Mercurial > libavcodec.hg
annotate ra288.c @ 7787:681a05d9b04f libavcodec
Ignore the spec, and calculate aspect ratio on width/height.
Fixes issue562 and issue621.
author | michael |
---|---|
date | Wed, 03 Sep 2008 19:04:46 +0000 |
parents | 2c22852d1998 |
children | ffd4b1364b62 |
rev | line source |
---|---|
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
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1 /* |
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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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
Change license headers to say 'FFmpeg' instead of 'this program/this library'
diego
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6 * |
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Change license headers to say 'FFmpeg' instead of 'this program/this library'
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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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Change license headers to say 'FFmpeg' instead of 'this program/this library'
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10 * version 2.1 of the License, or (at your option) any later version. |
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
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11 * |
3947
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diego
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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 |
e8543aab0cc9
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nickols_k
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15 * Lesser General Public License for more details. |
e8543aab0cc9
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nickols_k
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16 * |
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nickols_k
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17 * 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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3036
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18 * License along with FFmpeg; if not, write to the Free Software |
3036
0b546eab515d
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
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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
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20 */ |
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nickols_k
parents:
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21 |
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:
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22 #include "avcodec.h" |
7171 | 23 #define ALT_BITSTREAM_READER_LE |
24 #include "bitstream.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" |
2967 | 26 |
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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27 typedef struct { |
7439 | 28 float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A) |
29 float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB) | |
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 |
7439 | 31 float sp_hist[111]; ///< Speech data history (spec: SB) |
7418
83d005a098b5
Rename st{1,2}{a,b} context variables to more meaningful names: {sp,gain}_{rec,hist}
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32 |
83d005a098b5
Rename st{1,2}{a,b} context variables to more meaningful names: {sp,gain}_{rec,hist}
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33 /** Speech part of the gain autocorrelation (spec: REXP) */ |
83d005a098b5
Rename st{1,2}{a,b} context variables to more meaningful names: {sp,gain}_{rec,hist}
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34 float sp_rec[37]; |
83d005a098b5
Rename st{1,2}{a,b} context variables to more meaningful names: {sp,gain}_{rec,hist}
vitor
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35 |
83d005a098b5
Rename st{1,2}{a,b} context variables to more meaningful names: {sp,gain}_{rec,hist}
vitor
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7416
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36 float gain_hist[38]; ///< Log-gain history (spec: SBLG) |
83d005a098b5
Rename st{1,2}{a,b} context variables to more meaningful names: {sp,gain}_{rec,hist}
vitor
parents:
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37 |
83d005a098b5
Rename st{1,2}{a,b} context variables to more meaningful names: {sp,gain}_{rec,hist}
vitor
parents:
7416
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38 /** Recursive part of the gain autocorrelation (spec: REXPLG) */ |
83d005a098b5
Rename st{1,2}{a,b} context variables to more meaningful names: {sp,gain}_{rec,hist}
vitor
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7416
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39 float gain_rec[11]; |
83d005a098b5
Rename st{1,2}{a,b} context variables to more meaningful names: {sp,gain}_{rec,hist}
vitor
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7416
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40 |
7438
d9d070ff46bf
Rename two context vars: s/sb/sp_block/, s/lhist/gain_block/
vitor
parents:
7437
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41 float sp_block[41]; ///< Speech data of four blocks (spec: STTMP) |
d9d070ff46bf
Rename two context vars: s/sb/sp_block/, s/lhist/gain_block/
vitor
parents:
7437
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42 float gain_block[10]; ///< Gain data of four blocks (spec: GSTATE) |
7411
d41948bf80a1
Give the context structure of ra288 a better name: s/Real288_internal/RA288Context/
vitor
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7375
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43 } 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
nickols_k
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44 |
7451
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Modify all codecs to report their supported input and output sample format(s).
pross
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45 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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46 { |
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Modify all codecs to report their supported input and output sample format(s).
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47 avctx->sample_fmt = SAMPLE_FMT_S16; |
85ab7655ad4d
Modify all codecs to report their supported input and output sample format(s).
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48 return 0; |
85ab7655ad4d
Modify all codecs to report their supported input and output sample format(s).
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|
49 } |
85ab7655ad4d
Modify all codecs to report their supported input and output sample format(s).
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50 |
7294 | 51 static inline float scalar_product_float(const float * v1, const float * v2, |
52 int size) | |
7201
4d441356fa23
Create a function for float dot product instead of duplicating all over the file
vitor
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53 { |
4d441356fa23
Create a function for float dot product instead of duplicating all over the file
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54 float res = 0.; |
4d441356fa23
Create a function for float dot product instead of duplicating all over the file
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55 |
4d441356fa23
Create a function for float dot product instead of duplicating all over the file
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56 while (size--) |
4d441356fa23
Create a function for float dot product instead of duplicating all over the file
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57 res += *v1++ * *v2++; |
4d441356fa23
Create a function for float dot product instead of duplicating all over the file
vitor
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58 |
4d441356fa23
Create a function for float dot product instead of duplicating all over the file
vitor
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59 return res; |
4d441356fa23
Create a function for float dot product instead of duplicating all over the file
vitor
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60 } |
4d441356fa23
Create a function for float dot product instead of duplicating all over the file
vitor
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7195
diff
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61 |
7415
649496faa1d2
Move colmult() function to the beginning of file to group DSP-related functions.
vitor
parents:
7414
diff
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62 static void colmult(float *tgt, const float *m1, const float *m2, int n) |
649496faa1d2
Move colmult() function to the beginning of file to group DSP-related functions.
vitor
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7414
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63 { |
649496faa1d2
Move colmult() function to the beginning of file to group DSP-related functions.
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64 while (n--) |
7416 | 65 *tgt++ = *m1++ * *m2++; |
7415
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Move colmult() function to the beginning of file to group DSP-related functions.
vitor
parents:
7414
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66 } |
649496faa1d2
Move colmult() function to the beginning of file to group DSP-related functions.
vitor
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7414
diff
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67 |
7412
51da3d8200d0
Give the context variables of ra288 a better name: s/glob/ractx/
vitor
parents:
7411
diff
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68 static void decode(RA288Context *ractx, float gain, int cb_coef) |
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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69 { |
7448 | 70 int i, j; |
7268
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Do not declare as double a var that only stores a float
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71 double sumsum; |
7e51fc7805ac
Do not declare as double a var that only stores a float
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72 float sum, buffer[5]; |
7503
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Flip (by making buf[i] -> buf[size-i-1]) two buffers: {sp,gain}_block. This
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73 float *block = ractx->sp_block + 36; // Current block |
7167 | 74 |
7503
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Flip (by making buf[i] -> buf[size-i-1]) two buffers: {sp,gain}_block. This
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75 memmove(ractx->sp_block, ractx->sp_block + 5, 36*sizeof(*ractx->sp_block)); |
7167 | 76 |
7503
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Flip (by making buf[i] -> buf[size-i-1]) two buffers: {sp,gain}_block. This
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77 for (i=0; i < 5; i++) { |
d85a0a151344
Flip (by making buf[i] -> buf[size-i-1]) two buffers: {sp,gain}_block. This
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78 block[i] = 0.; |
d85a0a151344
Flip (by making buf[i] -> buf[size-i-1]) two buffers: {sp,gain}_block. This
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79 for (j=0; j < 36; j++) |
d85a0a151344
Flip (by making buf[i] -> buf[size-i-1]) two buffers: {sp,gain}_block. This
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80 block[i] -= block[i-1-j]*ractx->sp_lpc[j]; |
d85a0a151344
Flip (by making buf[i] -> buf[size-i-1]) two buffers: {sp,gain}_block. This
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81 } |
7167 | 82 |
7431
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Add a few comments pointing to the G.728 specification
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83 /* block 46 of G.728 spec */ |
7503
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84 sum = 32.; |
d85a0a151344
Flip (by making buf[i] -> buf[size-i-1]) two buffers: {sp,gain}_block. This
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85 for (i=0; i < 10; i++) |
d85a0a151344
Flip (by making buf[i] -> buf[size-i-1]) two buffers: {sp,gain}_block. This
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86 sum -= ractx->gain_block[9-i] * ractx->gain_lpc[i]; |
7167 | 87 |
7431
da24b77f46f6
Add a few comments pointing to the G.728 specification
vitor
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88 /* block 47 of G.728 spec */ |
7269 | 89 sum = av_clipf(sum, 0, 60); |
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
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|
90 |
7431
da24b77f46f6
Add a few comments pointing to the G.728 specification
vitor
parents:
7430
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|
91 /* block 48 of G.728 spec */ |
7440 | 92 sumsum = exp(sum * 0.1151292546497) * gain; /* pow(10.0,sum/20)*gain */ |
7167 | 93 |
7448 | 94 for (i=0; i < 5; i++) |
95 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
nickols_k
parents:
diff
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|
96 |
7275 | 97 sum = scalar_product_float(buffer, buffer, 5) / 5; |
98 | |
99 sum = FFMAX(sum, 1); | |
7167 | 100 |
101 /* shift and store */ | |
7503
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Flip (by making buf[i] -> buf[size-i-1]) two buffers: {sp,gain}_block. This
vitor
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102 memmove(ractx->gain_block, ractx->gain_block + 1, |
d85a0a151344
Flip (by making buf[i] -> buf[size-i-1]) two buffers: {sp,gain}_block. This
vitor
parents:
7492
diff
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103 9 * sizeof(*ractx->gain_block)); |
7167 | 104 |
7503
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Flip (by making buf[i] -> buf[size-i-1]) two buffers: {sp,gain}_block. This
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105 ractx->gain_block[9] = 10 * log10(sum) - 32; |
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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106 |
7448 | 107 for (i=1; i < 5; i++) |
108 for (j=i-1; j >= 0; j--) | |
109 buffer[i] -= ractx->sp_lpc[i-j-1] * buffer[j]; | |
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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110 |
7167 | 111 /* output */ |
7448 | 112 for (i=0; i < 5; i++) |
7503
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Flip (by making buf[i] -> buf[size-i-1]) two buffers: {sp,gain}_block. This
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113 block[i] = av_clipf(block[i] + buffer[i], -4095, 4095); |
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
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114 } |
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
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115 |
7369
b0d8a3985f96
Rename pred() to eval_lpc_coeffs() and add a doxy comment
vitor
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116 /** |
b0d8a3985f96
Rename pred() to eval_lpc_coeffs() and add a doxy comment
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117 * Converts autocorrelation coefficients to LPC coefficients using the |
b0d8a3985f96
Rename pred() to eval_lpc_coeffs() and add a doxy comment
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118 * Levinson-Durbin algorithm. See blocks 37 and 50 of the G.728 specification. |
b0d8a3985f96
Rename pred() to eval_lpc_coeffs() and add a doxy comment
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119 * |
7370
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Follow FFmpeg convention of returning negative values on error in eval_lpc_coeffs()
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120 * @return 0 if success, -1 if fail |
7369
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Rename pred() to eval_lpc_coeffs() and add a doxy comment
vitor
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121 */ |
b0d8a3985f96
Rename pred() to eval_lpc_coeffs() and add a doxy comment
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122 static int eval_lpc_coeffs(const float *in, float *tgt, int n) |
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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123 { |
7448 | 124 int i, j; |
7167 | 125 double f0, f1, f2; |
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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126 |
7167 | 127 if (in[n] == 0) |
7370
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Follow FFmpeg convention of returning negative values on error in eval_lpc_coeffs()
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128 return -1; |
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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129 |
7167 | 130 if ((f0 = *in) <= 0) |
7370
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Follow FFmpeg convention of returning negative values on error in eval_lpc_coeffs()
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131 return -1; |
7167 | 132 |
7296 | 133 in--; // To avoid a -1 subtraction in the inner loop |
134 | |
7448 | 135 for (i=1; i <= n; i++) { |
136 f1 = in[i+1]; | |
7195
5e5c1178ae9a
Write for loops in a more standard way. In my opinion, they are much more readable now.
vitor
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137 |
7448 | 138 for (j=0; j < i - 1; j++) |
139 f1 += in[i-j]*tgt[j]; | |
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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140 |
7448 | 141 tgt[i-1] = f2 = -f1/f0; |
142 for (j=0; j < i >> 1; j++) { | |
143 float temp = tgt[j] + tgt[i-j-2]*f2; | |
144 tgt[i-j-2] += tgt[j]*f2; | |
145 tgt[j] = temp; | |
7167 | 146 } |
147 if ((f0 += f1*f2) < 0) | |
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148 return -1; |
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149 } |
7295 | 150 |
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151 return 0; |
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152 } |
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153 |
7492 | 154 static void convolve(float *tgt, const float *src, int len, int n) |
7169 | 155 { |
7201
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156 for (; n >= 0; n--) |
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157 tgt[n] = scalar_product_float(src, src - n, len); |
7169 | 158 |
159 } | |
160 | |
7367 | 161 /** |
162 * Hybrid window filtering. See blocks 36 and 49 of the G.728 specification. | |
163 * | |
164 * @param order the order of the filter | |
165 * @param n the length of the input | |
7375 | 166 * @param non_rec the number of non-recursive samples |
7367 | 167 * @param out the filter output |
168 * @param in pointer to the input of the filter | |
169 * @param hist pointer to the input history of the filter. It is updated by | |
170 * this function. | |
7375 | 171 * @param out pointer to the non-recursive part of the output |
7367 | 172 * @param out2 pointer to the recursive part of the output |
173 * @param window pointer to the windowing function table | |
174 */ | |
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175 static void do_hybrid_window(int order, int n, int non_rec, const float *in, |
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176 float *out, float *hist, float *out2, |
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177 const float *window) |
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178 { |
7449
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179 int i; |
7437
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180 float buffer1[order + 1]; |
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181 float buffer2[order + 1]; |
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182 float work[order + n + non_rec]; |
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183 |
7367 | 184 /* update history */ |
7504 | 185 memmove(hist , hist + n, (order + non_rec)*sizeof(*hist)); |
186 memcpy (hist + order + non_rec, in , n *sizeof(*hist)); | |
7363
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187 |
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188 colmult(work, window, hist, order + n + non_rec); |
2967 | 189 |
7492 | 190 convolve(buffer1, work + order , n , order); |
191 convolve(buffer2, work + order + n, non_rec, order); | |
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192 |
7448 | 193 for (i=0; i <= order; i++) { |
194 out2[i] = out2[i] * 0.5625 + buffer1[i]; | |
195 out [i] = out2[i] + buffer2[i]; | |
7167 | 196 } |
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197 |
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198 /* Multiply by the white noise correcting factor (WNCF) */ |
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199 *out *= 257./256.; |
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200 } |
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201 |
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202 /** |
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203 * Backward synthesis filter. Find the LPC coefficients from past speech data. |
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204 */ |
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205 static void backward_filter(RA288Context *ractx) |
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206 { |
7428
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207 float temp1[37]; // RTMP in the spec |
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208 float temp2[11]; // GPTPMP in the spec |
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209 |
7503
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210 do_hybrid_window(36, 40, 35, ractx->sp_block+1, temp1, ractx->sp_hist, |
7428
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211 ractx->sp_rec, syn_window); |
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212 |
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213 if (!eval_lpc_coeffs(temp1, ractx->sp_lpc, 36)) |
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214 colmult(ractx->sp_lpc, ractx->sp_lpc, syn_bw_tab, 36); |
7169 | 215 |
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216 do_hybrid_window(10, 8, 20, ractx->gain_block+2, temp2, ractx->gain_hist, |
7428
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217 ractx->gain_rec, gain_window); |
7169 | 218 |
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219 if (!eval_lpc_coeffs(temp2, ractx->gain_lpc, 10)) |
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220 colmult(ractx->gain_lpc, ractx->gain_lpc, gain_bw_tab, 10); |
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221 } |
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222 |
7174 | 223 static int ra288_decode_frame(AVCodecContext * avctx, void *data, |
224 int *data_size, const uint8_t * buf, | |
225 int buf_size) | |
1313 | 226 { |
7174 | 227 int16_t *out = data; |
7448 | 228 int i, j; |
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229 RA288Context *ractx = avctx->priv_data; |
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230 GetBitContext gb; |
7167 | 231 |
7174 | 232 if (buf_size < avctx->block_align) { |
233 av_log(avctx, AV_LOG_ERROR, | |
234 "Error! Input buffer is too small [%d<%d]\n", | |
235 buf_size, avctx->block_align); | |
236 return 0; | |
237 } | |
238 | |
7505 | 239 if (*data_size < 32*5*2) |
240 return -1; | |
241 | |
7174 | 242 init_get_bits(&gb, buf, avctx->block_align * 8); |
1313 | 243 |
7448 | 244 for (i=0; i < 32; i++) { |
7192 | 245 float gain = amptable[get_bits(&gb, 3)]; |
7448 | 246 int cb_coef = get_bits(&gb, 6 + (i&1)); |
7443 | 247 |
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248 decode(ractx, gain, cb_coef); |
7167 | 249 |
7448 | 250 for (j=0; j < 5; j++) |
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251 *(out++) = 8 * ractx->sp_block[36 + j]; |
7167 | 252 |
7448 | 253 if ((i & 7) == 3) |
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254 backward_filter(ractx); |
7167 | 255 } |
256 | |
7174 | 257 *data_size = (char *)out - (char *)data; |
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258 return avctx->block_align; |
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259 } |
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260 |
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261 AVCodec ra_288_decoder = |
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262 { |
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263 "real_288", |
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264 CODEC_TYPE_AUDIO, |
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265 CODEC_ID_RA_288, |
7411
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266 sizeof(RA288Context), |
7451
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Modify all codecs to report their supported input and output sample format(s).
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267 ra288_decode_init, |
1304
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268 NULL, |
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269 NULL, |
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270 ra288_decode_frame, |
7040
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271 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"), |
1304
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272 }; |