | Fri, 26 Feb 2010 09:13:40 +0000 |
parents | e8461dae9672 |
children | 421c9441c11e |
rev | line source |
---|---|
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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1 /* |
e8543aab0cc9
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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 | |
1304
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7 * |
3947
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Change license headers to say 'FFmpeg' instead of 'this program/this library'
diego
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8 * This file is part of FFmpeg. |
c8c591fe26f8
Change license headers to say 'FFmpeg' instead of 'this program/this library'
diego
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9 * |
c8c591fe26f8
Change license headers to say 'FFmpeg' instead of 'this program/this library'
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10 * FFmpeg is free software; you can redistribute it and/or |
1304
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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 |
3947
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Change license headers to say 'FFmpeg' instead of 'this program/this library'
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13 * version 2.1 of the License, or (at your option) any later version. |
1304
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14 * |
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15 * FFmpeg is distributed in the hope that it will be useful, |
1304
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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 |
3947
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Change license headers to say 'FFmpeg' instead of 'this program/this library'
diego
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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.
diego
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22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
1304
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23 */ |
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24 |
e8543aab0cc9
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25 #include "avcodec.h" |
9428 | 26 #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:
1305
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27 #include "ra144.h" |
8049
611a21e4b01b
Split off celp_filters.[ch] from acelp_filters.[ch] for the QCELP decoder.
diego
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7847
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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 | |
1304
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33 |
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34 |
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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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35 typedef struct { |
10776
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Use correct context for av_log(), should prevent a crash for malformed files.
vitor
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9428
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36 AVCodecContext *avctx; |
816067b00019
Use correct context for av_log(), should prevent a crash for malformed files.
vitor
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37 |
6910 | 38 unsigned int old_energy; ///< previous frame energy |
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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39 |
7005 | 40 unsigned int lpc_tables[2][10]; |
7128
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Use lpc_coef[2] instead of lpc_coef and lpc_coef_old
vitor
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41 |
614bc53e67a8
Use lpc_coef[2] instead of lpc_coef and lpc_coef_old
vitor
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42 /** LPC coefficients: lpc_coef[0] is the coefficients of the current frame |
7847 | 43 * and lpc_coef[1] of the previous one. */ |
7128
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Use lpc_coef[2] instead of lpc_coef and lpc_coef_old
vitor
parents:
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44 unsigned int *lpc_coef[2]; |
614bc53e67a8
Use lpc_coef[2] instead of lpc_coef and lpc_coef_old
vitor
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45 |
614bc53e67a8
Use lpc_coef[2] instead of lpc_coef and lpc_coef_old
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46 unsigned int lpc_refl_rms[2]; |
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
nickols_k
parents:
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47 |
7847 | 48 /** The current subblock padded by the last 10 values of the previous one. */ |
7096
1ad555e54621
Avoid a memcpy. Code is also simpler now in my opinion.
vitor
parents:
7085
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49 int16_t curr_sblock[50]; |
1ad555e54621
Avoid a memcpy. Code is also simpler now in my opinion.
vitor
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50 |
7847 | 51 /** Adaptive codebook, its size is two units bigger to avoid a |
52 * buffer overflow. */ | |
7827
82fc486cd128
Cosmetics: Make the two additional units in size more clear
vitor
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7801
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53 uint16_t adapt_cb[146+2]; |
6894
eb3c16b99f27
Rename context struct to be more consistent with the rest of ffmpeg
vitor
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54 } RA144Context; |
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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55 |
7828 | 56 static av_cold int ra144_decode_init(AVCodecContext * avctx) |
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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57 { |
6895
c63e9357e2a8
Context vars are not global vars. glob is a bad name for it.
vitor
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58 RA144Context *ractx = avctx->priv_data; |
1304
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59 |
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Use correct context for av_log(), should prevent a crash for malformed files.
vitor
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60 ractx->avctx = avctx; |
816067b00019
Use correct context for av_log(), should prevent a crash for malformed files.
vitor
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61 |
7128
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Use lpc_coef[2] instead of lpc_coef and lpc_coef_old
vitor
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7127
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62 ractx->lpc_coef[0] = ractx->lpc_tables[0]; |
614bc53e67a8
Use lpc_coef[2] instead of lpc_coef and lpc_coef_old
vitor
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7127
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63 ractx->lpc_coef[1] = ractx->lpc_tables[1]; |
1304
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64 |
7451
85ab7655ad4d
Modify all codecs to report their supported input and output sample format(s).
pross
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65 avctx->sample_fmt = SAMPLE_FMT_S16; |
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diego
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66 return 0; |
1304
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nickols_k
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67 } |
e8543aab0cc9
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68 |
6899 | 69 /** |
6902 | 70 * Evaluate sqrt(x << 24). x must fit in 20 bits. This value is evaluated in an |
6899 | 71 * odd way to make the output identical to the binary decoder. |
72 */ | |
1304
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73 static int t_sqrt(unsigned int x) |
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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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74 { |
7104 | 75 int s = 2; |
6767
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76 while (x > 0xfff) { |
f95c5091b802
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77 s++; |
7507 | 78 x >>= 2; |
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79 } |
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80 |
7104 | 81 return ff_sqrt(x << 20) << s; |
1304
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82 } |
e8543aab0cc9
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83 |
6952 | 84 /** |
85 * Evaluate the LPC filter coefficients from the reflection coefficients. | |
6965 | 86 * Does the inverse of the eval_refl() function. |
6952 | 87 */ |
7120
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Consistency: make the output buffer the first argument of functions
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88 static void eval_coefs(int *coefs, const int *refl) |
1304
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89 { |
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90 int buffer[10]; |
6782 | 91 int *b1 = buffer; |
6964 | 92 int *b2 = coefs; |
7447 | 93 int i, j; |
2967 | 94 |
7447 | 95 for (i=0; i < 10; i++) { |
96 b1[i] = refl[i] << 4; | |
6782 | 97 |
7447 | 98 for (j=0; j < i; j++) |
99 b1[j] = ((refl[i] * b2[i-j-1]) >> 12) + b2[j]; | |
1304
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100 |
6781 | 101 FFSWAP(int *, b1, b2); |
1304
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102 } |
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103 |
7447 | 104 for (i=0; i < 10; i++) |
105 coefs[i] >>= 4; | |
1304
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106 } |
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107 |
7105
b55bedad89ad
Rename rotate_buffer() to copy_and_dup() and doxy it
vitor
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7104
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108 /** |
b55bedad89ad
Rename rotate_buffer() to copy_and_dup() and doxy it
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7104
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109 * Copy the last offset values of *source to *target. If those values are not |
b55bedad89ad
Rename rotate_buffer() to copy_and_dup() and doxy it
vitor
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7104
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110 * enough to fill the target buffer, fill it with another copy of those values. |
b55bedad89ad
Rename rotate_buffer() to copy_and_dup() and doxy it
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7104
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111 */ |
7120
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112 static void copy_and_dup(int16_t *target, const int16_t *source, int offset) |
1304
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113 { |
6831 | 114 source += BUFFERSIZE - offset; |
6767
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115 |
7829 | 116 memcpy(target, source, FFMIN(BLOCKSIZE, offset)*sizeof(*target)); |
117 if (offset < BLOCKSIZE) | |
7100 | 118 memcpy(target + offset, source, (BLOCKSIZE - offset)*sizeof(*target)); |
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119 } |
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120 |
7110 | 121 /** inverse root mean square */ |
7112 | 122 static int irms(const int16_t *data) |
1304
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123 { |
6838 | 124 unsigned int i, sum = 0; |
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125 |
6838 | 126 for (i=0; i < BLOCKSIZE; i++) |
127 sum += data[i] * data[i]; | |
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128 |
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129 if (sum == 0) |
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130 return 0; /* OOPS - division by zero */ |
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131 |
7112 | 132 return 0x20000000 / (t_sqrt(sum) >> 8); |
1304
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133 } |
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134 |
7120
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135 static void add_wav(int16_t *dest, int n, int skip_first, int *m, |
c98b43981d89
Consistency: make the output buffer the first argument of functions
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136 const int16_t *s1, const int8_t *s2, const int8_t *s3) |
1304
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137 { |
6873
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Make add_wav() receive a vector instead of three integers
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138 int i; |
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Make add_wav() receive a vector instead of three integers
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139 int v[3]; |
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140 |
6873
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Make add_wav() receive a vector instead of three integers
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6870
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141 v[0] = 0; |
6880 | 142 for (i=!skip_first; i<3; i++) |
7834
b9a26c7e64ef
Normalize triplets in gain_val_tab[][] so gain_exp_tab[] can be just an
vitor
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7830
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143 v[i] = (gain_val_tab[n][i] * m[i]) >> gain_exp_tab[n]; |
6767
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144 |
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145 if (v[0]) { |
9078 | 146 for (i=0; i < BLOCKSIZE; i++) |
147 dest[i] = (s1[i]*v[0] + s2[i]*v[1] + s3[i]*v[2]) >> 12; | |
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148 } else { |
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149 for (i=0; i < BLOCKSIZE; i++) |
9078 | 150 dest[i] = ( s2[i]*v[1] + s3[i]*v[2]) >> 12; |
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151 } |
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152 } |
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153 |
7113 | 154 static unsigned int rescale_rms(unsigned int rms, unsigned int energy) |
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155 { |
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156 return (rms * energy) >> 10; |
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157 } |
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158 |
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159 static unsigned int rms(const int *data) |
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160 { |
7447 | 161 int i; |
6850 | 162 unsigned int res = 0x10000; |
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163 int b = 10; |
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164 |
7447 | 165 for (i=0; i < 10; i++) { |
166 res = (((0x1000000 - data[i]*data[i]) >> 12) * res) >> 12; | |
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167 |
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168 if (res == 0) |
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169 return 0; |
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170 |
6850 | 171 while (res <= 0x3fff) { |
172 b++; | |
173 res <<= 2; | |
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174 } |
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175 } |
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176 |
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177 return t_sqrt(res) >> b; |
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178 } |
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179 |
7133 | 180 static void do_output_subblock(RA144Context *ractx, const uint16_t *lpc_coefs, |
181 int gval, GetBitContext *gb) | |
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182 { |
6887 | 183 uint16_t buffer_a[40]; |
184 uint16_t *block; | |
6907 | 185 int cba_idx = get_bits(gb, 7); // index of the adaptive CB, 0 if none |
6900 | 186 int gain = get_bits(gb, 8); |
187 int cb1_idx = get_bits(gb, 7); | |
188 int cb2_idx = get_bits(gb, 7); | |
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189 int m[3]; |
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190 |
6900 | 191 if (cba_idx) { |
7106 | 192 cba_idx += BLOCKSIZE/2 - 1; |
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193 copy_and_dup(buffer_a, ractx->adapt_cb, cba_idx); |
7112 | 194 m[0] = (irms(buffer_a) * gval) >> 12; |
6888 | 195 } else { |
196 m[0] = 0; | |
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197 } |
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198 |
7121 | 199 m[1] = (cb1_base[cb1_idx] * gval) >> 8; |
200 m[2] = (cb2_base[cb2_idx] * gval) >> 8; | |
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201 |
6896 | 202 memmove(ractx->adapt_cb, ractx->adapt_cb + BLOCKSIZE, |
7119 | 203 (BUFFERSIZE - BLOCKSIZE) * sizeof(*ractx->adapt_cb)); |
6882 | 204 |
6896 | 205 block = ractx->adapt_cb + BUFFERSIZE - BLOCKSIZE; |
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206 |
9081 | 207 add_wav(block, gain, cba_idx, m, cba_idx? buffer_a: NULL, |
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208 cb1_vects[cb1_idx], cb2_vects[cb2_idx]); |
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209 |
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210 memcpy(ractx->curr_sblock, ractx->curr_sblock + 40, |
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211 10*sizeof(*ractx->curr_sblock)); |
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212 |
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213 if (ff_celp_lp_synthesis_filter(ractx->curr_sblock + 10, lpc_coefs, |
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214 block, BLOCKSIZE, 10, 1, 0xfff)) |
7162
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215 memset(ractx->curr_sblock, 0, 50*sizeof(*ractx->curr_sblock)); |
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216 } |
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217 |
7077 | 218 static void int_to_int16(int16_t *out, const int *inp) |
6934 | 219 { |
220 int i; | |
221 | |
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222 for (i=0; i < 10; i++) |
7800 | 223 *out++ = *inp++; |
6934 | 224 } |
225 | |
6952 | 226 /** |
227 * Evaluate the reflection coefficients from the filter coefficients. | |
6965 | 228 * Does the inverse of the eval_coefs() function. |
6952 | 229 * |
7847 | 230 * @return 1 if one of the reflection coefficients is greater than |
6952 | 231 * 4095, 0 if not. |
232 */ | |
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233 static int eval_refl(int *refl, const int16_t *coefs, AVCodecContext *avctx) |
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234 { |
7516 | 235 int b, i, j; |
6806 | 236 int buffer1[10]; |
237 int buffer2[10]; | |
6852 | 238 int *bp1 = buffer1; |
239 int *bp2 = buffer2; | |
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240 |
6852 | 241 for (i=0; i < 10; i++) |
6964 | 242 buffer2[i] = coefs[i]; |
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243 |
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244 refl[9] = bp2[9]; |
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245 |
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246 if ((unsigned) bp2[9] + 0x1000 > 0x1fff) { |
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247 av_log(avctx, AV_LOG_ERROR, "Overflow. Broken sample?\n"); |
7116
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It makes more sense to return 1 here in case of overflow.
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248 return 1; |
6973 | 249 } |
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250 |
7516 | 251 for (i=8; i >= 0; i--) { |
252 b = 0x1000-((bp2[i+1] * bp2[i+1]) >> 12); | |
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253 |
7514 | 254 if (!b) |
7513 | 255 b = -2; |
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256 |
7516 | 257 for (j=0; j <= i; j++) |
258 bp1[j] = ((bp2[j] - ((refl[i+1] * bp2[i-j]) >> 12)) * (0x1000000 / b)) >> 12; | |
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259 |
7516 | 260 if ((unsigned) bp1[i] + 0x1000 > 0x1fff) |
7509
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261 return 1; |
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262 |
7801 | 263 refl[i] = bp1[i]; |
264 | |
6804
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Use correct type in FFSWAP (thanks to Benoit Fouet for pointing it out)
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265 FFSWAP(int *, bp1, bp2); |
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266 } |
7509
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267 return 0; |
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268 } |
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269 |
7830
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270 static int interp(RA144Context *ractx, int16_t *out, int a, |
7134 | 271 int copyold, int energy) |
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272 { |
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273 int work[10]; |
6934 | 274 int b = NBLOCKS - a; |
7447 | 275 int i; |
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276 |
7847 | 277 // Interpolate block coefficients from the this frame's forth block and |
278 // last frame's forth block. | |
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279 for (i=0; i<10; i++) |
7447 | 280 out[i] = (a * ractx->lpc_coef[0][i] + b * ractx->lpc_coef[1][i])>> 2; |
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281 |
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282 if (eval_refl(work, out, ractx->avctx)) { |
6921 | 283 // The interpolated coefficients are unstable, copy either new or old |
7847 | 284 // coefficients. |
7134 | 285 int_to_int16(out, ractx->lpc_coef[copyold]); |
286 return rescale_rms(ractx->lpc_refl_rms[copyold], energy); | |
6921 | 287 } else { |
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Split function rms() in two (it does two unrelated things)
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288 return rescale_rms(rms(work), energy); |
6921 | 289 } |
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290 } |
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291 |
7847 | 292 /** Uncompress one block (20 bytes -> 160*2 bytes). */ |
7132 | 293 static int ra144_decode_frame(AVCodecContext * avctx, void *vdata, |
9355
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Implement avcodec_decode_video2(), _audio3() and _subtitle2() which takes an
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|
294 int *data_size, AVPacket *avpkt) |
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295 { |
9355
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296 const uint8_t *buf = avpkt->data; |
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Implement avcodec_decode_video2(), _audio3() and _subtitle2() which takes an
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297 int buf_size = avpkt->size; |
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298 static const uint8_t sizes[10] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2}; |
7084 | 299 unsigned int refl_rms[4]; // RMS of the reflection coefficients |
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300 uint16_t block_coefs[4][10]; // LPC coefficients of each sub-block |
7005 | 301 unsigned int lpc_refl[10]; // LPC reflection coefficients of the frame |
7447 | 302 int i, j; |
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303 int16_t *data = vdata; |
6898 | 304 unsigned int energy; |
6806 | 305 |
6934 | 306 RA144Context *ractx = avctx->priv_data; |
6805
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307 GetBitContext gb; |
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308 |
7506
e94202a2e0b6
Check available size before writing in decode_frame()
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309 if (*data_size < 2*160) |
e94202a2e0b6
Check available size before writing in decode_frame()
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310 return -1; |
e94202a2e0b6
Check available size before writing in decode_frame()
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311 |
6854 | 312 if(buf_size < 20) { |
313 av_log(avctx, AV_LOG_ERROR, | |
314 "Frame too small (%d bytes). Truncated file?\n", buf_size); | |
7085
f55f71e2be57
Do not write junk in the end of truncated files. FATE
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7084
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315 *data_size = 0; |
6854 | 316 return buf_size; |
317 } | |
6805
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Read the bitstream where the data is actually needed instead of doing it in unpack_input()
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318 init_get_bits(&gb, buf, 20 * 8); |
1304
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319 |
6805
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320 for (i=0; i<10; i++) |
7127 | 321 lpc_refl[i] = lpc_refl_cb[i][get_bits(&gb, sizes[i])]; |
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322 |
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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323 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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324 ractx->lpc_refl_rms[0] = rms(lpc_refl); |
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325 |
7126 | 326 energy = energy_tab[get_bits(&gb, 5)]; |
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327 |
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328 refl_rms[0] = interp(ractx, block_coefs[0], 1, 1, ractx->old_energy); |
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329 refl_rms[1] = interp(ractx, block_coefs[1], 2, energy <= ractx->old_energy, |
6947 | 330 t_sqrt(energy*ractx->old_energy) >> 12); |
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331 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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332 refl_rms[3] = rescale_rms(ractx->lpc_refl_rms[0], energy); |
6968
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The function dec1() do two completly unrelated things. Split the rms calculation out of it.
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333 |
7128
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Use lpc_coef[2] instead of lpc_coef and lpc_coef_old
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334 int_to_int16(block_coefs[3], ractx->lpc_coef[0]); |
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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335 |
7447 | 336 for (i=0; i < 4; i++) { |
337 do_output_subblock(ractx, block_coefs[i], refl_rms[i], &gb); | |
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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338 |
7447 | 339 for (j=0; j < BLOCKSIZE; j++) |
340 *data++ = av_clip_int16(ractx->curr_sblock[j + 10] << 2); | |
6767
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cosmetics: Reindent file, patch by Vitor Sessak, vitor1001 gmail com,
diego
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341 } |
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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342 |
6898 | 343 ractx->old_energy = energy; |
7128
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Use lpc_coef[2] instead of lpc_coef and lpc_coef_old
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344 ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0]; |
6781 | 345 |
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346 FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]); |
6781 | 347 |
6828 | 348 *data_size = 2*160; |
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349 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
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350 } |
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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351 |
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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352 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
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353 { |
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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354 "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
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355 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
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356 CODEC_ID_RA_144, |
6894
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Rename context struct to be more consistent with the rest of ffmpeg
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357 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
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358 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
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359 NULL, |
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diff
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|
360 NULL, |
e8543aab0cc9
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361 ra144_decode_frame, |
7040
e943e1409077
Make AVCodec long_names definition conditional depending on CONFIG_SMALL.
stefano
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7005
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362 .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
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diff
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363 }; |