Mercurial > libavcodec.hg
annotate acelp_filters.c @ 7700:53fb5eb360d5 libavcodec
simplify mpeg12 avctx->pix_fmt set
author | bcoudurier |
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date | Tue, 26 Aug 2008 23:25:29 +0000 |
parents | 76cf62255b28 |
children | 611a21e4b01b |
rev | line source |
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6772 | 1 /* |
2 * various filters for ACELP-based codecs | |
3 * | |
4 * Copyright (c) 2008 Vladimir Voroshilov | |
5 * | |
6 * This file is part of FFmpeg. | |
7 * | |
8 * FFmpeg is free software; you can redistribute it and/or | |
9 * modify it under the terms of the GNU Lesser General Public | |
10 * License as published by the Free Software Foundation; either | |
11 * version 2.1 of the License, or (at your option) any later version. | |
12 * | |
13 * FFmpeg is distributed in the hope that it will be useful, | |
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
16 * Lesser General Public License for more details. | |
17 * | |
18 * You should have received a copy of the GNU Lesser General Public | |
19 * License along with FFmpeg; if not, write to the Free Software | |
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |
21 */ | |
22 | |
23 #include <inttypes.h> | |
24 | |
25 #include "avcodec.h" | |
26 #include "acelp_filters.h" | |
27 | |
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28 const int16_t ff_acelp_interp_filter[61] = |
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29 { /* (0.15) */ |
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30 29443, 28346, 25207, 20449, 14701, 8693, |
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31 3143, -1352, -4402, -5865, -5850, -4673, |
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32 -2783, -672, 1211, 2536, 3130, 2991, |
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33 2259, 1170, 0, -1001, -1652, -1868, |
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34 -1666, -1147, -464, 218, 756, 1060, |
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35 1099, 904, 550, 135, -245, -514, |
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36 -634, -602, -451, -231, 0, 191, |
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37 308, 340, 296, 198, 78, -36, |
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38 -120, -163, -165, -132, -79, -19, |
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39 34, 73, 91, 89, 70, 38, |
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40 0, |
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41 }; |
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42 |
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43 void ff_acelp_interpolate( |
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44 int16_t* out, |
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45 const int16_t* in, |
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46 const int16_t* filter_coeffs, |
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47 int precision, |
7649 | 48 int frac_pos, |
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49 int filter_length, |
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50 int length) |
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51 { |
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52 int n, i; |
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53 |
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54 assert(pitch_delay_frac >= 0 && pitch_delay_frac < precision); |
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55 |
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56 for(n=0; n<length; n++) |
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57 { |
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58 int idx = 0; |
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59 int v = 0x4000; |
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60 |
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61 for(i=0; i<filter_length;) |
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62 { |
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63 |
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64 /* The reference G.729 and AMR fixed point code performs clipping after |
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65 each of the two following accumulations. |
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66 Since clipping affects only the synthetic OVERFLOW test without |
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67 causing an int type overflow, it was moved outside the loop. */ |
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68 |
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69 /* R(x):=ac_v[-k+x] |
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70 v += R(n-i)*ff_acelp_interp_filter(t+6i) |
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71 v += R(n+i+1)*ff_acelp_interp_filter(6-t+6i) */ |
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72 |
7649 | 73 v += in[n + i] * filter_coeffs[idx + frac_pos]; |
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74 idx += precision; |
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75 i++; |
7649 | 76 v += in[n - i] * filter_coeffs[idx - frac_pos]; |
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77 } |
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78 if(av_clip_int16(v>>15) != (v>>15)) |
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79 av_log(NULL, AV_LOG_WARNING, "overflow that would need cliping in ff_acelp_interpolate()\n"); |
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80 out[n] = v >> 15; |
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81 } |
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82 } |
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83 |
6772 | 84 void ff_acelp_convolve_circ( |
85 int16_t* fc_out, | |
86 const int16_t* fc_in, | |
87 const int16_t* filter, | |
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88 int len) |
6772 | 89 { |
90 int i, k; | |
91 | |
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92 memset(fc_out, 0, len * sizeof(int16_t)); |
6772 | 93 |
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94 /* Since there are few pulses over an entire subframe (i.e. almost |
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95 all fc_in[i] are zero) it is faster to loop over fc_in first. */ |
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96 for(i=0; i<len; i++) |
6772 | 97 { |
98 if(fc_in[i]) | |
99 { | |
100 for(k=0; k<i; k++) | |
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101 fc_out[k] += (fc_in[i] * filter[len + k - i]) >> 15; |
6772 | 102 |
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103 for(k=i; k<len; k++) |
7652 | 104 fc_out[k] += (fc_in[i] * filter[ k - i]) >> 15; |
6772 | 105 } |
106 } | |
107 } | |
108 | |
109 int ff_acelp_lp_synthesis_filter( | |
110 int16_t *out, | |
111 const int16_t* filter_coeffs, | |
112 const int16_t* in, | |
113 int buffer_length, | |
114 int filter_length, | |
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115 int stop_on_overflow, |
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116 int rounder) |
6772 | 117 { |
118 int i,n; | |
119 | |
7211 | 120 // These two lines are to avoid a -1 subtraction in the main loop |
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121 filter_length++; |
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122 filter_coeffs--; |
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123 |
6772 | 124 for(n=0; n<buffer_length; n++) |
125 { | |
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126 int sum = rounder; |
6772 | 127 for(i=1; i<filter_length; i++) |
128 sum -= filter_coeffs[i] * out[n-i]; | |
129 | |
130 sum = (sum >> 12) + in[n]; | |
131 | |
132 if(sum + 0x8000 > 0xFFFFU) | |
133 { | |
134 if(stop_on_overflow) | |
135 return 1; | |
136 sum = (sum >> 31) ^ 32767; | |
137 } | |
138 out[n] = sum; | |
139 } | |
140 | |
141 return 0; | |
142 } | |
143 | |
144 void ff_acelp_high_pass_filter( | |
145 int16_t* out, | |
146 int hpf_f[2], | |
147 const int16_t* in, | |
148 int length) | |
149 { | |
150 int i; | |
151 int tmp; | |
152 | |
153 for(i=0; i<length; i++) | |
154 { | |
7695 | 155 tmp = (hpf_f[0]* 15836LL)>>13; |
156 tmp += (hpf_f[1]* -7667LL)>>13; | |
157 tmp += 7699 * (in[i] - 2*in[i-1] + in[i-2]); | |
6772 | 158 |
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159 /* With "+0x800" rounding, clipping is needed |
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160 for ALGTHM and SPEECH tests. */ |
7695 | 161 out[i] = av_clip_int16((tmp + 0x800) >> 12); |
6772 | 162 |
163 hpf_f[1] = hpf_f[0]; | |
164 hpf_f[0] = tmp; | |
165 } | |
166 } |