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Diffstat (limited to 'media/ffvpx/libavcodec/resample2.c')
-rw-r--r--media/ffvpx/libavcodec/resample2.c319
1 files changed, 0 insertions, 319 deletions
diff --git a/media/ffvpx/libavcodec/resample2.c b/media/ffvpx/libavcodec/resample2.c
deleted file mode 100644
index 56ae9f7229..0000000000
--- a/media/ffvpx/libavcodec/resample2.c
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@@ -1,319 +0,0 @@
-/*
- * audio resampling
- * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * audio resampling
- * @author Michael Niedermayer <michaelni@gmx.at>
- */
-
-#include "libavutil/avassert.h"
-#include "avcodec.h"
-#include "libavutil/common.h"
-
-#if FF_API_AVCODEC_RESAMPLE
-
-#ifndef CONFIG_RESAMPLE_HP
-#define FILTER_SHIFT 15
-
-typedef int16_t FELEM;
-typedef int32_t FELEM2;
-typedef int64_t FELEML;
-#define FELEM_MAX INT16_MAX
-#define FELEM_MIN INT16_MIN
-#define WINDOW_TYPE 9
-#elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
-#define FILTER_SHIFT 30
-
-#define FELEM int32_t
-#define FELEM2 int64_t
-#define FELEML int64_t
-#define FELEM_MAX INT32_MAX
-#define FELEM_MIN INT32_MIN
-#define WINDOW_TYPE 12
-#else
-#define FILTER_SHIFT 0
-
-typedef double FELEM;
-typedef double FELEM2;
-typedef double FELEML;
-#define WINDOW_TYPE 24
-#endif
-
-
-typedef struct AVResampleContext{
- const AVClass *av_class;
- FELEM *filter_bank;
- int filter_length;
- int ideal_dst_incr;
- int dst_incr;
- int index;
- int frac;
- int src_incr;
- int compensation_distance;
- int phase_shift;
- int phase_mask;
- int linear;
-}AVResampleContext;
-
-/**
- * 0th order modified bessel function of the first kind.
- */
-static double bessel(double x){
- double v=1;
- double lastv=0;
- double t=1;
- int i;
-
- x= x*x/4;
- for(i=1; v != lastv; i++){
- lastv=v;
- t *= x/(i*i);
- v += t;
- }
- return v;
-}
-
-/**
- * Build a polyphase filterbank.
- * @param factor resampling factor
- * @param scale wanted sum of coefficients for each filter
- * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
- * @return 0 on success, negative on error
- */
-static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
- int ph, i;
- double x, y, w;
- double *tab = av_malloc_array(tap_count, sizeof(*tab));
- const int center= (tap_count-1)/2;
-
- if (!tab)
- return AVERROR(ENOMEM);
-
- /* if upsampling, only need to interpolate, no filter */
- if (factor > 1.0)
- factor = 1.0;
-
- for(ph=0;ph<phase_count;ph++) {
- double norm = 0;
- for(i=0;i<tap_count;i++) {
- x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
- if (x == 0) y = 1.0;
- else y = sin(x) / x;
- switch(type){
- case 0:{
- const float d= -0.5; //first order derivative = -0.5
- x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
- if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
- else y= d*(-4 + 8*x - 5*x*x + x*x*x);
- break;}
- case 1:
- w = 2.0*x / (factor*tap_count) + M_PI;
- y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
- break;
- default:
- w = 2.0*x / (factor*tap_count*M_PI);
- y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
- break;
- }
-
- tab[i] = y;
- norm += y;
- }
-
- /* normalize so that an uniform color remains the same */
- for(i=0;i<tap_count;i++) {
-#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
- filter[ph * tap_count + i] = tab[i] / norm;
-#else
- filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
-#endif
- }
- }
-#if 0
- {
-#define LEN 1024
- int j,k;
- double sine[LEN + tap_count];
- double filtered[LEN];
- double maxff=-2, minff=2, maxsf=-2, minsf=2;
- for(i=0; i<LEN; i++){
- double ss=0, sf=0, ff=0;
- for(j=0; j<LEN+tap_count; j++)
- sine[j]= cos(i*j*M_PI/LEN);
- for(j=0; j<LEN; j++){
- double sum=0;
- ph=0;
- for(k=0; k<tap_count; k++)
- sum += filter[ph * tap_count + k] * sine[k+j];
- filtered[j]= sum / (1<<FILTER_SHIFT);
- ss+= sine[j + center] * sine[j + center];
- ff+= filtered[j] * filtered[j];
- sf+= sine[j + center] * filtered[j];
- }
- ss= sqrt(2*ss/LEN);
- ff= sqrt(2*ff/LEN);
- sf= 2*sf/LEN;
- maxff= FFMAX(maxff, ff);
- minff= FFMIN(minff, ff);
- maxsf= FFMAX(maxsf, sf);
- minsf= FFMIN(minsf, sf);
- if(i%11==0){
- 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);
- minff=minsf= 2;
- maxff=maxsf= -2;
- }
- }
- }
-#endif
-
- av_free(tab);
- return 0;
-}
-
-AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
- AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
- double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
- int phase_count= 1<<phase_shift;
-
- if (!c)
- return NULL;
-
- c->phase_shift= phase_shift;
- c->phase_mask= phase_count-1;
- c->linear= linear;
-
- c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
- c->filter_bank= av_mallocz_array(c->filter_length, (phase_count+1)*sizeof(FELEM));
- if (!c->filter_bank)
- goto error;
- if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
- goto error;
- memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
- c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
-
- if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))
- goto error;
- c->ideal_dst_incr= c->dst_incr;
-
- c->index= -phase_count*((c->filter_length-1)/2);
-
- return c;
-error:
- av_free(c->filter_bank);
- av_free(c);
- return NULL;
-}
-
-void av_resample_close(AVResampleContext *c){
- av_freep(&c->filter_bank);
- av_freep(&c);
-}
-
-void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
-// sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
- c->compensation_distance= compensation_distance;
- c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
-}
-
-int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
- int dst_index, i;
- int index= c->index;
- int frac= c->frac;
- int dst_incr_frac= c->dst_incr % c->src_incr;
- int dst_incr= c->dst_incr / c->src_incr;
- int compensation_distance= c->compensation_distance;
-
- if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
- int64_t index2= ((int64_t)index)<<32;
- int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
- dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
-
- for(dst_index=0; dst_index < dst_size; dst_index++){
- dst[dst_index] = src[index2>>32];
- index2 += incr;
- }
- index += dst_index * dst_incr;
- index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
- frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
- }else{
- for(dst_index=0; dst_index < dst_size; dst_index++){
- FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
- int sample_index= index >> c->phase_shift;
- FELEM2 val=0;
-
- if(sample_index < 0){
- for(i=0; i<c->filter_length; i++)
- val += src[FFABS(sample_index + i) % src_size] * filter[i];
- }else if(sample_index + c->filter_length > src_size){
- break;
- }else if(c->linear){
- FELEM2 v2=0;
- for(i=0; i<c->filter_length; i++){
- val += src[sample_index + i] * (FELEM2)filter[i];
- v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
- }
- val+=(v2-val)*(FELEML)frac / c->src_incr;
- }else{
- for(i=0; i<c->filter_length; i++){
- val += src[sample_index + i] * (FELEM2)filter[i];
- }
- }
-
-#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
- dst[dst_index] = av_clip_int16(lrintf(val));
-#else
- val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
- dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
-#endif
-
- frac += dst_incr_frac;
- index += dst_incr;
- if(frac >= c->src_incr){
- frac -= c->src_incr;
- index++;
- }
-
- if(dst_index + 1 == compensation_distance){
- compensation_distance= 0;
- dst_incr_frac= c->ideal_dst_incr % c->src_incr;
- dst_incr= c->ideal_dst_incr / c->src_incr;
- }
- }
- }
- *consumed= FFMAX(index, 0) >> c->phase_shift;
- if(index>=0) index &= c->phase_mask;
-
- if(compensation_distance){
- compensation_distance -= dst_index;
- av_assert2(compensation_distance > 0);
- }
- if(update_ctx){
- c->frac= frac;
- c->index= index;
- c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
- c->compensation_distance= compensation_distance;
- }
-
- return dst_index;
-}
-
-#endif