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avfilter: add audio dynamic equalizer filter
This commit is contained in:
Paul B Mahol
parent
466441a0d2
commit
996b13fac4
@ -40,6 +40,7 @@ version <next>:
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- adynamicsmooth audio filter
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- libplacebo filter
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- vflip_vulkan, hflip_vulkan and flip_vulkan filters
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- adynamicequalizer audio filter
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version 4.4:
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@ -843,6 +843,82 @@ Compute derivative/integral of audio stream.
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Applying both filters one after another produces original audio.
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@section adynamicequalizer
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Apply dynamic equalization to input audio stream.
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A description of the accepted options follows.
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@table @option
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@item threshold
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Set the detection threshold used to trigger equalization.
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Threshold detection is using bandpass filter.
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Default value is 0. Allowed range is from 0 to 100.
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@item dfrequency
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Set the detection frequency in Hz used for bandpass filter used to trigger equalization.
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Default value is 1000 Hz. Allowed range is between 2 and 1000000 Hz.
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@item dqfactor
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Set the detection resonance factor for bandpass filter used to trigger equalization.
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Default value is 1. Allowed range is from 0.001 to 1000.
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@item tfrequency
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Set the target frequency of equalization filter.
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Default value is 1000 Hz. Allowed range is between 2 and 1000000 Hz.
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@item tqfactor
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Set the target resonance factor for target equalization filter.
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Default value is 1. Allowed range is from 0.001 to 1000.
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@item attack
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Set the amount of milliseconds the signal from detection has to rise above
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the detection threshold before equalization starts.
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Default is 20. Allowed range is between 1 and 2000.
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@item release
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Set the amount of milliseconds the signal from detection has to fall below the
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detection threshold before equalization ends.
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Default is 200. Allowed range is between 1 and 2000.
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@item knee
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Curve the sharp knee around the detection threshold to calculate
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equalization gain more softly.
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Default is 1. Allowed range is between 0 and 8.
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@item ratio
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Set the ratio by which the equalization gain is raised.
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Default is 1. Allowed range is between 1 and 20.
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@item makeup
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Set the makeup offset in dB by which the equalization gain is raised.
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Default is 0. Allowed range is between 0 and 30.
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@item range
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Set the max allowed cut/boost amount in dB. Default is 0.
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Allowed range is from 0 to 200.
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@item slew
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Set the slew factor. Default is 1. Allowed range is from 1 to 200.
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@item mode
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Set the mode of filter operation, can be one of the following:
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@table @samp
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@item listen
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Output only isolated bandpass signal.
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@item cut
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Cut frequencies above detection threshold.
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@item boost
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Boost frequencies bellow detection threshold.
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@end table
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Default mode is @samp{cut}.
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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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@section adynamicsmooth
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Apply dynamic smoothing to input audio stream.
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@ -44,6 +44,7 @@ OBJS-$(CONFIG_ADECORRELATE_FILTER) += af_adecorrelate.o
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OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.o
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OBJS-$(CONFIG_ADENORM_FILTER) += af_adenorm.o
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OBJS-$(CONFIG_ADERIVATIVE_FILTER) += af_aderivative.o
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OBJS-$(CONFIG_ADYNAMICEQUALIZER_FILTER) += af_adynamicequalizer.o
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OBJS-$(CONFIG_ADYNAMICSMOOTH_FILTER) += af_adynamicsmooth.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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315
libavfilter/af_adynamicequalizer.c
Normal file
315
libavfilter/af_adynamicequalizer.c
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@ -0,0 +1,315 @@
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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 <float.h>
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#include "libavutil/opt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "formats.h"
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#include "hermite.h"
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typedef struct AudioDynamicEqualizerContext {
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const AVClass *class;
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double threshold;
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double dfrequency;
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double dqfactor;
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double tfrequency;
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double tqfactor;
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double ratio;
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double range;
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double makeup;
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double knee;
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double slew;
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double attack;
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double release;
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double attack_coef;
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double release_coef;
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int mode;
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AVFrame *state;
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} AudioDynamicEqualizerContext;
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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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AudioDynamicEqualizerContext *s = ctx->priv;
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s->state = ff_get_audio_buffer(inlink, 8);
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if (!s->state)
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return AVERROR(ENOMEM);
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return 0;
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}
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static double get_svf(double in, double *m, double *a, double *b)
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{
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const double v0 = in;
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const double v3 = v0 - b[1];
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const double v1 = a[0] * b[0] + a[1] * v3;
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const double v2 = b[1] + a[1] * b[0] + a[2] * v3;
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b[0] = 2. * v1 - b[0];
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b[1] = 2. * v2 - b[1];
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return m[0] * v0 + m[1] * v1 + m[2] * v2;
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}
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static inline double from_dB(double x)
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{
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return exp(0.05 * x * M_LN10);
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}
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static inline double to_dB(double x)
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{
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return 20. * log10(x);
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}
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static inline double sqr(double x)
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{
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return x * x;
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}
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static double get_gain(double in, double srate, double makeup,
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double aattack, double iratio, double knee, double range,
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double thresdb, double slewfactor, double *state,
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double attack_coeff, double release_coeff, double nc)
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{
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double width = (6. * knee) + 0.01;
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double cdb = 0.;
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double Lgain = 1.;
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double Lxg, Lxl, Lyg, Lyl, Ly1;
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double checkwidth = 0.;
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double slewwidth = 1.8;
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int attslew = 0;
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Lyg = 0.;
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Lxg = to_dB(fabs(in) + DBL_EPSILON);
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Lyg = Lxg + (iratio - 1.) * sqr(Lxg - thresdb + width * .5) / (2. * width);
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checkwidth = 2. * fabs(Lxg - thresdb);
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if (2. * (Lxg - thresdb) < -width) {
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Lyg = Lxg;
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} else if (checkwidth <= width) {
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Lyg = thresdb + (Lxg - thresdb) * iratio;
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if (checkwidth <= slewwidth) {
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if (Lyg >= state[2])
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attslew = 1;
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}
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} else if (2. * (Lxg-thresdb) > width) {
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Lyg = thresdb + (Lxg - thresdb) * iratio;
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}
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attack_coeff = attslew ? aattack : attack_coeff;
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Lxl = Lxg - Lyg;
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Ly1 = fmaxf(Lxl, release_coeff * state[1] +(1. - release_coeff) * Lxl);
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Lyl = attack_coeff * state[0] + (1. - attack_coeff) * Ly1;
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cdb = -Lyl;
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Lgain = from_dB(nc * fmin(cdb - makeup, range));
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state[0] = Lyl;
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state[1] = Ly1;
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state[2] = Lyg;
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return Lgain;
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}
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typedef struct ThreadData {
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AVFrame *in, *out;
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} ThreadData;
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static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
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{
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AudioDynamicEqualizerContext *s = ctx->priv;
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ThreadData *td = arg;
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AVFrame *in = td->in;
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AVFrame *out = td->out;
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const double sample_rate = in->sample_rate;
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const double makeup = s->makeup;
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const double iratio = 1. / s->ratio;
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const double range = s->range;
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const double dfrequency = fmin(s->dfrequency, sample_rate * 0.5);
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const double tfrequency = fmin(s->tfrequency, sample_rate * 0.5);
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const double threshold = log(s->threshold + DBL_EPSILON);
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const double release = s->release_coef;
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const double attack = s->attack_coef;
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const double dqfactor = s->dqfactor;
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const double tqfactor = s->tqfactor;
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const double fg = tan(M_PI * tfrequency / sample_rate);
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const double dg = tan(M_PI * dfrequency / sample_rate);
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const int start = (in->channels * jobnr) / nb_jobs;
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const int end = (in->channels * (jobnr+1)) / nb_jobs;
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const int mode = s->mode;
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const double knee = s->knee;
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const double slew = s->slew;
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const double aattack = exp(-1000. / ((s->attack + 2.0 * (slew - 1.)) * sample_rate));
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const double nc = mode == 0 ? 1. : -1.;
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double da[3], dm[3];
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{
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double k = 1. / dqfactor;
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da[0] = 1. / (1. + dg * (dg + k));
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da[1] = dg * da[0];
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da[2] = dg * da[1];
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dm[0] = 0.;
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dm[1] = 1.;
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dm[2] = 0.;
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}
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for (int ch = start; ch < end; ch++) {
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const double *src = (const double *)in->extended_data[ch];
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double *dst = (double *)out->extended_data[ch];
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double *state = (double *)s->state->extended_data[ch];
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for (int n = 0; n < out->nb_samples; n++) {
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double detect, gain, v, listen;
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double fa[3], fm[3];
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detect = listen = get_svf(src[n], dm, da, state);
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detect = fabs(detect);
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gain = get_gain(detect, sample_rate, makeup,
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aattack, iratio, knee, range, threshold, slew,
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&state[4], attack, release, nc);
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{
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double k = 1. / (tqfactor * gain);
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fa[0] = 1. / (1. + fg * (fg + k));
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fa[1] = fg * fa[0];
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fa[2] = fg * fa[1];
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fm[0] = 1.;
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fm[1] = k * (gain * gain - 1.);
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fm[2] = 0.;
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}
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v = get_svf(src[n], fm, fa, &state[2]);
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v = mode == -1 ? listen : v;
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dst[n] = ctx->is_disabled ? src[n] : v;
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}
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}
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return 0;
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}
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static double get_coef(double x, double sr)
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{
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return exp(-1000. / (x * sr));
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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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AVFilterLink *outlink = ctx->outputs[0];
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AudioDynamicEqualizerContext *s = ctx->priv;
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ThreadData td;
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AVFrame *out;
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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(outlink, 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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s->attack_coef = get_coef(s->attack, in->sample_rate);
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s->release_coef = get_coef(s->release, in->sample_rate);
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td.in = in;
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td.out = out;
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ff_filter_execute(ctx, filter_channels, &td, NULL,
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FFMIN(outlink->channels, ff_filter_get_nb_threads(ctx)));
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if (out != in)
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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 av_cold void uninit(AVFilterContext *ctx)
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{
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AudioDynamicEqualizerContext *s = ctx->priv;
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av_frame_free(&s->state);
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}
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#define OFFSET(x) offsetof(AudioDynamicEqualizerContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
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static const AVOption adynamicequalizer_options[] = {
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{ "threshold", "set detection threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 100, FLAGS },
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{ "dfrequency", "set detection frequency", OFFSET(dfrequency), AV_OPT_TYPE_DOUBLE, {.dbl=1000}, 2, 1000000, FLAGS },
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{ "dqfactor", "set detection Q factor", OFFSET(dqfactor), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.001, 1000, FLAGS },
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{ "tfrequency", "set target frequency", OFFSET(tfrequency), AV_OPT_TYPE_DOUBLE, {.dbl=1000}, 2, 1000000, FLAGS },
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{ "tqfactor", "set target Q factor", OFFSET(tqfactor), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.001, 1000, FLAGS },
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{ "attack", "set attack duration", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 1, 2000, FLAGS },
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{ "release", "set release duration", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=200}, 1, 2000, FLAGS },
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{ "knee", "set knee factor", OFFSET(knee), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 8, FLAGS },
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{ "ratio", "set ratio factor", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 20, FLAGS },
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{ "makeup", "set makeup gain", OFFSET(makeup), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 30, FLAGS },
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{ "range", "set max gain", OFFSET(range), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 200, FLAGS },
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{ "slew", "set slew factor", OFFSET(slew), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 200, FLAGS },
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{ "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, -1, 1, FLAGS, "mode" },
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{ "listen", 0, 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, FLAGS, "mode" },
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{ "cut", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "mode" },
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{ "boost", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "mode" },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(adynamicequalizer);
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static const AVFilterPad inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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.config_props = config_input,
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},
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};
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static const AVFilterPad 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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};
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const AVFilter ff_af_adynamicequalizer = {
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.name = "adynamicequalizer",
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.description = NULL_IF_CONFIG_SMALL("Apply Dynamic Equalization of input audio."),
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.priv_size = sizeof(AudioDynamicEqualizerContext),
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.priv_class = &adynamicequalizer_class,
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.uninit = uninit,
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FILTER_INPUTS(inputs),
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FILTER_OUTPUTS(outputs),
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FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
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.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
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AVFILTER_FLAG_SLICE_THREADS,
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.process_command = ff_filter_process_command,
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};
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@ -37,6 +37,7 @@ extern const AVFilter ff_af_adecorrelate;
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extern const AVFilter ff_af_adelay;
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extern const AVFilter ff_af_adenorm;
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extern const AVFilter ff_af_aderivative;
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extern const AVFilter ff_af_adynamicequalizer;
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extern const AVFilter ff_af_adynamicsmooth;
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extern const AVFilter ff_af_aecho;
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extern const AVFilter ff_af_aemphasis;
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@ -30,7 +30,7 @@
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#include "libavutil/version.h"
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#define LIBAVFILTER_VERSION_MAJOR 8
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#define LIBAVFILTER_VERSION_MINOR 19
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#define LIBAVFILTER_VERSION_MINOR 20
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#define LIBAVFILTER_VERSION_MICRO 100
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