mirror of https://github.com/FFmpeg/FFmpeg.git
Paul B Mahol
9 years ago
parent
cc6a59d2b9
commit
7f1b14bc57
6 changed files with 424 additions and 1 deletions
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/*
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* Copyright (c) Markus Schmidt and Christian Holschuh |
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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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#include "libavutil/opt.h" |
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#include "avfilter.h" |
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#include "internal.h" |
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#include "audio.h" |
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typedef struct LFOContext { |
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double freq; |
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double offset; |
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int srate; |
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double amount; |
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double pwidth; |
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double phase; |
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} LFOContext; |
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typedef struct SRContext { |
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double target; |
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double real; |
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double samples; |
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double last; |
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} SRContext; |
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typedef struct ACrusherContext { |
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const AVClass *class; |
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double level_in; |
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double level_out; |
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double bits; |
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double mix; |
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int mode; |
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double dc; |
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double idc; |
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double aa; |
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double samples; |
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int is_lfo; |
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double lforange; |
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double lforate; |
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double sqr; |
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double aa1; |
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double coeff; |
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int round; |
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double sov; |
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double smin; |
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double sdiff; |
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LFOContext lfo; |
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SRContext *sr; |
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} ACrusherContext; |
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#define OFFSET(x) offsetof(ACrusherContext, x) |
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
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static const AVOption acrusher_options[] = { |
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{ "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A }, |
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{ "level_out","set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A }, |
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{ "bits", "set bit reduction", OFFSET(bits), AV_OPT_TYPE_DOUBLE, {.dbl=8}, 1, 64, A }, |
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{ "mix", "set mix", OFFSET(mix), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, 0, 1, A }, |
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{ "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "mode" }, |
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{ "lin", "linear", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "mode" }, |
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{ "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "mode" }, |
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{ "dc", "set DC", OFFSET(dc), AV_OPT_TYPE_DOUBLE, {.dbl=1}, .25, 4, A }, |
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{ "aa", "set anti-aliasing", OFFSET(aa), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, 0, 1, A }, |
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{ "samples", "set sample reduction", OFFSET(samples), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 250, A }, |
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{ "lfo", "enable LFO", OFFSET(is_lfo), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A }, |
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{ "lforange", "set LFO depth", OFFSET(lforange), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 1, 250, A }, |
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{ "lforate", "set LFO rate", OFFSET(lforate), AV_OPT_TYPE_DOUBLE, {.dbl=.3}, .01, 200, A }, |
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{ NULL } |
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}; |
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AVFILTER_DEFINE_CLASS(acrusher); |
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static double samplereduction(ACrusherContext *s, SRContext *sr, double in) |
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{ |
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sr->samples++; |
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if (sr->samples >= s->round) { |
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sr->target += s->samples; |
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sr->real += s->round; |
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if (sr->target + s->samples >= sr->real + 1) { |
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sr->last = in; |
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sr->target = 0; |
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sr->real = 0; |
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} |
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sr->samples = 0; |
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} |
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return sr->last; |
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} |
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static double add_dc(double s, double dc, double idc) |
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{ |
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return s > 0 ? s * dc : s * idc; |
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} |
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static double remove_dc(double s, double dc, double idc) |
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{ |
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return s > 0 ? s * idc : s * dc; |
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} |
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static inline double factor(double y, double k, double aa1, double aa) |
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{ |
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return 0.5 * (sin(M_PI * (fabs(y - k) - aa1) / aa - M_PI_2) + 1); |
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} |
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static double bitreduction(ACrusherContext *s, double in) |
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{ |
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const double sqr = s->sqr; |
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const double coeff = s->coeff; |
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const double aa = s->aa; |
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const double aa1 = s->aa1; |
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double y, k; |
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// add dc
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in = add_dc(in, s->dc, s->idc); |
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// main rounding calculation depending on mode
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// the idea for anti-aliasing:
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// you need a function f which brings you to the scale, where
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// you want to round and the function f_b (with f(f_b)=id) which
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// brings you back to your original scale.
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//
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// then you can use the logic below in the following way:
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// y = f(in) and k = roundf(y)
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// if (y > k + aa1)
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// k = f_b(k) + ( f_b(k+1) - f_b(k) ) * 0.5 * (sin(x - PI/2) + 1)
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// if (y < k + aa1)
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// k = f_b(k) - ( f_b(k+1) - f_b(k) ) * 0.5 * (sin(x - PI/2) + 1)
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//
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// whereas x = (fabs(f(in) - k) - aa1) * PI / aa
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// for both cases.
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switch (s->mode) { |
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case 0: |
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default: |
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// linear
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y = in * coeff; |
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k = roundf(y); |
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if (k - aa1 <= y && y <= k + aa1) { |
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k /= coeff; |
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} else if (y > k + aa1) { |
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k = k / coeff + ((k + 1) / coeff - k / coeff) * |
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factor(y, k, aa1, aa); |
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} else { |
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k = k / coeff - (k / coeff - (k - 1) / coeff) * |
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factor(y, k, aa1, aa); |
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} |
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break; |
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case 1: |
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// logarithmic
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y = sqr * log(fabs(in)) + sqr * sqr; |
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k = roundf(y); |
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if(!in) { |
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k = 0; |
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} else if (k - aa1 <= y && y <= k + aa1) { |
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k = in / fabs(in) * exp(k / sqr - sqr); |
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} else if (y > k + aa1) { |
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double x = exp(k / sqr - sqr); |
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k = FFSIGN(in) * (x + (exp((k + 1) / sqr - sqr) - x) * |
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factor(y, k, aa1, aa)); |
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} else { |
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double x = exp(k / sqr - sqr); |
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k = in / fabs(in) * (x - (x - exp((k - 1) / sqr - sqr)) * |
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factor(y, k, aa1, aa)); |
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} |
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break; |
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} |
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// mix between dry and wet signal
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k += (in - k) * s->mix; |
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// remove dc
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k = remove_dc(k, s->dc, s->idc); |
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return k; |
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} |
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static double lfo_get(LFOContext *lfo) |
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{ |
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double phs = FFMIN(100., lfo->phase / FFMIN(1.99, FFMAX(0.01, lfo->pwidth)) + lfo->offset); |
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double val; |
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if (phs > 1) |
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phs = fmod(phs, 1.); |
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val = sin((phs * 360.) * M_PI / 180); |
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return val * lfo->amount; |
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} |
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static void lfo_advance(LFOContext *lfo, unsigned count) |
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{ |
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lfo->phase = fabs(lfo->phase + count * lfo->freq * (1. / lfo->srate)); |
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if (lfo->phase >= 1.) |
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lfo->phase = fmod(lfo->phase, 1.); |
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} |
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static int filter_frame(AVFilterLink *inlink, AVFrame *in) |
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{ |
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AVFilterContext *ctx = inlink->dst; |
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ACrusherContext *s = ctx->priv; |
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AVFilterLink *outlink = ctx->outputs[0]; |
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AVFrame *out; |
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const double *src = (const double *)in->data[0]; |
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double *dst; |
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const double level_in = s->level_in; |
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const double level_out = s->level_out; |
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const double mix = s->mix; |
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int n, c; |
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if (av_frame_is_writable(in)) { |
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out = in; |
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} else { |
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out = ff_get_audio_buffer(inlink, in->nb_samples); |
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if (!out) { |
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av_frame_free(&in); |
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return AVERROR(ENOMEM); |
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} |
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av_frame_copy_props(out, in); |
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} |
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dst = (double *)out->data[0]; |
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for (n = 0; n < in->nb_samples; n++) { |
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if (s->is_lfo) { |
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s->samples = s->smin + s->sdiff * (lfo_get(&s->lfo) + 0.5); |
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s->round = round(s->samples); |
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} |
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for (c = 0; c < inlink->channels; c++) { |
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double sample = src[c] * level_in; |
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sample = mix * samplereduction(s, &s->sr[c], sample) + src[c] * (1. - mix) * level_in; |
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dst[c] = bitreduction(s, sample) * level_out; |
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} |
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src += c; |
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dst += c; |
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if (s->is_lfo) |
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lfo_advance(&s->lfo, 1); |
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} |
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if (in != out) |
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av_frame_free(&in); |
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return ff_filter_frame(outlink, out); |
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} |
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static int query_formats(AVFilterContext *ctx) |
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{ |
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AVFilterFormats *formats; |
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AVFilterChannelLayouts *layouts; |
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static const enum AVSampleFormat sample_fmts[] = { |
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AV_SAMPLE_FMT_DBL, |
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AV_SAMPLE_FMT_NONE |
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}; |
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int ret; |
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layouts = ff_all_channel_counts(); |
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if (!layouts) |
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return AVERROR(ENOMEM); |
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ret = ff_set_common_channel_layouts(ctx, layouts); |
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if (ret < 0) |
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return ret; |
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formats = ff_make_format_list(sample_fmts); |
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if (!formats) |
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return AVERROR(ENOMEM); |
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ret = ff_set_common_formats(ctx, formats); |
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if (ret < 0) |
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return ret; |
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formats = ff_all_samplerates(); |
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if (!formats) |
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return AVERROR(ENOMEM); |
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return ff_set_common_samplerates(ctx, formats); |
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} |
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static av_cold void uninit(AVFilterContext *ctx) |
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{ |
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ACrusherContext *s = ctx->priv; |
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av_freep(&s->sr); |
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} |
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static int config_input(AVFilterLink *inlink) |
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{ |
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AVFilterContext *ctx = inlink->dst; |
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ACrusherContext *s = ctx->priv; |
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double rad, sun, smax, sov; |
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s->idc = 1. / s->dc; |
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s->coeff = exp2(s->bits) - 1; |
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s->sqr = sqrt(s->coeff / 2); |
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s->aa1 = (1. - s->aa) / 2.; |
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s->round = round(s->samples); |
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rad = s->lforange / 2.; |
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s->smin = FFMAX(s->samples - rad, 1.); |
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sun = s->samples - rad - s->smin; |
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smax = FFMIN(s->samples + rad, 250.); |
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sov = s->samples + rad - smax; |
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smax -= sun; |
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s->smin -= sov; |
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s->sdiff = smax - s->smin; |
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s->lfo.freq = s->lforate; |
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s->lfo.pwidth = 1.; |
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s->lfo.srate = inlink->sample_rate; |
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s->lfo.amount = .5; |
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s->sr = av_calloc(inlink->channels, sizeof(*s->sr)); |
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if (!s->sr) |
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return AVERROR(ENOMEM); |
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return 0; |
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} |
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static const AVFilterPad avfilter_af_acrusher_inputs[] = { |
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{ |
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.name = "default", |
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.type = AVMEDIA_TYPE_AUDIO, |
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.config_props = config_input, |
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.filter_frame = filter_frame, |
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}, |
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{ NULL } |
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}; |
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static const AVFilterPad avfilter_af_acrusher_outputs[] = { |
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{ |
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.name = "default", |
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.type = AVMEDIA_TYPE_AUDIO, |
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}, |
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{ NULL } |
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}; |
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AVFilter ff_af_acrusher = { |
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.name = "acrusher", |
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.description = NULL_IF_CONFIG_SMALL("Reduce audio bit resolution."), |
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.priv_size = sizeof(ACrusherContext), |
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.priv_class = &acrusher_class, |
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.uninit = uninit, |
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.query_formats = query_formats, |
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.inputs = avfilter_af_acrusher_inputs, |
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.outputs = avfilter_af_acrusher_outputs, |
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}; |
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