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, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22 /**
23 * @file
24 * audio resampling
25 * @author Michael Niedermayer <michaelni@gmx.at>
26 */
27
28 #include "avcodec.h"
29 #include "dsputil.h"
30
31 #ifndef CONFIG_RESAMPLE_HP
32 #define FILTER_SHIFT 15
33
34 #define FELEM int16_t
35 #define FELEM2 int32_t
36 #define FELEML int64_t
37 #define FELEM_MAX INT16_MAX
38 #define FELEM_MIN INT16_MIN
39 #define WINDOW_TYPE 9
40 #elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
41 #define FILTER_SHIFT 30
42
43 #define FELEM int32_t
44 #define FELEM2 int64_t
45 #define FELEML int64_t
46 #define FELEM_MAX INT32_MAX
47 #define FELEM_MIN INT32_MIN
48 #define WINDOW_TYPE 12
49 #else
50 #define FILTER_SHIFT 0
51
52 #define FELEM double
53 #define FELEM2 double
54 #define FELEML double
55 #define WINDOW_TYPE 24
56 #endif
57
58
59 typedef struct AVResampleContext{
60 const AVClass *av_class;
61 FELEM *filter_bank;
62 int filter_length;
63 int ideal_dst_incr;
64 int dst_incr;
65 int index;
66 int frac;
67 int src_incr;
68 int compensation_distance;
69 int phase_shift;
70 int phase_mask;
71 int linear;
72 }AVResampleContext;
73
74 /**
75 * 0th order modified bessel function of the first kind.
76 */
bessel(double x)77 static double bessel(double x){
78 double v=1;
79 double lastv=0;
80 double t=1;
81 int i;
82
83 x= x*x/4;
84 for(i=1; v != lastv; i++){
85 lastv=v;
86 t *= x/(i*i);
87 v += t;
88 }
89 return v;
90 }
91
92 /**
93 * builds a polyphase filterbank.
94 * @param factor resampling factor
95 * @param scale wanted sum of coefficients for each filter
96 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
97 * @return 0 on success, negative on error
98 */
build_filter(FELEM * filter,double factor,int tap_count,int phase_count,int scale,int type)99 static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
100 int ph, i;
101 double x, y, w;
102 double *tab = av_malloc(tap_count * sizeof(*tab));
103 const int center= (tap_count-1)/2;
104
105 if (!tab)
106 return AVERROR(ENOMEM);
107
108 /* if upsampling, only need to interpolate, no filter */
109 if (factor > 1.0)
110 factor = 1.0;
111
112 for(ph=0;ph<phase_count;ph++) {
113 double norm = 0;
114 for(i=0;i<tap_count;i++) {
115 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
116 if (x == 0) y = 1.0;
117 else y = sin(x) / x;
118 switch(type){
119 case 0:{
120 const float d= -0.5; //first order derivative = -0.5
121 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
122 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
123 else y= d*(-4 + 8*x - 5*x*x + x*x*x);
124 break;}
125 case 1:
126 w = 2.0*x / (factor*tap_count) + M_PI;
127 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
128 break;
129 default:
130 w = 2.0*x / (factor*tap_count*M_PI);
131 y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
132 break;
133 }
134
135 tab[i] = y;
136 norm += y;
137 }
138
139 /* normalize so that an uniform color remains the same */
140 for(i=0;i<tap_count;i++) {
141 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
142 filter[ph * tap_count + i] = tab[i] / norm;
143 #else
144 filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
145 #endif
146 }
147 }
148 #if 0
149 {
150 #define LEN 1024
151 int j,k;
152 double sine[LEN + tap_count];
153 double filtered[LEN];
154 double maxff=-2, minff=2, maxsf=-2, minsf=2;
155 for(i=0; i<LEN; i++){
156 double ss=0, sf=0, ff=0;
157 for(j=0; j<LEN+tap_count; j++)
158 sine[j]= cos(i*j*M_PI/LEN);
159 for(j=0; j<LEN; j++){
160 double sum=0;
161 ph=0;
162 for(k=0; k<tap_count; k++)
163 sum += filter[ph * tap_count + k] * sine[k+j];
164 filtered[j]= sum / (1<<FILTER_SHIFT);
165 ss+= sine[j + center] * sine[j + center];
166 ff+= filtered[j] * filtered[j];
167 sf+= sine[j + center] * filtered[j];
168 }
169 ss= sqrt(2*ss/LEN);
170 ff= sqrt(2*ff/LEN);
171 sf= 2*sf/LEN;
172 maxff= FFMAX(maxff, ff);
173 minff= FFMIN(minff, ff);
174 maxsf= FFMAX(maxsf, sf);
175 minsf= FFMIN(minsf, sf);
176 if(i%11==0){
177 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);
178 minff=minsf= 2;
179 maxff=maxsf= -2;
180 }
181 }
182 }
183 #endif
184
185 av_free(tab);
186 return 0;
187 }
188
av_resample_init(int out_rate,int in_rate,int filter_size,int phase_shift,int linear,double cutoff)189 AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
190 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
191 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
192 int phase_count= 1<<phase_shift;
193
194 if (!c)
195 return NULL;
196
197 c->phase_shift= phase_shift;
198 c->phase_mask= phase_count-1;
199 c->linear= linear;
200
201 c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
202 c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
203 if (!c->filter_bank)
204 goto error;
205 if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
206 goto error;
207 memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
208 c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
209
210 c->src_incr= out_rate;
211 c->ideal_dst_incr= c->dst_incr= in_rate * phase_count;
212 c->index= -phase_count*((c->filter_length-1)/2);
213
214 return c;
215 error:
216 av_free(c->filter_bank);
217 av_free(c);
218 return NULL;
219 }
220
av_resample_close(AVResampleContext * c)221 void av_resample_close(AVResampleContext *c){
222 av_freep(&c->filter_bank);
223 av_freep(&c);
224 }
225
av_resample_compensate(AVResampleContext * c,int sample_delta,int compensation_distance)226 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
227 // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
228 c->compensation_distance= compensation_distance;
229 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
230 }
231
av_resample(AVResampleContext * c,short * dst,short * src,int * consumed,int src_size,int dst_size,int update_ctx)232 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
233 int dst_index, i;
234 int index= c->index;
235 int frac= c->frac;
236 int dst_incr_frac= c->dst_incr % c->src_incr;
237 int dst_incr= c->dst_incr / c->src_incr;
238 int compensation_distance= c->compensation_distance;
239
240 if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
241 int64_t index2= ((int64_t)index)<<32;
242 int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
243 dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
244
245 for(dst_index=0; dst_index < dst_size; dst_index++){
246 dst[dst_index] = src[index2>>32];
247 index2 += incr;
248 }
249 frac += dst_index * dst_incr_frac;
250 index += dst_index * dst_incr;
251 index += frac / c->src_incr;
252 frac %= c->src_incr;
253 }else{
254 for(dst_index=0; dst_index < dst_size; dst_index++){
255 FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
256 int sample_index= index >> c->phase_shift;
257 FELEM2 val=0;
258
259 if(sample_index < 0){
260 for(i=0; i<c->filter_length; i++)
261 val += src[FFABS(sample_index + i) % src_size] * filter[i];
262 }else if(sample_index + c->filter_length > src_size){
263 break;
264 }else if(c->linear){
265 FELEM2 v2=0;
266 for(i=0; i<c->filter_length; i++){
267 val += src[sample_index + i] * (FELEM2)filter[i];
268 v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
269 }
270 val+=(v2-val)*(FELEML)frac / c->src_incr;
271 }else{
272 for(i=0; i<c->filter_length; i++){
273 val += src[sample_index + i] * (FELEM2)filter[i];
274 }
275 }
276
277 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
278 dst[dst_index] = av_clip_int16(lrintf(val));
279 #else
280 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
281 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
282 #endif
283
284 frac += dst_incr_frac;
285 index += dst_incr;
286 if(frac >= c->src_incr){
287 frac -= c->src_incr;
288 index++;
289 }
290
291 if(dst_index + 1 == compensation_distance){
292 compensation_distance= 0;
293 dst_incr_frac= c->ideal_dst_incr % c->src_incr;
294 dst_incr= c->ideal_dst_incr / c->src_incr;
295 }
296 }
297 }
298 *consumed= FFMAX(index, 0) >> c->phase_shift;
299 if(index>=0) index &= c->phase_mask;
300
301 if(compensation_distance){
302 compensation_distance -= dst_index;
303 assert(compensation_distance > 0);
304 }
305 if(update_ctx){
306 c->frac= frac;
307 c->index= index;
308 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
309 c->compensation_distance= compensation_distance;
310 }
311 #if 0
312 if(update_ctx && !c->compensation_distance){
313 #undef rand
314 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
315 av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
316 }
317 #endif
318
319 return dst_index;
320 }
321