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Diffstat (limited to 'media/ffvpx/libavcodec/resample2.c')
| -rw-r--r-- | media/ffvpx/libavcodec/resample2.c | 319 |
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 56ae9f722..000000000 --- a/media/ffvpx/libavcodec/resample2.c +++ /dev/null @@ -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 |