mirror of https://git.ffmpeg.org/ffmpeg.git
avfilter: add aexciter audio filter
This commit is contained in:
Paul B Mahol
parent
129978af6b
commit
579e4e57a2
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@ -69,6 +69,7 @@ version <next>:
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- xbm_pipe demuxer
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- colorize filter
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- CRI parser
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- aexciter audio filter
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version 4.3:
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@ -1003,6 +1003,62 @@ aeval=val(0)|-val(1)
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@end example
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@end itemize
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@section aexciter
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An exciter is used to produce high sound that is not present in the
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original signal. This is done by creating harmonic distortions of the
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signal which are restricted in range and added to the original signal.
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An Exciter raises the upper end of an audio signal without simply raising
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the higher frequencies like an equalizer would do to create a more
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"crisp" or "brilliant" sound.
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The filter accepts the following options:
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@table @option
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@item level_in
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Set input level prior processing of signal.
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Allowed range is from 0 to 64.
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Default value is 1.
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@item level_out
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Set output level after processing of signal.
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Allowed range is from 0 to 64.
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Default value is 1.
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@item amount
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Set the amount of harmonics added to original signal.
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Allowed range is from 0 to 64.
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Default value is 1.
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@item drive
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Set the amount of newly created harmonics.
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Allowed range is from 0.1 to 10.
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Default value is 8.5.
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@item blend
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Set the octave of newly created harmonics.
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Allowed range is from -10 to 10.
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Default value is 0.
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@item freq
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Set the lower frequency limit of producing harmonics in Hz.
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Allowed range is from 2000 to 12000 Hz.
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Default is 7500 Hz.
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@item ceil
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Set the upper frequency limit of producing harmonics.
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Allowed range is from 9999 to 20000 Hz.
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If value is lower than 10000 Hz no limit is applied.
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@item listen
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Mute the original signal and output only added harmonics.
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By default is disabled.
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@end table
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@subsection Commands
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This filter supports the all above options as @ref{commands}.
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@anchor{afade}
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@section afade
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@ -46,6 +46,7 @@ OBJS-$(CONFIG_ADERIVATIVE_FILTER) += af_aderivative.o
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OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
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OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o
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OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o
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OBJS-$(CONFIG_AEXCITER_FILTER) += af_aexciter.o
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OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
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OBJS-$(CONFIG_AFFTDN_FILTER) += af_afftdn.o
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OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o
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@ -0,0 +1,317 @@
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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 ChannelParams {
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double blend_old, drive_old;
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double rdrive, rbdr, kpa, kpb, kna, knb, ap,
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an, imr, kc, srct, sq, pwrq;
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double prev_med, prev_out;
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double hp[5], lp[5];
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double hw[4][2], lw[2][2];
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} ChannelParams;
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typedef struct AExciterContext {
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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 amount;
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double drive;
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double blend;
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double freq;
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double ceil;
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int listen;
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ChannelParams *cp;
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} AExciterContext;
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#define OFFSET(x) offsetof(AExciterContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
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static const AVOption aexciter_options[] = {
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{ "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A },
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{ "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A },
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{ "amount", "set amount", OFFSET(amount), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A },
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{ "drive", "set harmonics", OFFSET(drive), AV_OPT_TYPE_DOUBLE, {.dbl=8.5}, 0.1, 10, A },
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{ "blend", "set blend harmonics", OFFSET(blend), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -10, 10, A },
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{ "freq", "set scope", OFFSET(freq), AV_OPT_TYPE_DOUBLE, {.dbl=7500}, 2000, 12000, A },
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{ "ceil", "set ceiling", OFFSET(ceil), AV_OPT_TYPE_DOUBLE, {.dbl=9999}, 9999, 20000, A },
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{ "listen", "enable listen mode", OFFSET(listen), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(aexciter);
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static inline double M(double x)
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{
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return (fabs(x) > 0.00000001) ? x : 0.0;
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}
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static inline double D(double x)
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{
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x = fabs(x);
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return (x > 0.00000001) ? sqrt(x) : 0.0;
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}
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static void set_params(ChannelParams *p,
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double blend, double drive,
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double srate, double freq,
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double ceil)
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{
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double a0, a1, a2, b0, b1, b2, w0, alpha;
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p->rdrive = 12.0 / drive;
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p->rbdr = p->rdrive / (10.5 - blend) * 780.0 / 33.0;
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p->kpa = D(2.0 * (p->rdrive*p->rdrive) - 1.0) + 1.0;
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p->kpb = (2.0 - p->kpa) / 2.0;
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p->ap = ((p->rdrive*p->rdrive) - p->kpa + 1.0) / 2.0;
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p->kc = p->kpa / D(2.0 * D(2.0 * (p->rdrive*p->rdrive) - 1.0) - 2.0 * p->rdrive*p->rdrive);
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p->srct = (0.1 * srate) / (0.1 * srate + 1.0);
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p->sq = p->kc*p->kc + 1.0;
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p->knb = -1.0 * p->rbdr / D(p->sq);
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p->kna = 2.0 * p->kc * p->rbdr / D(p->sq);
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p->an = p->rbdr*p->rbdr / p->sq;
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p->imr = 2.0 * p->knb + D(2.0 * p->kna + 4.0 * p->an - 1.0);
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p->pwrq = 2.0 / (p->imr + 1.0);
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w0 = 2 * M_PI * freq / srate;
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alpha = sin(w0) / (2. * 0.707);
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a0 = 1 + alpha;
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a1 = -2 * cos(w0);
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a2 = 1 - alpha;
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b0 = (1 + cos(w0)) / 2;
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b1 = -(1 + cos(w0));
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b2 = (1 + cos(w0)) / 2;
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p->hp[0] =-a1 / a0;
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p->hp[1] =-a2 / a0;
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p->hp[2] = b0 / a0;
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p->hp[3] = b1 / a0;
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p->hp[4] = b2 / a0;
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w0 = 2 * M_PI * ceil / srate;
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alpha = sin(w0) / (2. * 0.707);
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a0 = 1 + alpha;
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a1 = -2 * cos(w0);
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a2 = 1 - alpha;
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b0 = (1 - cos(w0)) / 2;
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b1 = 1 - cos(w0);
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b2 = (1 - cos(w0)) / 2;
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p->lp[0] =-a1 / a0;
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p->lp[1] =-a2 / a0;
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p->lp[2] = b0 / a0;
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p->lp[3] = b1 / a0;
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p->lp[4] = b2 / a0;
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}
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static double bprocess(double in, const double *const c,
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double *w1, double *w2)
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{
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double out = c[2] * in + *w1;
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*w1 = c[3] * in + *w2 + c[0] * out;
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*w2 = c[4] * in + c[1] * out;
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return out;
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}
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static double distortion_process(AExciterContext *s, ChannelParams *p, double in)
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{
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double proc = in, med;
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proc = bprocess(proc, p->hp, &p->hw[0][0], &p->hw[0][1]);
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proc = bprocess(proc, p->hp, &p->hw[1][0], &p->hw[1][1]);
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if (proc >= 0.0) {
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med = (D(p->ap + proc * (p->kpa - proc)) + p->kpb) * p->pwrq;
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} else {
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med = (D(p->an - proc * (p->kna + proc)) + p->knb) * p->pwrq * -1.0;
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}
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proc = p->srct * (med - p->prev_med + p->prev_out);
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p->prev_med = M(med);
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p->prev_out = M(proc);
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proc = bprocess(proc, p->hp, &p->hw[2][0], &p->hw[2][1]);
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proc = bprocess(proc, p->hp, &p->hw[3][0], &p->hw[3][1]);
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if (s->ceil >= 10000.) {
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proc = bprocess(proc, p->lp, &p->lw[0][0], &p->lw[0][1]);
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proc = bprocess(proc, p->lp, &p->lw[1][0], &p->lw[1][1]);
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}
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return proc;
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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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AExciterContext *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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const double level_in = s->level_in;
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const double level_out = s->level_out;
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const double amount = s->amount;
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const double listen = 1.0 - s->listen;
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double *dst;
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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 (int n = 0; n < in->nb_samples; n++) {
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for (int c = 0; c < inlink->channels; c++) {
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double sample = src[c] * level_in;
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sample = distortion_process(s, &s->cp[c], sample);
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sample = sample * amount + listen * src[c];
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sample *= level_out;
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if (ctx->is_disabled)
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dst[c] = src[c];
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else
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dst[c] = sample;
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}
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src += inlink->channels;
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dst += inlink->channels;
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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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AExciterContext *s = ctx->priv;
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av_freep(&s->cp);
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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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AExciterContext *s = ctx->priv;
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if (!s->cp)
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s->cp = av_calloc(inlink->channels, sizeof(*s->cp));
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if (!s->cp)
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return AVERROR(ENOMEM);
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for (int i = 0; i < inlink->channels; i++)
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set_params(&s->cp[i], s->blend, s->drive, inlink->sample_rate,
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s->freq, s->ceil);
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return 0;
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}
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static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
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char *res, int res_len, int flags)
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{
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AVFilterLink *inlink = ctx->inputs[0];
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int ret;
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ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
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if (ret < 0)
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return ret;
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return config_input(inlink);
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}
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static const AVFilterPad avfilter_af_aexciter_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_aexciter_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_aexciter = {
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.name = "aexciter",
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.description = NULL_IF_CONFIG_SMALL("Enhance high frequency part of audio."),
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.priv_size = sizeof(AExciterContext),
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.priv_class = &aexciter_class,
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.uninit = uninit,
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.query_formats = query_formats,
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.inputs = avfilter_af_aexciter_inputs,
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.outputs = avfilter_af_aexciter_outputs,
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.process_command = process_command,
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.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
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};
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@ -39,6 +39,7 @@ extern AVFilter ff_af_aderivative;
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extern AVFilter ff_af_aecho;
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extern AVFilter ff_af_aemphasis;
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extern AVFilter ff_af_aeval;
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extern AVFilter ff_af_aexciter;
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extern AVFilter ff_af_afade;
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extern AVFilter ff_af_afftdn;
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extern AVFilter ff_af_afftfilt;
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@ -30,7 +30,7 @@
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#include "libavutil/version.h"
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#define LIBAVFILTER_VERSION_MAJOR 7
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#define LIBAVFILTER_VERSION_MINOR 103
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#define LIBAVFILTER_VERSION_MINOR 104
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#define LIBAVFILTER_VERSION_MICRO 100
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