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#include "common.h"
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#include "dsputil.h"
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#define PHASE_SHIFT 10
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#define PHASE_COUNT (1<<PHASE_SHIFT)
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#define PHASE_MASK (PHASE_COUNT-1)
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#define FILTER_SHIFT 15
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#define FELEM2 int32_t
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#define FELEM_MAX INT16_MAX
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#define FELEM_MIN INT16_MIN
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#define FILTER_SHIFT 22
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#define FELEM2 int64_t
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#define FELEM_MAX INT32_MAX
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#define FELEM_MIN INT32_MIN
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typedef struct AVResampleContext{
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int ideal_dst_incr;
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* @param scale wanted sum of coefficients for each filter
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* @param type 0->cubic, 1->blackman nuttall windowed sinc, 2->kaiser windowed sinc beta=16
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void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_count, int scale, int type){
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void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
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double x, y, w, tab[tap_count];
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const int center= (tap_count-1)/2;
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/* normalize so that an uniform color remains the same */
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for(i=0;i<tap_count;i++) {
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v = clip(lrintf(tab[i] * scale / norm + e), -32768, 32767);
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v = clip(lrintf(tab[i] * scale / norm + e), FELEM_MIN, FELEM_MAX);
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filter[ph * tap_count + i] = v;
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e += tab[i] * scale / norm - v;
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* initalizes a audio resampler.
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* note, if either rate is not a integer then simply scale both rates up so they are
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AVResampleContext *av_resample_init(int out_rate, int in_rate){
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AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
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AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
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double factor= FFMIN(out_rate / (double)in_rate, 1.0);
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memset(c, 0, sizeof(AVResampleContext));
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c->filter_length= ceil(16.0/factor);
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c->filter_bank= av_mallocz(c->filter_length*(PHASE_COUNT+1)*sizeof(short));
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av_build_filter(c->filter_bank, factor, c->filter_length, PHASE_COUNT, 1<<FILTER_SHIFT, 1);
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c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1)/2 + 1]= (1<<FILTER_SHIFT)-1;
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c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1)/2 + 2]= 1;
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double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
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int phase_count= 1<<phase_shift;
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c->phase_shift= phase_shift;
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c->phase_mask= phase_count-1;
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c->filter_length= FFMAX(ceil(filter_size/factor), 1);
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c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
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av_build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, 1);
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memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
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c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
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c->src_incr= out_rate;
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c->ideal_dst_incr= c->dst_incr= in_rate * PHASE_COUNT;
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c->index= -PHASE_COUNT*((c->filter_length-1)/2);
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c->ideal_dst_incr= c->dst_incr= in_rate * phase_count;
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c->index= -phase_count*((c->filter_length-1)/2);
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* Compensates samplerate/timestamp drift. The compensation is done by changing
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* the resampler parameters, so no audible clicks or similar distortions ocur
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* @param compensation_distance distance in output samples over which the compensation should be performed
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* @param sample_delta number of output samples which should be output less
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* example: av_resample_compensate(c, 10, 500)
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* here instead of 510 samples only 500 samples would be output
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* note, due to rounding the actual compensation might be slightly different,
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* especially if the compensation_distance is large and the in_rate used during init is small
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void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
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// sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
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c->compensation_distance= compensation_distance;
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int frac= c->frac;
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int dst_incr_frac= c->dst_incr % c->src_incr;
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int dst_incr= c->dst_incr / c->src_incr;
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if(c->compensation_distance && c->compensation_distance < dst_size)
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dst_size= c->compensation_distance;
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int compensation_distance= c->compensation_distance;
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if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
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int64_t index2= ((int64_t)index)<<32;
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int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
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dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
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for(dst_index=0; dst_index < dst_size; dst_index++){
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dst[dst_index] = src[index2>>32];
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frac += dst_index * dst_incr_frac;
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index += dst_index * dst_incr;
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index += frac / c->src_incr;
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for(dst_index=0; dst_index < dst_size; dst_index++){
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short *filter= c->filter_bank + c->filter_length*(index & PHASE_MASK);
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int sample_index= index >> PHASE_SHIFT;
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FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
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int sample_index= index >> c->phase_shift;
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if(sample_index < 0){
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for(i=0; i<c->filter_length; i++)
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val += src[ABS(sample_index + i) % src_size] * filter[i];
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}else if(sample_index + c->filter_length > src_size){
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int sub_phase= (frac<<12) / c->src_incr;
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int sub_phase= (frac<<8) / c->src_incr;
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for(i=0; i<c->filter_length; i++){
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int64_t coeff= filter[i]*(4096 - sub_phase) + filter[i + c->filter_length]*sub_phase;
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int64_t coeff= filter[i]*(256 - sub_phase) + filter[i + c->filter_length]*sub_phase;
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v += src[sample_index + i] * coeff;
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for(i=0; i<c->filter_length; i++){
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val += src[sample_index + i] * filter[i];
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val += src[sample_index + i] * (FELEM2)filter[i];
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val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
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frac -= c->src_incr;
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*consumed= FFMAX(index, 0) >> PHASE_SHIFT;
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index= FFMIN(index, 0);
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if(dst_index + 1 == compensation_distance){
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compensation_distance= 0;
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dst_incr_frac= c->ideal_dst_incr % c->src_incr;
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dst_incr= c->ideal_dst_incr / c->src_incr;
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*consumed= FFMAX(index, 0) >> c->phase_shift;
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if(index>=0) index &= c->phase_mask;
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if(compensation_distance){
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compensation_distance -= dst_index;
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assert(compensation_distance > 0);
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if(c->compensation_distance){
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c->compensation_distance -= dst_index;
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if(!c->compensation_distance)
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c->dst_incr= c->ideal_dst_incr;
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c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
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c->compensation_distance= compensation_distance;
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if(update_ctx && !c->compensation_distance){