Mercurial > libavcodec.hg
comparison resample2.c @ 2082:3dc9bbe1b152 libavcodec
polyphase kaiser windowed sinc and blackman nuttall windowed sinc audio resample filters
author | michael |
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date | Thu, 17 Jun 2004 15:43:23 +0000 |
parents | |
children | 76cdbe832239 |
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2081:d3015863f745 | 2082:3dc9bbe1b152 |
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1 /* | |
2 * audio resampling | |
3 * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> | |
4 * | |
5 * This library is free software; you can redistribute it and/or | |
6 * modify it under the terms of the GNU Lesser General Public | |
7 * License as published by the Free Software Foundation; either | |
8 * version 2 of the License, or (at your option) any later version. | |
9 * | |
10 * This library is distributed in the hope that it will be useful, | |
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
13 * Lesser General Public License for more details. | |
14 * | |
15 * You should have received a copy of the GNU Lesser General Public | |
16 * License along with this library; if not, write to the Free Software | |
17 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA | |
18 * | |
19 */ | |
20 | |
21 /** | |
22 * @file resample2.c | |
23 * audio resampling | |
24 * @author Michael Niedermayer <michaelni@gmx.at> | |
25 */ | |
26 | |
27 #include "avcodec.h" | |
28 #include "common.h" | |
29 | |
30 #define PHASE_SHIFT 10 | |
31 #define PHASE_COUNT (1<<PHASE_SHIFT) | |
32 #define PHASE_MASK (PHASE_COUNT-1) | |
33 #define FILTER_SHIFT 15 | |
34 | |
35 typedef struct AVResampleContext{ | |
36 short *filter_bank; | |
37 int filter_length; | |
38 int ideal_dst_incr; | |
39 int dst_incr; | |
40 int index; | |
41 int frac; | |
42 int src_incr; | |
43 int compensation_distance; | |
44 }AVResampleContext; | |
45 | |
46 /** | |
47 * 0th order modified bessel function of the first kind. | |
48 */ | |
49 double bessel(double x){ | |
50 double v=1; | |
51 double t=1; | |
52 int i; | |
53 | |
54 for(i=1; i<50; i++){ | |
55 t *= i; | |
56 v += pow(x*x/4, i)/(t*t); | |
57 } | |
58 return v; | |
59 } | |
60 | |
61 /** | |
62 * builds a polyphase filterbank. | |
63 * @param factor resampling factor | |
64 * @param scale wanted sum of coefficients for each filter | |
65 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2->kaiser windowed sinc beta=16 | |
66 */ | |
67 void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_count, int scale, int type){ | |
68 int ph, i, v; | |
69 double x, y, w, tab[tap_count]; | |
70 const int center= (tap_count-1)/2; | |
71 | |
72 /* if upsampling, only need to interpolate, no filter */ | |
73 if (factor > 1.0) | |
74 factor = 1.0; | |
75 | |
76 for(ph=0;ph<phase_count;ph++) { | |
77 double norm = 0; | |
78 double e= 0; | |
79 for(i=0;i<tap_count;i++) { | |
80 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor; | |
81 if (x == 0) y = 1.0; | |
82 else y = sin(x) / x; | |
83 switch(type){ | |
84 case 0:{ | |
85 const float d= -0.5; //first order derivative = -0.5 | |
86 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor); | |
87 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x); | |
88 else y= d*(-4 + 8*x - 5*x*x + x*x*x); | |
89 break;} | |
90 case 1: | |
91 w = 2.0*x / (factor*tap_count) + M_PI; | |
92 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w); | |
93 break; | |
94 case 2: | |
95 w = 2.0*x / (factor*tap_count*M_PI); | |
96 y *= bessel(16*sqrt(FFMAX(1-w*w, 0))) / bessel(16); | |
97 break; | |
98 } | |
99 | |
100 tab[i] = y; | |
101 norm += y; | |
102 } | |
103 | |
104 /* normalize so that an uniform color remains the same */ | |
105 for(i=0;i<tap_count;i++) { | |
106 v = clip(lrintf(tab[i] * scale / norm) + e, -32768, 32767); | |
107 filter[ph * tap_count + i] = v; | |
108 e += tab[i] * scale / norm - v; | |
109 } | |
110 } | |
111 } | |
112 | |
113 /** | |
114 * initalizes a audio resampler. | |
115 * note, if either rate is not a integer then simply scale both rates up so they are | |
116 */ | |
117 AVResampleContext *av_resample_init(int out_rate, int in_rate){ | |
118 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext)); | |
119 double factor= FFMIN(out_rate / (double)in_rate, 1.0); | |
120 | |
121 memset(c, 0, sizeof(AVResampleContext)); | |
122 | |
123 c->filter_length= ceil(16.0/factor); | |
124 c->filter_bank= av_mallocz(c->filter_length*(PHASE_COUNT+1)*sizeof(short)); | |
125 av_build_filter(c->filter_bank, factor, c->filter_length, PHASE_COUNT, 1<<FILTER_SHIFT, 1); | |
126 c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1) + 1]= (1<<FILTER_SHIFT)-1; | |
127 c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1) + 2]= 1; | |
128 | |
129 c->src_incr= out_rate; | |
130 c->ideal_dst_incr= c->dst_incr= in_rate * PHASE_COUNT; | |
131 c->index= -PHASE_COUNT*((c->filter_length-1)/2); | |
132 | |
133 return c; | |
134 } | |
135 | |
136 void av_resample_close(AVResampleContext *c){ | |
137 av_freep(&c->filter_bank); | |
138 av_freep(&c); | |
139 } | |
140 | |
141 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){ | |
142 assert(!c->compensation_distance); //FIXME | |
143 | |
144 c->compensation_distance= compensation_distance; | |
145 c->dst_incr-= c->ideal_dst_incr * sample_delta / compensation_distance; | |
146 } | |
147 | |
148 /** | |
149 * resamples. | |
150 * @param src an array of unconsumed samples | |
151 * @param consumed the number of samples of src which have been consumed are returned here | |
152 * @param src_size the number of unconsumed samples available | |
153 * @param dst_size the amount of space in samples available in dst | |
154 * @param update_ctx if this is 0 then the context wont be modified, that way several channels can be resampled with the same context | |
155 * @return the number of samples written in dst or -1 if an error occured | |
156 */ | |
157 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){ | |
158 int dst_index, i; | |
159 int index= c->index; | |
160 int frac= c->frac; | |
161 int dst_incr_frac= c->dst_incr % c->src_incr; | |
162 int dst_incr= c->dst_incr / c->src_incr; | |
163 | |
164 if(c->compensation_distance && c->compensation_distance < dst_size) | |
165 dst_size= c->compensation_distance; | |
166 | |
167 for(dst_index=0; dst_index < dst_size; dst_index++){ | |
168 short *filter= c->filter_bank + c->filter_length*(index & PHASE_MASK); | |
169 int sample_index= index >> PHASE_SHIFT; | |
170 int val=0; | |
171 | |
172 if(sample_index < 0){ | |
173 for(i=0; i<c->filter_length; i++) | |
174 val += src[ABS(sample_index + i)] * filter[i]; | |
175 }else if(sample_index + c->filter_length > src_size){ | |
176 break; | |
177 }else{ | |
178 #if 0 | |
179 int64_t v=0; | |
180 int sub_phase= (frac<<12) / c->src_incr; | |
181 for(i=0; i<c->filter_length; i++){ | |
182 int64_t coeff= filter[i]*(4096 - sub_phase) + filter[i + c->filter_length]*sub_phase; | |
183 v += src[sample_index + i] * coeff; | |
184 } | |
185 val= v>>12; | |
186 #else | |
187 for(i=0; i<c->filter_length; i++){ | |
188 val += src[sample_index + i] * filter[i]; | |
189 } | |
190 #endif | |
191 } | |
192 | |
193 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT; | |
194 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val; | |
195 | |
196 frac += dst_incr_frac; | |
197 index += dst_incr; | |
198 if(frac >= c->src_incr){ | |
199 frac -= c->src_incr; | |
200 index++; | |
201 } | |
202 } | |
203 if(update_ctx){ | |
204 if(c->compensation_distance){ | |
205 c->compensation_distance -= index; | |
206 if(!c->compensation_distance) | |
207 c->dst_incr= c->ideal_dst_incr; | |
208 } | |
209 c->frac= frac; | |
210 c->index=0; | |
211 } | |
212 *consumed= index >> PHASE_SHIFT; | |
213 return dst_index; | |
214 } |