3 * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
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.
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.
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
24 * @author Michael Niedermayer <michaelni@gmx.at>
31 #define PHASE_SHIFT 10
32 #define PHASE_COUNT (1<<PHASE_SHIFT)
33 #define PHASE_MASK (PHASE_COUNT-1)
34 #define FILTER_SHIFT 15
36 typedef struct AVResampleContext{
44 int compensation_distance;
48 * 0th order modified bessel function of the first kind.
50 double bessel(double x){
57 v += pow(x*x/4, i)/(t*t);
63 * builds a polyphase filterbank.
64 * @param factor resampling factor
65 * @param scale wanted sum of coefficients for each filter
66 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2->kaiser windowed sinc beta=16
68 void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_count, int scale, int type){
70 double x, y, w, tab[tap_count];
71 const int center= (tap_count-1)/2;
73 /* if upsampling, only need to interpolate, no filter */
77 for(ph=0;ph<phase_count;ph++) {
80 for(i=0;i<tap_count;i++) {
81 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
86 const float d= -0.5; //first order derivative = -0.5
87 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
88 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
89 else y= d*(-4 + 8*x - 5*x*x + x*x*x);
92 w = 2.0*x / (factor*tap_count) + M_PI;
93 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
96 w = 2.0*x / (factor*tap_count*M_PI);
97 y *= bessel(16*sqrt(FFMAX(1-w*w, 0)));
105 /* normalize so that an uniform color remains the same */
106 for(i=0;i<tap_count;i++) {
107 v = clip(lrintf(tab[i] * scale / norm) + e, -32768, 32767);
108 filter[ph * tap_count + i] = v;
109 e += tab[i] * scale / norm - v;
115 * initalizes a audio resampler.
116 * note, if either rate is not a integer then simply scale both rates up so they are
118 AVResampleContext *av_resample_init(int out_rate, int in_rate){
119 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
120 double factor= FFMIN(out_rate / (double)in_rate, 1.0);
122 memset(c, 0, sizeof(AVResampleContext));
124 c->filter_length= ceil(16.0/factor);
125 c->filter_bank= av_mallocz(c->filter_length*(PHASE_COUNT+1)*sizeof(short));
126 av_build_filter(c->filter_bank, factor, c->filter_length, PHASE_COUNT, 1<<FILTER_SHIFT, 1);
127 c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1)/2 + 1]= (1<<FILTER_SHIFT)-1;
128 c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1)/2 + 2]= 1;
130 c->src_incr= out_rate;
131 c->ideal_dst_incr= c->dst_incr= in_rate * PHASE_COUNT;
132 c->index= -PHASE_COUNT*((c->filter_length-1)/2);
137 void av_resample_close(AVResampleContext *c){
138 av_freep(&c->filter_bank);
142 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
143 // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
144 c->compensation_distance= compensation_distance;
145 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
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
157 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
161 int dst_incr_frac= c->dst_incr % c->src_incr;
162 int dst_incr= c->dst_incr / c->src_incr;
164 if(c->compensation_distance && c->compensation_distance < dst_size)
165 dst_size= c->compensation_distance;
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;
172 if(sample_index < 0){
173 for(i=0; i<c->filter_length; i++)
174 val += src[ABS(sample_index + i) % src_size] * filter[i];
175 }else if(sample_index + c->filter_length > src_size){
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;
187 for(i=0; i<c->filter_length; i++){
188 val += src[sample_index + i] * filter[i];
193 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
194 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
196 frac += dst_incr_frac;
198 if(frac >= c->src_incr){
203 *consumed= FFMAX(index, 0) >> PHASE_SHIFT;
204 index= FFMIN(index, 0);
207 if(c->compensation_distance){
208 c->compensation_distance -= dst_index;
209 if(!c->compensation_distance)
210 c->dst_incr= c->ideal_dst_incr;
216 if(update_ctx && !c->compensation_distance){
218 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
219 av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);