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Diffstat (limited to 'media/ffvpx/libavcodec/resample2.c')
| -rw-r--r-- | media/ffvpx/libavcodec/resample2.c | 319 |
1 files changed, 319 insertions, 0 deletions
diff --git a/media/ffvpx/libavcodec/resample2.c b/media/ffvpx/libavcodec/resample2.c new file mode 100644 index 000000000..56ae9f722 --- /dev/null +++ b/media/ffvpx/libavcodec/resample2.c @@ -0,0 +1,319 @@ +/* + * 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 |