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623 lines
22 KiB
623 lines
22 KiB
/* |
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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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/** |
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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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|
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#include "libavutil/avassert.h" |
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#include "libavutil/cpu.h" |
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#include "resample.h" |
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|
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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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/** |
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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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|
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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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|
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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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|
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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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/** |
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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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|
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if (!tab || !sin_lut) |
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goto fail; |
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|
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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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|
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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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|
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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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|
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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); |
|
av_freep(cc); |
|
} |
|
|
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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, |
|
double precision, int cheby, int exact_rational) |
|
{ |
|
double cutoff = cutoff0? cutoff0 : 0.97; |
|
double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); |
|
int phase_count= 1<<phase_shift; |
|
int phase_count_compensation = phase_count; |
|
int filter_length = FFMAX((int)ceil(filter_size/factor), 1); |
|
|
|
if (filter_length > 1) |
|
filter_length = FFALIGN(filter_length, 2); |
|
|
|
if (exact_rational) { |
|
int phase_count_exact, phase_count_exact_den; |
|
|
|
av_reduce(&phase_count_exact, &phase_count_exact_den, out_rate, in_rate, INT_MAX); |
|
if (phase_count_exact <= phase_count) { |
|
phase_count_compensation = phase_count_exact * (phase_count / phase_count_exact); |
|
phase_count = phase_count_exact; |
|
} |
|
} |
|
|
|
if (!c || c->phase_count != phase_count || c->linear!=linear || c->factor != factor |
|
|| c->filter_length != filter_length || c->format != format |
|
|| c->filter_type != filter_type || c->kaiser_beta != kaiser_beta) { |
|
resample_free(&c); |
|
c = av_mallocz(sizeof(*c)); |
|
if (!c) |
|
return NULL; |
|
|
|
c->format= format; |
|
|
|
c->felem_size= av_get_bytes_per_sample(c->format); |
|
|
|
switch(c->format){ |
|
case AV_SAMPLE_FMT_S16P: |
|
c->filter_shift = 15; |
|
break; |
|
case AV_SAMPLE_FMT_S32P: |
|
c->filter_shift = 30; |
|
break; |
|
case AV_SAMPLE_FMT_FLTP: |
|
case AV_SAMPLE_FMT_DBLP: |
|
c->filter_shift = 0; |
|
break; |
|
default: |
|
av_log(NULL, AV_LOG_ERROR, "Unsupported sample format\n"); |
|
av_assert0(0); |
|
} |
|
|
|
if (filter_size/factor > INT32_MAX/256) { |
|
av_log(NULL, AV_LOG_ERROR, "Filter length too large\n"); |
|
goto error; |
|
} |
|
|
|
c->phase_count = phase_count; |
|
c->linear = linear; |
|
c->factor = factor; |
|
c->filter_length = filter_length; |
|
c->filter_alloc = FFALIGN(c->filter_length, 8); |
|
c->filter_bank = av_calloc(c->filter_alloc, (phase_count+1)*c->felem_size); |
|
c->filter_type = filter_type; |
|
c->kaiser_beta = kaiser_beta; |
|
c->phase_count_compensation = phase_count_compensation; |
|
if (!c->filter_bank) |
|
goto error; |
|
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)) |
|
goto error; |
|
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); |
|
} |
|
|
|
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, |
|
};
|
|
|