Imported Debian version 2.5.3~trusty1
[deb_ffmpeg.git] / ffmpeg / libavcodec / resample2.c
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 "libavutil/avassert.h"
29 #include "avcodec.h"
30 #include "libavutil/common.h"
31
32 #if FF_API_AVCODEC_RESAMPLE
33
34 #ifndef CONFIG_RESAMPLE_HP
35 #define FILTER_SHIFT 15
36
37 #define FELEM int16_t
38 #define FELEM2 int32_t
39 #define FELEML int64_t
40 #define FELEM_MAX INT16_MAX
41 #define FELEM_MIN INT16_MIN
42 #define WINDOW_TYPE 9
43 #elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
44 #define FILTER_SHIFT 30
45
46 #define FELEM int32_t
47 #define FELEM2 int64_t
48 #define FELEML int64_t
49 #define FELEM_MAX INT32_MAX
50 #define FELEM_MIN INT32_MIN
51 #define WINDOW_TYPE 12
52 #else
53 #define FILTER_SHIFT 0
54
55 #define FELEM double
56 #define FELEM2 double
57 #define FELEML double
58 #define WINDOW_TYPE 24
59 #endif
60
61
62 typedef struct AVResampleContext{
63 const AVClass *av_class;
64 FELEM *filter_bank;
65 int filter_length;
66 int ideal_dst_incr;
67 int dst_incr;
68 int index;
69 int frac;
70 int src_incr;
71 int compensation_distance;
72 int phase_shift;
73 int phase_mask;
74 int linear;
75 }AVResampleContext;
76
77 /**
78 * 0th order modified bessel function of the first kind.
79 */
80 static double bessel(double x){
81 double v=1;
82 double lastv=0;
83 double t=1;
84 int i;
85
86 x= x*x/4;
87 for(i=1; v != lastv; i++){
88 lastv=v;
89 t *= x/(i*i);
90 v += t;
91 }
92 return v;
93 }
94
95 /**
96 * Build a polyphase filterbank.
97 * @param factor resampling factor
98 * @param scale wanted sum of coefficients for each filter
99 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
100 * @return 0 on success, negative on error
101 */
102 static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
103 int ph, i;
104 double x, y, w;
105 double *tab = av_malloc_array(tap_count, sizeof(*tab));
106 const int center= (tap_count-1)/2;
107
108 if (!tab)
109 return AVERROR(ENOMEM);
110
111 /* if upsampling, only need to interpolate, no filter */
112 if (factor > 1.0)
113 factor = 1.0;
114
115 for(ph=0;ph<phase_count;ph++) {
116 double norm = 0;
117 for(i=0;i<tap_count;i++) {
118 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
119 if (x == 0) y = 1.0;
120 else y = sin(x) / x;
121 switch(type){
122 case 0:{
123 const float d= -0.5; //first order derivative = -0.5
124 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
125 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
126 else y= d*(-4 + 8*x - 5*x*x + x*x*x);
127 break;}
128 case 1:
129 w = 2.0*x / (factor*tap_count) + M_PI;
130 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
131 break;
132 default:
133 w = 2.0*x / (factor*tap_count*M_PI);
134 y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
135 break;
136 }
137
138 tab[i] = y;
139 norm += y;
140 }
141
142 /* normalize so that an uniform color remains the same */
143 for(i=0;i<tap_count;i++) {
144 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
145 filter[ph * tap_count + i] = tab[i] / norm;
146 #else
147 filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
148 #endif
149 }
150 }
151 #if 0
152 {
153 #define LEN 1024
154 int j,k;
155 double sine[LEN + tap_count];
156 double filtered[LEN];
157 double maxff=-2, minff=2, maxsf=-2, minsf=2;
158 for(i=0; i<LEN; i++){
159 double ss=0, sf=0, ff=0;
160 for(j=0; j<LEN+tap_count; j++)
161 sine[j]= cos(i*j*M_PI/LEN);
162 for(j=0; j<LEN; j++){
163 double sum=0;
164 ph=0;
165 for(k=0; k<tap_count; k++)
166 sum += filter[ph * tap_count + k] * sine[k+j];
167 filtered[j]= sum / (1<<FILTER_SHIFT);
168 ss+= sine[j + center] * sine[j + center];
169 ff+= filtered[j] * filtered[j];
170 sf+= sine[j + center] * filtered[j];
171 }
172 ss= sqrt(2*ss/LEN);
173 ff= sqrt(2*ff/LEN);
174 sf= 2*sf/LEN;
175 maxff= FFMAX(maxff, ff);
176 minff= FFMIN(minff, ff);
177 maxsf= FFMAX(maxsf, sf);
178 minsf= FFMIN(minsf, sf);
179 if(i%11==0){
180 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);
181 minff=minsf= 2;
182 maxff=maxsf= -2;
183 }
184 }
185 }
186 #endif
187
188 av_free(tab);
189 return 0;
190 }
191
192 AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
193 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
194 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
195 int phase_count= 1<<phase_shift;
196
197 if (!c)
198 return NULL;
199
200 c->phase_shift= phase_shift;
201 c->phase_mask= phase_count-1;
202 c->linear= linear;
203
204 c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
205 c->filter_bank= av_mallocz_array(c->filter_length, (phase_count+1)*sizeof(FELEM));
206 if (!c->filter_bank)
207 goto error;
208 if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
209 goto error;
210 memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
211 c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
212
213 if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))
214 goto error;
215 c->ideal_dst_incr= c->dst_incr;
216
217 c->index= -phase_count*((c->filter_length-1)/2);
218
219 return c;
220 error:
221 av_free(c->filter_bank);
222 av_free(c);
223 return NULL;
224 }
225
226 void av_resample_close(AVResampleContext *c){
227 av_freep(&c->filter_bank);
228 av_freep(&c);
229 }
230
231 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
232 // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
233 c->compensation_distance= compensation_distance;
234 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
235 }
236
237 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
238 int dst_index, i;
239 int index= c->index;
240 int frac= c->frac;
241 int dst_incr_frac= c->dst_incr % c->src_incr;
242 int dst_incr= c->dst_incr / c->src_incr;
243 int compensation_distance= c->compensation_distance;
244
245 if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
246 int64_t index2= ((int64_t)index)<<32;
247 int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
248 dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
249
250 for(dst_index=0; dst_index < dst_size; dst_index++){
251 dst[dst_index] = src[index2>>32];
252 index2 += incr;
253 }
254 index += dst_index * dst_incr;
255 index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
256 frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
257 }else{
258 for(dst_index=0; dst_index < dst_size; dst_index++){
259 FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
260 int sample_index= index >> c->phase_shift;
261 FELEM2 val=0;
262
263 if(sample_index < 0){
264 for(i=0; i<c->filter_length; i++)
265 val += src[FFABS(sample_index + i) % src_size] * filter[i];
266 }else if(sample_index + c->filter_length > src_size){
267 break;
268 }else if(c->linear){
269 FELEM2 v2=0;
270 for(i=0; i<c->filter_length; i++){
271 val += src[sample_index + i] * (FELEM2)filter[i];
272 v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
273 }
274 val+=(v2-val)*(FELEML)frac / c->src_incr;
275 }else{
276 for(i=0; i<c->filter_length; i++){
277 val += src[sample_index + i] * (FELEM2)filter[i];
278 }
279 }
280
281 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
282 dst[dst_index] = av_clip_int16(lrintf(val));
283 #else
284 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
285 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
286 #endif
287
288 frac += dst_incr_frac;
289 index += dst_incr;
290 if(frac >= c->src_incr){
291 frac -= c->src_incr;
292 index++;
293 }
294
295 if(dst_index + 1 == compensation_distance){
296 compensation_distance= 0;
297 dst_incr_frac= c->ideal_dst_incr % c->src_incr;
298 dst_incr= c->ideal_dst_incr / c->src_incr;
299 }
300 }
301 }
302 *consumed= FFMAX(index, 0) >> c->phase_shift;
303 if(index>=0) index &= c->phase_mask;
304
305 if(compensation_distance){
306 compensation_distance -= dst_index;
307 av_assert2(compensation_distance > 0);
308 }
309 if(update_ctx){
310 c->frac= frac;
311 c->index= index;
312 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
313 c->compensation_distance= compensation_distance;
314 }
315
316 return dst_index;
317 }
318
319 #endif