1 /*
2 * audio resampling
3 * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, see <http://www.gnu.org/licenses/>.
19 */
20
21 /**
22 * @file libavcodec/resample2.c
23 * audio resampling
24 * @author Michael Niedermayer <michaelni@gmx.at>
25 */
26
27 #include "avcodec.h"
28 #include "dsputil.h"
29
30 #ifndef CONFIG_RESAMPLE_HP
31 #define FILTER_SHIFT 15
32
33 #define FELEM int16_t
34 #define FELEM2 int32_t
35 #define FELEML int64_t
36 #define FELEM_MAX INT16_MAX
37 #define FELEM_MIN INT16_MIN
38 #define WINDOW_TYPE 9
39 #elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
40 #define FILTER_SHIFT 30
41
42 #define FELEM int32_t
43 #define FELEM2 int64_t
44 #define FELEML int64_t
45 #define FELEM_MAX INT32_MAX
46 #define FELEM_MIN INT32_MIN
47 #define WINDOW_TYPE 12
48 #else
49 #define FILTER_SHIFT 0
50
51 #define FELEM double
52 #define FELEM2 double
53 #define FELEML double
54 #define WINDOW_TYPE 24
55 #endif
56
57
58 typedef struct AVResampleContext{
59 FELEM *filter_bank;
60 int filter_length;
61 int ideal_dst_incr;
62 int dst_incr;
63 int index;
64 int frac;
65 int src_incr;
66 int compensation_distance;
67 int phase_shift;
68 int phase_mask;
69 int linear;
70 }AVResampleContext;
71
72 /**
73 * 0th order modified bessel function of the first kind.
74 */
bessel(double x)75 static double bessel(double x){
76 double v=1;
77 double t=1;
78 int i;
79
80 x= x*x/4;
81 for(i=1; i<50; i++){
82 t *= x/(i*i);
83 v += t;
84 }
85 return v;
86 }
87
88 /**
89 * builds a polyphase filterbank.
90 * @param factor resampling factor
91 * @param scale wanted sum of coefficients for each filter
92 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
93 */
av_build_filter(FELEM * filter,double factor,int tap_count,int phase_count,int scale,int type)94 void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
95 int ph, i;
96 double x, y, w, tab[tap_count];
97 const int center= (tap_count-1)/2;
98
99 /* if upsampling, only need to interpolate, no filter */
100 if (factor > 1.0)
101 factor = 1.0;
102
103 for(ph=0;ph<phase_count;ph++) {
104 double norm = 0;
105 for(i=0;i<tap_count;i++) {
106 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
107 if (x == 0) y = 1.0;
108 else y = sin(x) / x;
109 switch(type){
110 case 0:{
111 const float d= -0.5; //first order derivative = -0.5
112 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
113 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
114 else y= d*(-4 + 8*x - 5*x*x + x*x*x);
115 break;}
116 case 1:
117 w = 2.0*x / (factor*tap_count) + M_PI;
118 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
119 break;
120 default:
121 w = 2.0*x / (factor*tap_count*M_PI);
122 y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
123 break;
124 }
125
126 tab[i] = y;
127 norm += y;
128 }
129
130 /* normalize so that an uniform color remains the same */
131 for(i=0;i<tap_count;i++) {
132 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
133 filter[ph * tap_count + i] = tab[i] / norm;
134 #else
135 filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
136 #endif
137 }
138 }
139 #if 0
140 {
141 #define LEN 1024
142 int j,k;
143 double sine[LEN + tap_count];
144 double filtered[LEN];
145 double maxff=-2, minff=2, maxsf=-2, minsf=2;
146 for(i=0; i<LEN; i++){
147 double ss=0, sf=0, ff=0;
148 for(j=0; j<LEN+tap_count; j++)
149 sine[j]= cos(i*j*M_PI/LEN);
150 for(j=0; j<LEN; j++){
151 double sum=0;
152 ph=0;
153 for(k=0; k<tap_count; k++)
154 sum += filter[ph * tap_count + k] * sine[k+j];
155 filtered[j]= sum / (1<<FILTER_SHIFT);
156 ss+= sine[j + center] * sine[j + center];
157 ff+= filtered[j] * filtered[j];
158 sf+= sine[j + center] * filtered[j];
159 }
160 ss= sqrt(2*ss/LEN);
161 ff= sqrt(2*ff/LEN);
162 sf= 2*sf/LEN;
163 maxff= FFMAX(maxff, ff);
164 minff= FFMIN(minff, ff);
165 maxsf= FFMAX(maxsf, sf);
166 minsf= FFMIN(minsf, sf);
167 if(i%11==0){
168 av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf);
169 minff=minsf= 2;
170 maxff=maxsf= -2;
171 }
172 }
173 }
174 #endif
175 }
176
av_resample_init(int out_rate,int in_rate,int filter_size,int phase_shift,int linear,double cutoff)177 AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
178 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
179 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
180 int phase_count= 1<<phase_shift;
181
182 c->phase_shift= phase_shift;
183 c->phase_mask= phase_count-1;
184 c->linear= linear;
185
186 c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
187 c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
188 av_build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE);
189 memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
190 c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
191
192 c->src_incr= out_rate;
193 c->ideal_dst_incr= c->dst_incr= in_rate * phase_count;
194 c->index= -phase_count*((c->filter_length-1)/2);
195
196 return c;
197 }
198
av_resample_close(AVResampleContext * c)199 void av_resample_close(AVResampleContext *c){
200 av_freep(&c->filter_bank);
201 av_freep(&c);
202 }
203
av_resample_compensate(AVResampleContext * c,int sample_delta,int compensation_distance)204 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
205 // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
206 c->compensation_distance= compensation_distance;
207 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
208 }
209
av_resample(AVResampleContext * c,short * dst,short * src,int * consumed,int src_size,int dst_size,int update_ctx)210 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
211 int dst_index, i;
212 int index= c->index;
213 int frac= c->frac;
214 int dst_incr_frac= c->dst_incr % c->src_incr;
215 int dst_incr= c->dst_incr / c->src_incr;
216 int compensation_distance= c->compensation_distance;
217
218 if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
219 int64_t index2= ((int64_t)index)<<32;
220 int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
221 dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
222
223 for(dst_index=0; dst_index < dst_size; dst_index++){
224 dst[dst_index] = src[index2>>32];
225 index2 += incr;
226 }
227 frac += dst_index * dst_incr_frac;
228 index += dst_index * dst_incr;
229 index += frac / c->src_incr;
230 frac %= c->src_incr;
231 }else{
232 for(dst_index=0; dst_index < dst_size; dst_index++){
233 FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
234 int sample_index= index >> c->phase_shift;
235 FELEM2 val=0;
236
237 if(sample_index < 0){
238 for(i=0; i<c->filter_length; i++)
239 val += src[FFABS(sample_index + i) % src_size] * filter[i];
240 }else if(sample_index + c->filter_length > src_size){
241 break;
242 }else if(c->linear){
243 FELEM2 v2=0;
244 for(i=0; i<c->filter_length; i++){
245 val += src[sample_index + i] * (FELEM2)filter[i];
246 v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
247 }
248 val+=(v2-val)*(FELEML)frac / c->src_incr;
249 }else{
250 for(i=0; i<c->filter_length; i++){
251 val += src[sample_index + i] * (FELEM2)filter[i];
252 }
253 }
254
255 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
256 dst[dst_index] = av_clip_int16(lrintf(val));
257 #else
258 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
259 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
260 #endif
261
262 frac += dst_incr_frac;
263 index += dst_incr;
264 if(frac >= c->src_incr){
265 frac -= c->src_incr;
266 index++;
267 }
268
269 if(dst_index + 1 == compensation_distance){
270 compensation_distance= 0;
271 dst_incr_frac= c->ideal_dst_incr % c->src_incr;
272 dst_incr= c->ideal_dst_incr / c->src_incr;
273 }
274 }
275 }
276 *consumed= FFMAX(index, 0) >> c->phase_shift;
277 if(index>=0) index &= c->phase_mask;
278
279 if(compensation_distance){
280 compensation_distance -= dst_index;
281 assert(compensation_distance > 0);
282 }
283 if(update_ctx){
284 c->frac= frac;
285 c->index= index;
286 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
287 c->compensation_distance= compensation_distance;
288 }
289 #if 0
290 if(update_ctx && !c->compensation_distance){
291 #undef rand
292 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
293 av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
294 }
295 #endif
296
297 return dst_index;
298 }
299