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https://github.com/pineappleEA/pineapple-src.git
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623 lines
22 KiB
C
Executable File
623 lines
22 KiB
C
Executable File
/*
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* audio resampling
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* Copyright (c) 2004-2012 Michael Niedermayer <michaelni@gmx.at>
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* bessel function: Copyright (c) 2006 Xiaogang Zhang
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* audio resampling
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* @author Michael Niedermayer <michaelni@gmx.at>
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*/
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#include "libavutil/avassert.h"
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#include "resample.h"
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static inline double eval_poly(const double *coeff, int size, double x) {
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double sum = coeff[size-1];
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int i;
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for (i = size-2; i >= 0; --i) {
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sum *= x;
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sum += coeff[i];
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}
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return sum;
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}
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/**
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* 0th order modified bessel function of the first kind.
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* Algorithm taken from the Boost project, source:
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* https://searchcode.com/codesearch/view/14918379/
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* Use, modification and distribution are subject to the
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* Boost Software License, Version 1.0 (see notice below).
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* Boost Software License - Version 1.0 - August 17th, 2003
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Permission is hereby granted, free of charge, to any person or organization
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obtaining a copy of the software and accompanying documentation covered by
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this license (the "Software") to use, reproduce, display, distribute,
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execute, and transmit the Software, and to prepare derivative works of the
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Software, and to permit third-parties to whom the Software is furnished to
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do so, all subject to the following:
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The copyright notices in the Software and this entire statement, including
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the above license grant, this restriction and the following disclaimer,
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must be included in all copies of the Software, in whole or in part, and
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all derivative works of the Software, unless such copies or derivative
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works are solely in the form of machine-executable object code generated by
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a source language processor.
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THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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FITNESS FOR A PARTICULAR PURPOSE, TITLE AND NON-INFRINGEMENT. IN NO EVENT
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SHALL THE COPYRIGHT HOLDERS OR ANYONE DISTRIBUTING THE SOFTWARE BE LIABLE
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FOR ANY DAMAGES OR OTHER LIABILITY, WHETHER IN CONTRACT, TORT OR OTHERWISE,
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ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
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DEALINGS IN THE SOFTWARE.
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*/
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static double bessel(double x) {
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// Modified Bessel function of the first kind of order zero
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// minimax rational approximations on intervals, see
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// Blair and Edwards, Chalk River Report AECL-4928, 1974
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static const double p1[] = {
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-2.2335582639474375249e+15,
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-5.5050369673018427753e+14,
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-3.2940087627407749166e+13,
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-8.4925101247114157499e+11,
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-1.1912746104985237192e+10,
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-1.0313066708737980747e+08,
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-5.9545626019847898221e+05,
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-2.4125195876041896775e+03,
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-7.0935347449210549190e+00,
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-1.5453977791786851041e-02,
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-2.5172644670688975051e-05,
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-3.0517226450451067446e-08,
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-2.6843448573468483278e-11,
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-1.5982226675653184646e-14,
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-5.2487866627945699800e-18,
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};
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static const double q1[] = {
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-2.2335582639474375245e+15,
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7.8858692566751002988e+12,
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-1.2207067397808979846e+10,
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1.0377081058062166144e+07,
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-4.8527560179962773045e+03,
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1.0,
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};
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static const double p2[] = {
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-2.2210262233306573296e-04,
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1.3067392038106924055e-02,
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-4.4700805721174453923e-01,
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5.5674518371240761397e+00,
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-2.3517945679239481621e+01,
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3.1611322818701131207e+01,
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-9.6090021968656180000e+00,
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};
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static const double q2[] = {
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-5.5194330231005480228e-04,
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3.2547697594819615062e-02,
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-1.1151759188741312645e+00,
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1.3982595353892851542e+01,
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-6.0228002066743340583e+01,
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8.5539563258012929600e+01,
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-3.1446690275135491500e+01,
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1.0,
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};
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double y, r, factor;
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if (x == 0)
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return 1.0;
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x = fabs(x);
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if (x <= 15) {
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y = x * x;
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return eval_poly(p1, FF_ARRAY_ELEMS(p1), y) / eval_poly(q1, FF_ARRAY_ELEMS(q1), y);
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}
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else {
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y = 1 / x - 1.0 / 15;
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r = eval_poly(p2, FF_ARRAY_ELEMS(p2), y) / eval_poly(q2, FF_ARRAY_ELEMS(q2), y);
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factor = exp(x) / sqrt(x);
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return factor * r;
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}
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}
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/**
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* builds a polyphase filterbank.
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* @param factor resampling factor
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* @param scale wanted sum of coefficients for each filter
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* @param filter_type filter type
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* @param kaiser_beta kaiser window beta
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* @return 0 on success, negative on error
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*/
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static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int alloc, int phase_count, int scale,
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int filter_type, double kaiser_beta){
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int ph, i;
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int ph_nb = phase_count % 2 ? phase_count : phase_count / 2 + 1;
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double x, y, w, t, s;
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double *tab = av_malloc_array(tap_count+1, sizeof(*tab));
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double *sin_lut = av_malloc_array(ph_nb, sizeof(*sin_lut));
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const int center= (tap_count-1)/2;
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double norm = 0;
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int ret = AVERROR(ENOMEM);
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if (!tab || !sin_lut)
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goto fail;
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av_assert0(tap_count == 1 || tap_count % 2 == 0);
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/* if upsampling, only need to interpolate, no filter */
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if (factor > 1.0)
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factor = 1.0;
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if (factor == 1.0) {
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for (ph = 0; ph < ph_nb; ph++)
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sin_lut[ph] = sin(M_PI * ph / phase_count) * (center & 1 ? 1 : -1);
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}
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for(ph = 0; ph < ph_nb; ph++) {
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s = sin_lut[ph];
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for(i=0;i<tap_count;i++) {
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x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
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if (x == 0) y = 1.0;
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else if (factor == 1.0)
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y = s / x;
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else
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y = sin(x) / x;
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switch(filter_type){
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case SWR_FILTER_TYPE_CUBIC:{
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const float d= -0.5; //first order derivative = -0.5
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x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
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if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
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else y= d*(-4 + 8*x - 5*x*x + x*x*x);
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break;}
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case SWR_FILTER_TYPE_BLACKMAN_NUTTALL:
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w = 2.0*x / (factor*tap_count);
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t = -cos(w);
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y *= 0.3635819 - 0.4891775 * t + 0.1365995 * (2*t*t-1) - 0.0106411 * (4*t*t*t - 3*t);
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break;
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case SWR_FILTER_TYPE_KAISER:
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w = 2.0*x / (factor*tap_count*M_PI);
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y *= bessel(kaiser_beta*sqrt(FFMAX(1-w*w, 0)));
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break;
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default:
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av_assert0(0);
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}
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tab[i] = y;
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s = -s;
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if (!ph)
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norm += y;
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}
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/* normalize so that an uniform color remains the same */
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switch(c->format){
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case AV_SAMPLE_FMT_S16P:
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for(i=0;i<tap_count;i++)
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((int16_t*)filter)[ph * alloc + i] = av_clip_int16(lrintf(tab[i] * scale / norm));
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if (phase_count % 2) break;
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for (i = 0; i < tap_count; i++)
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((int16_t*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((int16_t*)filter)[ph * alloc + i];
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break;
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case AV_SAMPLE_FMT_S32P:
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for(i=0;i<tap_count;i++)
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((int32_t*)filter)[ph * alloc + i] = av_clipl_int32(llrint(tab[i] * scale / norm));
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if (phase_count % 2) break;
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for (i = 0; i < tap_count; i++)
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((int32_t*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((int32_t*)filter)[ph * alloc + i];
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break;
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case AV_SAMPLE_FMT_FLTP:
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for(i=0;i<tap_count;i++)
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((float*)filter)[ph * alloc + i] = tab[i] * scale / norm;
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if (phase_count % 2) break;
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for (i = 0; i < tap_count; i++)
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((float*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((float*)filter)[ph * alloc + i];
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break;
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case AV_SAMPLE_FMT_DBLP:
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for(i=0;i<tap_count;i++)
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((double*)filter)[ph * alloc + i] = tab[i] * scale / norm;
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if (phase_count % 2) break;
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for (i = 0; i < tap_count; i++)
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((double*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((double*)filter)[ph * alloc + i];
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break;
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}
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}
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#if 0
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{
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#define LEN 1024
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int j,k;
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double sine[LEN + tap_count];
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double filtered[LEN];
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double maxff=-2, minff=2, maxsf=-2, minsf=2;
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for(i=0; i<LEN; i++){
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double ss=0, sf=0, ff=0;
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for(j=0; j<LEN+tap_count; j++)
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sine[j]= cos(i*j*M_PI/LEN);
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for(j=0; j<LEN; j++){
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double sum=0;
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ph=0;
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for(k=0; k<tap_count; k++)
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sum += filter[ph * tap_count + k] * sine[k+j];
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filtered[j]= sum / (1<<FILTER_SHIFT);
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ss+= sine[j + center] * sine[j + center];
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ff+= filtered[j] * filtered[j];
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sf+= sine[j + center] * filtered[j];
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}
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ss= sqrt(2*ss/LEN);
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ff= sqrt(2*ff/LEN);
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sf= 2*sf/LEN;
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maxff= FFMAX(maxff, ff);
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minff= FFMIN(minff, ff);
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maxsf= FFMAX(maxsf, sf);
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minsf= FFMIN(minsf, sf);
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if(i%11==0){
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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);
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minff=minsf= 2;
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maxff=maxsf= -2;
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}
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}
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}
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#endif
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ret = 0;
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fail:
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av_free(tab);
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av_free(sin_lut);
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return ret;
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}
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static void resample_free(ResampleContext **cc){
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ResampleContext *c = *cc;
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if(!c)
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return;
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av_freep(&c->filter_bank);
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av_freep(cc);
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}
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static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
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double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, double kaiser_beta,
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double precision, int cheby, int exact_rational)
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{
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double cutoff = cutoff0? cutoff0 : 0.97;
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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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int phase_count_compensation = phase_count;
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int filter_length = FFMAX((int)ceil(filter_size/factor), 1);
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if (filter_length > 1)
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filter_length = FFALIGN(filter_length, 2);
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if (exact_rational) {
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int phase_count_exact, phase_count_exact_den;
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av_reduce(&phase_count_exact, &phase_count_exact_den, out_rate, in_rate, INT_MAX);
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if (phase_count_exact <= phase_count) {
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phase_count_compensation = phase_count_exact * (phase_count / phase_count_exact);
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phase_count = phase_count_exact;
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}
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}
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if (!c || c->phase_count != phase_count || c->linear!=linear || c->factor != factor
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|| c->filter_length != filter_length || c->format != format
|
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|| c->filter_type != filter_type || c->kaiser_beta != kaiser_beta) {
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resample_free(&c);
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c = av_mallocz(sizeof(*c));
|
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if (!c)
|
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return NULL;
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c->format= format;
|
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|
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c->felem_size= av_get_bytes_per_sample(c->format);
|
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|
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switch(c->format){
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case AV_SAMPLE_FMT_S16P:
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c->filter_shift = 15;
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break;
|
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case AV_SAMPLE_FMT_S32P:
|
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c->filter_shift = 30;
|
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break;
|
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case AV_SAMPLE_FMT_FLTP:
|
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case AV_SAMPLE_FMT_DBLP:
|
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c->filter_shift = 0;
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break;
|
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default:
|
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av_log(NULL, AV_LOG_ERROR, "Unsupported sample format\n");
|
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av_assert0(0);
|
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}
|
|
|
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if (filter_size/factor > INT32_MAX/256) {
|
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av_log(NULL, AV_LOG_ERROR, "Filter length too large\n");
|
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goto error;
|
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}
|
|
|
|
c->phase_count = phase_count;
|
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c->linear = linear;
|
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c->factor = factor;
|
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c->filter_length = filter_length;
|
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c->filter_alloc = FFALIGN(c->filter_length, 8);
|
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c->filter_bank = av_calloc(c->filter_alloc, (phase_count+1)*c->felem_size);
|
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c->filter_type = filter_type;
|
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c->kaiser_beta = kaiser_beta;
|
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c->phase_count_compensation = phase_count_compensation;
|
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if (!c->filter_bank)
|
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goto error;
|
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if (build_filter(c, (void*)c->filter_bank, factor, c->filter_length, c->filter_alloc, phase_count, 1<<c->filter_shift, filter_type, kaiser_beta))
|
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goto error;
|
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memcpy(c->filter_bank + (c->filter_alloc*phase_count+1)*c->felem_size, c->filter_bank, (c->filter_alloc-1)*c->felem_size);
|
|
memcpy(c->filter_bank + (c->filter_alloc*phase_count )*c->felem_size, c->filter_bank + (c->filter_alloc - 1)*c->felem_size, c->felem_size);
|
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}
|
|
|
|
c->compensation_distance= 0;
|
|
if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))
|
|
goto error;
|
|
while (c->dst_incr < (1<<20) && c->src_incr < (1<<20)) {
|
|
c->dst_incr *= 2;
|
|
c->src_incr *= 2;
|
|
}
|
|
c->ideal_dst_incr = c->dst_incr;
|
|
c->dst_incr_div = c->dst_incr / c->src_incr;
|
|
c->dst_incr_mod = c->dst_incr % c->src_incr;
|
|
|
|
c->index= -phase_count*((c->filter_length-1)/2);
|
|
c->frac= 0;
|
|
|
|
swri_resample_dsp_init(c);
|
|
|
|
return c;
|
|
error:
|
|
av_freep(&c->filter_bank);
|
|
av_free(c);
|
|
return NULL;
|
|
}
|
|
|
|
static int rebuild_filter_bank_with_compensation(ResampleContext *c)
|
|
{
|
|
uint8_t *new_filter_bank;
|
|
int new_src_incr, new_dst_incr;
|
|
int phase_count = c->phase_count_compensation;
|
|
int ret;
|
|
|
|
if (phase_count == c->phase_count)
|
|
return 0;
|
|
|
|
av_assert0(!c->frac && !c->dst_incr_mod);
|
|
|
|
new_filter_bank = av_calloc(c->filter_alloc, (phase_count + 1) * c->felem_size);
|
|
if (!new_filter_bank)
|
|
return AVERROR(ENOMEM);
|
|
|
|
ret = build_filter(c, new_filter_bank, c->factor, c->filter_length, c->filter_alloc,
|
|
phase_count, 1 << c->filter_shift, c->filter_type, c->kaiser_beta);
|
|
if (ret < 0) {
|
|
av_freep(&new_filter_bank);
|
|
return ret;
|
|
}
|
|
memcpy(new_filter_bank + (c->filter_alloc*phase_count+1)*c->felem_size, new_filter_bank, (c->filter_alloc-1)*c->felem_size);
|
|
memcpy(new_filter_bank + (c->filter_alloc*phase_count )*c->felem_size, new_filter_bank + (c->filter_alloc - 1)*c->felem_size, c->felem_size);
|
|
|
|
if (!av_reduce(&new_src_incr, &new_dst_incr, c->src_incr,
|
|
c->dst_incr * (int64_t)(phase_count/c->phase_count), INT32_MAX/2))
|
|
{
|
|
av_freep(&new_filter_bank);
|
|
return AVERROR(EINVAL);
|
|
}
|
|
|
|
c->src_incr = new_src_incr;
|
|
c->dst_incr = new_dst_incr;
|
|
while (c->dst_incr < (1<<20) && c->src_incr < (1<<20)) {
|
|
c->dst_incr *= 2;
|
|
c->src_incr *= 2;
|
|
}
|
|
c->ideal_dst_incr = c->dst_incr;
|
|
c->dst_incr_div = c->dst_incr / c->src_incr;
|
|
c->dst_incr_mod = c->dst_incr % c->src_incr;
|
|
c->index *= phase_count / c->phase_count;
|
|
c->phase_count = phase_count;
|
|
av_freep(&c->filter_bank);
|
|
c->filter_bank = new_filter_bank;
|
|
return 0;
|
|
}
|
|
|
|
static int set_compensation(ResampleContext *c, int sample_delta, int compensation_distance){
|
|
int ret;
|
|
|
|
if (compensation_distance && sample_delta) {
|
|
ret = rebuild_filter_bank_with_compensation(c);
|
|
if (ret < 0)
|
|
return ret;
|
|
}
|
|
|
|
c->compensation_distance= compensation_distance;
|
|
if (compensation_distance)
|
|
c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
|
|
else
|
|
c->dst_incr = c->ideal_dst_incr;
|
|
|
|
c->dst_incr_div = c->dst_incr / c->src_incr;
|
|
c->dst_incr_mod = c->dst_incr % c->src_incr;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
|
|
int i;
|
|
int av_unused mm_flags = av_get_cpu_flags();
|
|
int need_emms = c->format == AV_SAMPLE_FMT_S16P && ARCH_X86_32 &&
|
|
(mm_flags & (AV_CPU_FLAG_MMX2 | AV_CPU_FLAG_SSE2)) == AV_CPU_FLAG_MMX2;
|
|
int64_t max_src_size = (INT64_MAX/2 / c->phase_count) / c->src_incr;
|
|
|
|
if (c->compensation_distance)
|
|
dst_size = FFMIN(dst_size, c->compensation_distance);
|
|
src_size = FFMIN(src_size, max_src_size);
|
|
|
|
*consumed = 0;
|
|
|
|
if (c->filter_length == 1 && c->phase_count == 1) {
|
|
int64_t index2= (1LL<<32)*c->frac/c->src_incr + (1LL<<32)*c->index;
|
|
int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
|
|
int new_size = (src_size * (int64_t)c->src_incr - c->frac + c->dst_incr - 1) / c->dst_incr;
|
|
|
|
dst_size = FFMAX(FFMIN(dst_size, new_size), 0);
|
|
if (dst_size > 0) {
|
|
for (i = 0; i < dst->ch_count; i++) {
|
|
c->dsp.resample_one(dst->ch[i], src->ch[i], dst_size, index2, incr);
|
|
if (i+1 == dst->ch_count) {
|
|
c->index += dst_size * c->dst_incr_div;
|
|
c->index += (c->frac + dst_size * (int64_t)c->dst_incr_mod) / c->src_incr;
|
|
av_assert2(c->index >= 0);
|
|
*consumed = c->index;
|
|
c->frac = (c->frac + dst_size * (int64_t)c->dst_incr_mod) % c->src_incr;
|
|
c->index = 0;
|
|
}
|
|
}
|
|
}
|
|
} else {
|
|
int64_t end_index = (1LL + src_size - c->filter_length) * c->phase_count;
|
|
int64_t delta_frac = (end_index - c->index) * c->src_incr - c->frac;
|
|
int delta_n = (delta_frac + c->dst_incr - 1) / c->dst_incr;
|
|
int (*resample_func)(struct ResampleContext *c, void *dst,
|
|
const void *src, int n, int update_ctx);
|
|
|
|
dst_size = FFMAX(FFMIN(dst_size, delta_n), 0);
|
|
if (dst_size > 0) {
|
|
/* resample_linear and resample_common should have same behavior
|
|
* when frac and dst_incr_mod are zero */
|
|
resample_func = (c->linear && (c->frac || c->dst_incr_mod)) ?
|
|
c->dsp.resample_linear : c->dsp.resample_common;
|
|
for (i = 0; i < dst->ch_count; i++)
|
|
*consumed = resample_func(c, dst->ch[i], src->ch[i], dst_size, i+1 == dst->ch_count);
|
|
}
|
|
}
|
|
|
|
if(need_emms)
|
|
emms_c();
|
|
|
|
if (c->compensation_distance) {
|
|
c->compensation_distance -= dst_size;
|
|
if (!c->compensation_distance) {
|
|
c->dst_incr = c->ideal_dst_incr;
|
|
c->dst_incr_div = c->dst_incr / c->src_incr;
|
|
c->dst_incr_mod = c->dst_incr % c->src_incr;
|
|
}
|
|
}
|
|
|
|
return dst_size;
|
|
}
|
|
|
|
static int64_t get_delay(struct SwrContext *s, int64_t base){
|
|
ResampleContext *c = s->resample;
|
|
int64_t num = s->in_buffer_count - (c->filter_length-1)/2;
|
|
num *= c->phase_count;
|
|
num -= c->index;
|
|
num *= c->src_incr;
|
|
num -= c->frac;
|
|
return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr * c->phase_count);
|
|
}
|
|
|
|
static int64_t get_out_samples(struct SwrContext *s, int in_samples) {
|
|
ResampleContext *c = s->resample;
|
|
// The + 2 are added to allow implementations to be slightly inaccurate, they should not be needed currently.
|
|
// They also make it easier to proof that changes and optimizations do not
|
|
// break the upper bound.
|
|
int64_t num = s->in_buffer_count + 2LL + in_samples;
|
|
num *= c->phase_count;
|
|
num -= c->index;
|
|
num = av_rescale_rnd(num, s->out_sample_rate, ((int64_t)s->in_sample_rate) * c->phase_count, AV_ROUND_UP) + 2;
|
|
|
|
if (c->compensation_distance) {
|
|
if (num > INT_MAX)
|
|
return AVERROR(EINVAL);
|
|
|
|
num = FFMAX(num, (num * c->ideal_dst_incr - 1) / c->dst_incr + 1);
|
|
}
|
|
return num;
|
|
}
|
|
|
|
static int resample_flush(struct SwrContext *s) {
|
|
ResampleContext *c = s->resample;
|
|
AudioData *a= &s->in_buffer;
|
|
int i, j, ret;
|
|
int reflection = (FFMIN(s->in_buffer_count, c->filter_length) + 1) / 2;
|
|
|
|
if((ret = swri_realloc_audio(a, s->in_buffer_index + s->in_buffer_count + reflection)) < 0)
|
|
return ret;
|
|
av_assert0(a->planar);
|
|
for(i=0; i<a->ch_count; i++){
|
|
for(j=0; j<reflection; j++){
|
|
memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,
|
|
a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
|
|
}
|
|
}
|
|
s->in_buffer_count += reflection;
|
|
return 0;
|
|
}
|
|
|
|
// in fact the whole handle multiple ridiculously small buffers might need more thinking...
|
|
static int invert_initial_buffer(ResampleContext *c, AudioData *dst, const AudioData *src,
|
|
int in_count, int *out_idx, int *out_sz)
|
|
{
|
|
int n, ch, num = FFMIN(in_count + *out_sz, c->filter_length + 1), res;
|
|
|
|
if (c->index >= 0)
|
|
return 0;
|
|
|
|
if ((res = swri_realloc_audio(dst, c->filter_length * 2 + 1)) < 0)
|
|
return res;
|
|
|
|
// copy
|
|
for (n = *out_sz; n < num; n++) {
|
|
for (ch = 0; ch < src->ch_count; ch++) {
|
|
memcpy(dst->ch[ch] + ((c->filter_length + n) * c->felem_size),
|
|
src->ch[ch] + ((n - *out_sz) * c->felem_size), c->felem_size);
|
|
}
|
|
}
|
|
|
|
// if not enough data is in, return and wait for more
|
|
if (num < c->filter_length + 1) {
|
|
*out_sz = num;
|
|
*out_idx = c->filter_length;
|
|
return INT_MAX;
|
|
}
|
|
|
|
// else invert
|
|
for (n = 1; n <= c->filter_length; n++) {
|
|
for (ch = 0; ch < src->ch_count; ch++) {
|
|
memcpy(dst->ch[ch] + ((c->filter_length - n) * c->felem_size),
|
|
dst->ch[ch] + ((c->filter_length + n) * c->felem_size),
|
|
c->felem_size);
|
|
}
|
|
}
|
|
|
|
res = num - *out_sz;
|
|
*out_idx = c->filter_length;
|
|
while (c->index < 0) {
|
|
--*out_idx;
|
|
c->index += c->phase_count;
|
|
}
|
|
*out_sz = FFMAX(*out_sz + c->filter_length,
|
|
1 + c->filter_length * 2) - *out_idx;
|
|
|
|
return FFMAX(res, 0);
|
|
}
|
|
|
|
struct Resampler const swri_resampler={
|
|
resample_init,
|
|
resample_free,
|
|
multiple_resample,
|
|
resample_flush,
|
|
set_compensation,
|
|
get_delay,
|
|
invert_initial_buffer,
|
|
get_out_samples,
|
|
};
|