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https://github.com/FFmpeg/FFmpeg.git
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avfilter: add high-order parametric multiband equalizer filter
Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol
parent
80508178e1
commit
56c7d2b4da
@ -47,6 +47,7 @@ version <next>:
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- DXVA2-accelerated VP9 decoding
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- SOFAlizer: virtual binaural acoustics filter
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- VAAPI VP9 hwaccel
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- audio high-order multiband parametric equalizer
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version 2.8:
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1
configure
vendored
1
configure
vendored
@ -2841,6 +2841,7 @@ unix_protocol_select="network"
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# filters
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amovie_filter_deps="avcodec avformat"
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anequalizer_filter_deps="cabs cexp"
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aresample_filter_deps="swresample"
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ass_filter_deps="libass"
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asyncts_filter_deps="avresample"
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@ -992,6 +992,100 @@ stream ends. The default value is 2 seconds.
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@end table
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@section anequalizer
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High-order parametric multiband equalizer for each channel.
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It accepts the following parameters:
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@table @option
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@item params
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This option string is in format:
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"c@var{chn} f=@var{cf} w=@var{w} g=@var{g} t=@var{f} | ..."
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Each equalizer band is separated by '|'.
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@table @option
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@item chn
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Set channel number to which equalization will be applied.
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If input doesn't have that channel the entry is ignored.
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@item cf
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Set central frequency for band.
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If input doesn't have that frequency the entry is ignored.
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@item w
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Set band width in hertz.
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@item g
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Set band gain in dB.
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@item f
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Set filter type for band, optional, can be:
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@table @samp
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@item 0
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Butterworth, this is default.
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@item 1
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Chebyshev type 1.
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@item 2
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Chebyshev type 2.
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@end table
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@end table
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@item curves
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With this option activated frequency response of anequalizer is displayed
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in video stream.
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@item size
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Set video stream size. Only useful if curves option is activated.
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@item mgain
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Set max gain that will be displayed. Only useful if curves option is activated.
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Setting this to reasonable value allows to display gain which is derived from
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neighbour bands which are too close to each other and thus produce higher gain
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when both are activated.
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@item fscale
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Set frequency scale used to draw frequency response in video output.
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Can be linear or logarithmic. Default is logarithmic.
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@item colors
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Set color for each channel curve which is going to be displayed in video stream.
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This is list of color names separated by space or by '|'.
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Unrecognised or missing colors will be replaced by white color.
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@end table
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@subsection Examples
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@itemize
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@item
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Lower gain by 10 of central frequency 200Hz and width 100 Hz
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for first 2 channels using Chebyshev type 1 filter:
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@example
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anequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1
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@end example
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@end itemize
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@subsection Commands
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This filter supports the following commands:
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@table @option
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@item change
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Alter existing filter parameters.
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Syntax for the commands is : "@var{fN}|f=@var{freq}|w=@var{width}|g=@var{gain}"
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@var{fN} is existing filter number, starting from 0, if no such filter is available
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error is returned.
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@var{freq} set new frequency parameter.
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@var{width} set new width parameter in herz.
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@var{gain} set new gain parameter in dB.
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Full filter invocation with asendcmd may look like this:
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asendcmd=c='4.0 anequalizer change 0|f=200|w=50|g=1',anequalizer=...
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@end table
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@section anull
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Pass the audio source unchanged to the output.
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@ -29,6 +29,7 @@ OBJS-$(CONFIG_ACROSSFADE_FILTER) += af_afade.o
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OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.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-$(CONFIG_ANEQUALIZER_FILTER) += af_anequalizer.o
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OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o
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OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
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OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
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751
libavfilter/af_anequalizer.c
Normal file
751
libavfilter/af_anequalizer.c
Normal file
@ -0,0 +1,751 @@
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/*
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* Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
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* Copyright (c) 2015 Paul B Mahol
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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 <complex.h>
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#include "libavutil/intreadwrite.h"
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#include "libavutil/avstring.h"
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#include "libavutil/opt.h"
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#include "libavutil/parseutils.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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#define FILTER_ORDER 4
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enum FilterType {
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BUTTERWORTH,
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CHEBYSHEV1,
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CHEBYSHEV2,
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NB_TYPES
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};
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typedef struct FoSection {
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double a0, a1, a2, a3, a4;
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double b0, b1, b2, b3, b4;
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double num[4];
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double denum[4];
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} FoSection;
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typedef struct EqualizatorFilter {
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int ignore;
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int channel;
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int type;
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double freq;
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double gain;
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double width;
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FoSection section[2];
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} EqualizatorFilter;
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typedef struct AudioNEqualizerContext {
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const AVClass *class;
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char *args;
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char *colors;
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int draw_curves;
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int w, h;
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double mag;
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int fscale;
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int nb_filters;
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int nb_allocated;
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EqualizatorFilter *filters;
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AVFrame *video;
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} AudioNEqualizerContext;
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#define OFFSET(x) offsetof(AudioNEqualizerContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM
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#define V AV_OPT_FLAG_VIDEO_PARAM
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#define F AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption anequalizer_options[] = {
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{ "params", NULL, OFFSET(args), AV_OPT_TYPE_STRING, {.str=""}, 0, 0, A|F },
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{ "curves", "draw frequency response curves", OFFSET(draw_curves), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, V|F },
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{ "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, V|F },
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{ "mgain", "set max gain", OFFSET(mag), AV_OPT_TYPE_DOUBLE, {.dbl=60}, -900, 900, V|F },
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{ "fscale", "set frequency scale", OFFSET(fscale), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, V|F, "fscale" },
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{ "lin", "linear", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, V|F, "fscale" },
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{ "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, V|F, "fscale" },
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{ "colors", "set channels curves colors", OFFSET(colors), AV_OPT_TYPE_STRING, {.str = "red|green|blue|yellow|orange|lime|pink|magenta|brown" }, 0, 0, V|F },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(anequalizer);
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static void draw_curves(AVFilterContext *ctx, AVFilterLink *inlink, AVFrame *out)
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{
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AudioNEqualizerContext *s = ctx->priv;
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char *colors, *color, *saveptr = NULL;
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int ch, i, n;
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colors = av_strdup(s->colors);
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if (!colors)
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return;
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memset(out->data[0], 0, s->h * out->linesize[0]);
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for (ch = 0; ch < inlink->channels; ch++) {
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uint8_t fg[4] = { 0xff, 0xff, 0xff, 0xff };
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int prev_v = -1;
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double f;
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color = av_strtok(ch == 0 ? colors : NULL, " |", &saveptr);
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if (color)
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av_parse_color(fg, color, -1, ctx);
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for (f = 0; f < s->w; f++) {
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double complex z;
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double complex H = 1;
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double w;
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int v, y, x;
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w = M_PI * (s->fscale ? pow(s->w - 1, f / s->w) : f) / (s->w - 1);
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z = 1. / cexp(I * w);
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for (n = 0; n < s->nb_filters; n++) {
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if (s->filters[n].channel != ch ||
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s->filters[n].ignore)
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continue;
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for (i = 0; i < FILTER_ORDER / 2; i++) {
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FoSection *S = &s->filters[n].section[i];
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H *= (((((S->b4 * z + S->b3) * z + S->b2) * z + S->b1) * z + S->b0) /
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((((S->a4 * z + S->a3) * z + S->a2) * z + S->a1) * z + S->a0));
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}
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}
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v = av_clip((1. + -20 * log10(cabs(H)) / s->mag) * s->h / 2, 0, s->h - 1);
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x = lrint(f);
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if (prev_v == -1)
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prev_v = v;
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if (v <= prev_v) {
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for (y = v; y <= prev_v; y++)
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AV_WL32(out->data[0] + y * out->linesize[0] + x * 4, AV_RL32(fg));
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} else {
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for (y = prev_v; y <= v; y++)
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AV_WL32(out->data[0] + y * out->linesize[0] + x * 4, AV_RL32(fg));
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}
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prev_v = v;
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}
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}
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av_free(colors);
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}
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static int config_video(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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AudioNEqualizerContext *s = ctx->priv;
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AVFilterLink *inlink = ctx->inputs[0];
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AVFrame *out;
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outlink->w = s->w;
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outlink->h = s->h;
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av_frame_free(&s->video);
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s->video = out = ff_get_video_buffer(outlink, outlink->w, outlink->h);
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if (!out)
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return AVERROR(ENOMEM);
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outlink->sample_aspect_ratio = (AVRational){1,1};
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draw_curves(ctx, inlink, out);
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return 0;
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}
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static av_cold int init(AVFilterContext *ctx)
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{
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AudioNEqualizerContext *s = ctx->priv;
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AVFilterPad pad, vpad;
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pad = (AVFilterPad){
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.name = av_strdup("out0"),
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.type = AVMEDIA_TYPE_AUDIO,
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};
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if (!pad.name)
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return AVERROR(ENOMEM);
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if (s->draw_curves) {
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vpad = (AVFilterPad){
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.name = av_strdup("out1"),
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.type = AVMEDIA_TYPE_VIDEO,
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.config_props = config_video,
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};
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if (!vpad.name)
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return AVERROR(ENOMEM);
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}
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ff_insert_outpad(ctx, 0, &pad);
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if (s->draw_curves)
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ff_insert_outpad(ctx, 1, &vpad);
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return 0;
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}
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterLink *inlink = ctx->inputs[0];
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AVFilterLink *outlink = ctx->outputs[0];
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AudioNEqualizerContext *s = ctx->priv;
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AVFilterFormats *formats;
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AVFilterChannelLayouts *layouts;
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static const enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_RGBA, AV_PIX_FMT_NONE };
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_NONE
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};
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int ret;
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if (s->draw_curves) {
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AVFilterLink *videolink = ctx->outputs[1];
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formats = ff_make_format_list(pix_fmts);
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if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0)
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return ret;
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}
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formats = ff_make_format_list(sample_fmts);
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if ((ret = ff_formats_ref(formats, &inlink->out_formats)) < 0 ||
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(ret = ff_formats_ref(formats, &outlink->in_formats)) < 0)
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return ret;
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layouts = ff_all_channel_counts();
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if ((ret = ff_channel_layouts_ref(layouts, &inlink->out_channel_layouts)) < 0 ||
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(ret = ff_channel_layouts_ref(layouts, &outlink->in_channel_layouts)) < 0)
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return ret;
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formats = ff_all_samplerates();
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if ((ret = ff_formats_ref(formats, &inlink->out_samplerates)) < 0 ||
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(ret = ff_formats_ref(formats, &outlink->in_samplerates)) < 0)
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return ret;
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return 0;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AudioNEqualizerContext *s = ctx->priv;
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av_freep(&ctx->output_pads[0].name);
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if (s->draw_curves)
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av_freep(&ctx->output_pads[1].name);
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av_frame_free(&s->video);
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av_freep(&s->filters);
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s->nb_filters = 0;
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s->nb_allocated = 0;
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}
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static void butterworth_fo_section(FoSection *S, double beta,
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double si, double g, double g0,
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double D, double c0)
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{
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if (c0 == 1 || c0 == -1) {
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S->b0 = (g*g*beta*beta + 2*g*g0*si*beta + g0*g0)/D;
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S->b1 = 2*c0*(g*g*beta*beta - g0*g0)/D;
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S->b2 = (g*g*beta*beta - 2*g0*g*beta*si + g0*g0)/D;
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S->b3 = 0;
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S->b4 = 0;
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S->a0 = 1;
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S->a1 = 2*c0*(beta*beta - 1)/D;
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S->a2 = (beta*beta - 2*beta*si + 1)/D;
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S->a3 = 0;
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S->a4 = 0;
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} else {
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S->b0 = (g*g*beta*beta + 2*g*g0*si*beta + g0*g0)/D;
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S->b1 = -4*c0*(g0*g0 + g*g0*si*beta)/D;
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S->b2 = 2*(g0*g0*(1 + 2*c0*c0) - g*g*beta*beta)/D;
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S->b3 = -4*c0*(g0*g0 - g*g0*si*beta)/D;
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S->b4 = (g*g*beta*beta - 2*g*g0*si*beta + g0*g0)/D;
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S->a0 = 1;
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S->a1 = -4*c0*(1 + si*beta)/D;
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S->a2 = 2*(1 + 2*c0*c0 - beta*beta)/D;
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S->a3 = -4*c0*(1 - si*beta)/D;
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S->a4 = (beta*beta - 2*si*beta + 1)/D;
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}
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}
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static void butterworth_bp_filter(EqualizatorFilter *f,
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int N, double w0, double wb,
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double G, double Gb, double G0)
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{
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double g, c0, g0, beta;
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double epsilon;
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int r = N % 2;
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int L = (N - r) / 2;
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int i;
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if (G == 0 && G0 == 0) {
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f->section[0].a0 = 1;
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f->section[0].b0 = 1;
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f->section[1].a0 = 1;
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f->section[1].b0 = 1;
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return;
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}
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G = exp10( G/20);
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Gb = exp10(Gb/20);
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G0 = exp10(G0/20);
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epsilon = sqrt((G * G - Gb * Gb) / (Gb * Gb - G0 * G0));
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g = pow(G, 1.0 / N);
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g0 = pow(G0, 1.0 / N);
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beta = pow(epsilon, -1.0 / N) * tan(wb/2);
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c0 = cos(w0);
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for (i = 1; i <= L; i++) {
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double ui = (2.0 * i - 1) / N;
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double si = sin(M_PI * ui / 2.0);
|
||||
double Di = beta * beta + 2 * si * beta + 1;
|
||||
|
||||
butterworth_fo_section(&f->section[i - 1], beta, si, g, g0, Di, c0);
|
||||
}
|
||||
}
|
||||
|
||||
static void chebyshev1_fo_section(FoSection *S, double a,
|
||||
double c, double tetta_b,
|
||||
double g0, double si, double b,
|
||||
double D, double c0)
|
||||
{
|
||||
if (c0 == 1 || c0 == -1) {
|
||||
S->b0 = (tetta_b*tetta_b*(b*b+g0*g0*c*c) + 2*g0*b*si*tetta_b*tetta_b + g0*g0)/D;
|
||||
S->b1 = 2*c0*(tetta_b*tetta_b*(b*b+g0*g0*c*c) - g0*g0)/D;
|
||||
S->b2 = (tetta_b*tetta_b*(b*b+g0*g0*c*c) - 2*g0*b*si*tetta_b + g0*g0)/D;
|
||||
S->b3 = 0;
|
||||
S->b4 = 0;
|
||||
|
||||
S->a0 = 1;
|
||||
S->a1 = 2*c0*(tetta_b*tetta_b*(a*a+c*c) - 1)/D;
|
||||
S->a2 = (tetta_b*tetta_b*(a*a+c*c) - 2*a*si*tetta_b + 1)/D;
|
||||
S->a3 = 0;
|
||||
S->a4 = 0;
|
||||
} else {
|
||||
S->b0 = ((b*b + g0*g0*c*c)*tetta_b*tetta_b + 2*g0*b*si*tetta_b + g0*g0)/D;
|
||||
S->b1 = -4*c0*(g0*g0 + g0*b*si*tetta_b)/D;
|
||||
S->b2 = 2*(g0*g0*(1 + 2*c0*c0) - (b*b + g0*g0*c*c)*tetta_b*tetta_b)/D;
|
||||
S->b3 = -4*c0*(g0*g0 - g0*b*si*tetta_b)/D;
|
||||
S->b4 = ((b*b + g0*g0*c*c)*tetta_b*tetta_b - 2*g0*b*si*tetta_b + g0*g0)/D;
|
||||
|
||||
S->a0 = 1;
|
||||
S->a1 = -4*c0*(1 + a*si*tetta_b)/D;
|
||||
S->a2 = 2*(1 + 2*c0*c0 - (a*a + c*c)*tetta_b*tetta_b)/D;
|
||||
S->a3 = -4*c0*(1 - a*si*tetta_b)/D;
|
||||
S->a4 = ((a*a + c*c)*tetta_b*tetta_b - 2*a*si*tetta_b + 1)/D;
|
||||
}
|
||||
}
|
||||
|
||||
static void chebyshev1_bp_filter(EqualizatorFilter *f,
|
||||
int N, double w0, double wb,
|
||||
double G, double Gb, double G0)
|
||||
{
|
||||
double a, b, c0, g0, alfa, beta, tetta_b;
|
||||
double epsilon;
|
||||
int r = N % 2;
|
||||
int L = (N - r) / 2;
|
||||
int i;
|
||||
|
||||
if (G == 0 && G0 == 0) {
|
||||
f->section[0].a0 = 1;
|
||||
f->section[0].b0 = 1;
|
||||
f->section[1].a0 = 1;
|
||||
f->section[1].b0 = 1;
|
||||
return;
|
||||
}
|
||||
|
||||
G = exp10( G/20);
|
||||
Gb = exp10(Gb/20);
|
||||
G0 = exp10(G0/20);
|
||||
|
||||
epsilon = sqrt((G*G - Gb*Gb) / (Gb*Gb - G0*G0));
|
||||
g0 = pow(G0,1.0/N);
|
||||
alfa = pow(1.0/epsilon + sqrt(1 + pow(epsilon,-2.0)), 1.0/N);
|
||||
beta = pow(G/epsilon + Gb * sqrt(1 + pow(epsilon,-2.0)), 1.0/N);
|
||||
a = 0.5 * (alfa - 1.0/alfa);
|
||||
b = 0.5 * (beta - g0*g0*(1/beta));
|
||||
tetta_b = tan(wb/2);
|
||||
c0 = cos(w0);
|
||||
|
||||
for (i = 1; i <= L; i++) {
|
||||
double ui = (2.0*i-1.0)/N;
|
||||
double ci = cos(M_PI*ui/2.0);
|
||||
double si = sin(M_PI*ui/2.0);
|
||||
double Di = (a*a + ci*ci)*tetta_b*tetta_b + 2.0*a*si*tetta_b + 1;
|
||||
|
||||
chebyshev1_fo_section(&f->section[i - 1], a, ci, tetta_b, g0, si, b, Di, c0);
|
||||
}
|
||||
}
|
||||
|
||||
static void chebyshev2_fo_section(FoSection *S, double a,
|
||||
double c, double tetta_b,
|
||||
double g, double si, double b,
|
||||
double D, double c0)
|
||||
{
|
||||
if (c0 == 1 || c0 == -1) {
|
||||
S->b0 = (g*g*tetta_b*tetta_b + 2*tetta_b*g*b*si + b*b + g*g*c*c)/D;
|
||||
S->b1 = 2*c0*(g*g*tetta_b*tetta_b - b*b - g*g*c*c)/D;
|
||||
S->b2 = (g*g*tetta_b*tetta_b - 2*tetta_b*g*b*si + b*b + g*g*c*c)/D;
|
||||
S->b3 = 0;
|
||||
S->b4 = 0;
|
||||
|
||||
S->a0 = 1;
|
||||
S->a1 = 2*c0*(tetta_b*tetta_b - a*a - c*c)/D;
|
||||
S->a2 = (tetta_b*tetta_b - 2*tetta_b*a*si + a*a + c*c)/D;
|
||||
S->a3 = 0;
|
||||
S->a4 = 0;
|
||||
} else {
|
||||
S->b0 = (g*g*tetta_b*tetta_b + 2*g*b*si*tetta_b + b*b + g*g*c*c)/D;
|
||||
S->b1 = -4*c0*(b*b + g*g*c*c + g*b*si*tetta_b)/D;
|
||||
S->b2 = 2*((b*b + g*g*c*c)*(1 + 2*c0*c0) - g*g*tetta_b*tetta_b)/D;
|
||||
S->b3 = -4*c0*(b*b + g*g*c*c - g*b*si*tetta_b)/D;
|
||||
S->b4 = (g*g*tetta_b*tetta_b - 2*g*b*si*tetta_b + b*b + g*g*c*c)/D;
|
||||
|
||||
S->a0 = 1;
|
||||
S->a1 = -4*c0*(a*a + c*c + a*si*tetta_b)/D;
|
||||
S->a2 = 2*((a*a + c*c)*(1 + 2*c0*c0) - tetta_b*tetta_b)/D;
|
||||
S->a3 = -4*c0*(a*a + c*c - a*si*tetta_b)/D;
|
||||
S->a4 = (tetta_b*tetta_b - 2*a*si*tetta_b + a*a + c*c)/D;
|
||||
}
|
||||
}
|
||||
|
||||
static void chebyshev2_bp_filter(EqualizatorFilter *f,
|
||||
int N, double w0, double wb,
|
||||
double G, double Gb, double G0)
|
||||
{
|
||||
double a, b, c0, tetta_b;
|
||||
double epsilon, g, eu, ew;
|
||||
int r = N % 2;
|
||||
int L = (N - r) / 2;
|
||||
int i;
|
||||
|
||||
if (G == 0 && G0 == 0) {
|
||||
f->section[0].a0 = 1;
|
||||
f->section[0].b0 = 1;
|
||||
f->section[1].a0 = 1;
|
||||
f->section[1].b0 = 1;
|
||||
return;
|
||||
}
|
||||
|
||||
G = exp10( G/20);
|
||||
Gb = exp10(Gb/20);
|
||||
G0 = exp10(G0/20);
|
||||
|
||||
epsilon = sqrt((G*G - Gb*Gb) / (Gb*Gb - G0*G0));
|
||||
g = pow(G, 1.0 / N);
|
||||
eu = pow(epsilon + sqrt(1 + epsilon*epsilon), 1.0/N);
|
||||
ew = pow(G0*epsilon + Gb*sqrt(1 + epsilon*epsilon), 1.0/N);
|
||||
a = (eu - 1.0/eu)/2.0;
|
||||
b = (ew - g*g/ew)/2.0;
|
||||
tetta_b = tan(wb/2);
|
||||
c0 = cos(w0);
|
||||
|
||||
for (i = 1; i <= L; i++) {
|
||||
double ui = (2.0 * i - 1.0)/N;
|
||||
double ci = cos(M_PI * ui / 2.0);
|
||||
double si = sin(M_PI * ui / 2.0);
|
||||
double Di = tetta_b*tetta_b + 2*a*si*tetta_b + a*a + ci*ci;
|
||||
|
||||
chebyshev2_fo_section(&f->section[i - 1], a, ci, tetta_b, g, si, b, Di, c0);
|
||||
}
|
||||
}
|
||||
|
||||
static double butterworth_compute_bw_gain_db(double gain)
|
||||
{
|
||||
double bw_gain = 0;
|
||||
|
||||
if (gain <= -6)
|
||||
bw_gain = gain + 3;
|
||||
else if(gain > -6 && gain < 6)
|
||||
bw_gain = gain * 0.5;
|
||||
else if(gain >= 6)
|
||||
bw_gain = gain - 3;
|
||||
|
||||
return bw_gain;
|
||||
}
|
||||
|
||||
static double chebyshev1_compute_bw_gain_db(double gain)
|
||||
{
|
||||
double bw_gain = 0;
|
||||
|
||||
if (gain <= -6)
|
||||
bw_gain = gain + 1;
|
||||
else if(gain > -6 && gain < 6)
|
||||
bw_gain = gain * 0.9;
|
||||
else if(gain >= 6)
|
||||
bw_gain = gain - 1;
|
||||
|
||||
return bw_gain;
|
||||
}
|
||||
|
||||
static double chebyshev2_compute_bw_gain_db(double gain)
|
||||
{
|
||||
double bw_gain = 0;
|
||||
|
||||
if (gain <= -6)
|
||||
bw_gain = -3;
|
||||
else if(gain > -6 && gain < 6)
|
||||
bw_gain = gain * 0.3;
|
||||
else if(gain >= 6)
|
||||
bw_gain = 3;
|
||||
|
||||
return bw_gain;
|
||||
}
|
||||
|
||||
static inline double hz_2_rad(double x, double fs)
|
||||
{
|
||||
return 2 * M_PI * x / fs;
|
||||
}
|
||||
|
||||
static void equalizer(EqualizatorFilter *f, double sample_rate)
|
||||
{
|
||||
double w0 = hz_2_rad(f->freq, sample_rate);
|
||||
double wb = hz_2_rad(f->width, sample_rate);
|
||||
double bw_gain;
|
||||
|
||||
switch (f->type) {
|
||||
case BUTTERWORTH:
|
||||
bw_gain = butterworth_compute_bw_gain_db(f->gain);
|
||||
butterworth_bp_filter(f, FILTER_ORDER, w0, wb, f->gain, bw_gain, 0);
|
||||
break;
|
||||
case CHEBYSHEV1:
|
||||
bw_gain = chebyshev1_compute_bw_gain_db(f->gain);
|
||||
chebyshev1_bp_filter(f, FILTER_ORDER, w0, wb, f->gain, bw_gain, 0);
|
||||
break;
|
||||
case CHEBYSHEV2:
|
||||
bw_gain = chebyshev2_compute_bw_gain_db(f->gain);
|
||||
chebyshev2_bp_filter(f, FILTER_ORDER, w0, wb, f->gain, bw_gain, 0);
|
||||
break;
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
static int add_filter(AudioNEqualizerContext *s, AVFilterLink *inlink)
|
||||
{
|
||||
equalizer(&s->filters[s->nb_filters], inlink->sample_rate);
|
||||
if (s->nb_filters >= s->nb_allocated) {
|
||||
EqualizatorFilter *filters;
|
||||
|
||||
filters = av_calloc(s->nb_allocated, 2 * sizeof(*s->filters));
|
||||
if (!filters)
|
||||
return AVERROR(ENOMEM);
|
||||
memcpy(filters, s->filters, sizeof(*s->filters) * s->nb_allocated);
|
||||
av_free(s->filters);
|
||||
s->filters = filters;
|
||||
s->nb_allocated *= 2;
|
||||
}
|
||||
s->nb_filters++;
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int config_input(AVFilterLink *inlink)
|
||||
{
|
||||
AVFilterContext *ctx = inlink->dst;
|
||||
AudioNEqualizerContext *s = ctx->priv;
|
||||
char *args = av_strdup(s->args);
|
||||
char *saveptr = NULL;
|
||||
int ret = 0;
|
||||
|
||||
if (!args)
|
||||
return AVERROR(ENOMEM);
|
||||
|
||||
s->nb_allocated = 32 * inlink->channels;
|
||||
s->filters = av_calloc(inlink->channels, 32 * sizeof(*s->filters));
|
||||
if (!s->filters) {
|
||||
s->nb_allocated = 0;
|
||||
return AVERROR(ENOMEM);
|
||||
}
|
||||
|
||||
while (1) {
|
||||
char *arg = av_strtok(s->nb_filters == 0 ? args : NULL, "|", &saveptr);
|
||||
|
||||
if (!arg)
|
||||
break;
|
||||
|
||||
s->filters[s->nb_filters].type = 0;
|
||||
if (sscanf(arg, "c%d f=%lf w=%lf g=%lf t=%d", &s->filters[s->nb_filters].channel,
|
||||
&s->filters[s->nb_filters].freq,
|
||||
&s->filters[s->nb_filters].width,
|
||||
&s->filters[s->nb_filters].gain,
|
||||
&s->filters[s->nb_filters].type) != 5 &&
|
||||
sscanf(arg, "c%d f=%lf w=%lf g=%lf", &s->filters[s->nb_filters].channel,
|
||||
&s->filters[s->nb_filters].freq,
|
||||
&s->filters[s->nb_filters].width,
|
||||
&s->filters[s->nb_filters].gain) != 4 ) {
|
||||
av_free(args);
|
||||
return AVERROR(EINVAL);
|
||||
}
|
||||
|
||||
if (s->filters[s->nb_filters].freq < 0 ||
|
||||
s->filters[s->nb_filters].freq > inlink->sample_rate / 2)
|
||||
s->filters[s->nb_filters].ignore = 1;
|
||||
|
||||
if (s->filters[s->nb_filters].channel < 0 ||
|
||||
s->filters[s->nb_filters].channel >= inlink->channels)
|
||||
s->filters[s->nb_filters].ignore = 1;
|
||||
|
||||
av_clip(s->filters[s->nb_filters].type, 0, NB_TYPES - 1);
|
||||
ret = add_filter(s, inlink);
|
||||
if (ret < 0)
|
||||
break;
|
||||
}
|
||||
|
||||
av_free(args);
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
|
||||
char *res, int res_len, int flags)
|
||||
{
|
||||
AudioNEqualizerContext *s = ctx->priv;
|
||||
AVFilterLink *inlink = ctx->inputs[0];
|
||||
int ret = AVERROR(ENOSYS);
|
||||
|
||||
if (!strcmp(cmd, "change")) {
|
||||
double freq, width, gain;
|
||||
int filter;
|
||||
|
||||
if (sscanf(args, "%d|f=%lf|w=%lf|g=%lf", &filter, &freq, &width, &gain) != 4)
|
||||
return AVERROR(EINVAL);
|
||||
|
||||
if (filter < 0 || filter >= s->nb_filters)
|
||||
return AVERROR(EINVAL);
|
||||
|
||||
if (freq < 0 || freq > inlink->sample_rate / 2)
|
||||
return AVERROR(EINVAL);
|
||||
|
||||
s->filters[filter].freq = freq;
|
||||
s->filters[filter].width = width;
|
||||
s->filters[filter].gain = gain;
|
||||
equalizer(&s->filters[filter], inlink->sample_rate);
|
||||
if (s->draw_curves)
|
||||
draw_curves(ctx, inlink, s->video);
|
||||
|
||||
ret = 0;
|
||||
}
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
static inline double section_process(FoSection *S, double in)
|
||||
{
|
||||
double out;
|
||||
|
||||
out = S->b0 * in;
|
||||
out+= S->b1 * S->num[0] - S->denum[0] * S->a1;
|
||||
out+= S->b2 * S->num[1] - S->denum[1] * S->a2;
|
||||
out+= S->b3 * S->num[2] - S->denum[2] * S->a3;
|
||||
out+= S->b4 * S->num[3] - S->denum[3] * S->a4;
|
||||
|
||||
S->num[3] = S->num[2];
|
||||
S->num[2] = S->num[1];
|
||||
S->num[1] = S->num[0];
|
||||
S->num[0] = in;
|
||||
|
||||
S->denum[3] = S->denum[2];
|
||||
S->denum[2] = S->denum[1];
|
||||
S->denum[1] = S->denum[0];
|
||||
S->denum[0] = out;
|
||||
|
||||
return out;
|
||||
}
|
||||
|
||||
static double process_sample(FoSection *s1, double in)
|
||||
{
|
||||
double p0 = in, p1;
|
||||
int i;
|
||||
|
||||
for (i = 0; i < FILTER_ORDER / 2; i++) {
|
||||
p1 = section_process(&s1[i], p0);
|
||||
p0 = p1;
|
||||
}
|
||||
|
||||
return p1;
|
||||
}
|
||||
|
||||
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
|
||||
{
|
||||
AVFilterContext *ctx = inlink->dst;
|
||||
AudioNEqualizerContext *s = ctx->priv;
|
||||
AVFilterLink *outlink = ctx->outputs[0];
|
||||
double *bptr;
|
||||
int i, n;
|
||||
|
||||
for (i = 0; i < s->nb_filters; i++) {
|
||||
EqualizatorFilter *f = &s->filters[i];
|
||||
|
||||
if (f->gain == 0. || f->ignore)
|
||||
continue;
|
||||
|
||||
bptr = (double *)buf->extended_data[f->channel];
|
||||
for (n = 0; n < buf->nb_samples; n++) {
|
||||
double sample = bptr[n];
|
||||
|
||||
sample = process_sample(f->section, sample);
|
||||
bptr[n] = sample;
|
||||
}
|
||||
}
|
||||
|
||||
if (s->draw_curves) {
|
||||
const int64_t pts = buf->pts +
|
||||
av_rescale_q(buf->nb_samples, (AVRational){ 1, inlink->sample_rate },
|
||||
outlink->time_base);
|
||||
int ret;
|
||||
|
||||
s->video->pts = pts;
|
||||
ret = ff_filter_frame(ctx->outputs[1], av_frame_clone(s->video));
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
}
|
||||
|
||||
return ff_filter_frame(outlink, buf);
|
||||
}
|
||||
|
||||
static const AVFilterPad inputs[] = {
|
||||
{
|
||||
.name = "default",
|
||||
.type = AVMEDIA_TYPE_AUDIO,
|
||||
.config_props = config_input,
|
||||
.filter_frame = filter_frame,
|
||||
.needs_writable = 1,
|
||||
},
|
||||
{ NULL }
|
||||
};
|
||||
|
||||
AVFilter ff_af_anequalizer = {
|
||||
.name = "anequalizer",
|
||||
.description = NULL_IF_CONFIG_SMALL("Apply high-order audio parametric multi band equalizer."),
|
||||
.priv_size = sizeof(AudioNEqualizerContext),
|
||||
.priv_class = &anequalizer_class,
|
||||
.init = init,
|
||||
.uninit = uninit,
|
||||
.query_formats = query_formats,
|
||||
.inputs = inputs,
|
||||
.outputs = NULL,
|
||||
.flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS,
|
||||
.process_command = process_command,
|
||||
};
|
||||
@ -59,6 +59,7 @@ void avfilter_register_all(void)
|
||||
REGISTER_FILTER(ALLPASS, allpass, af);
|
||||
REGISTER_FILTER(AMERGE, amerge, af);
|
||||
REGISTER_FILTER(AMIX, amix, af);
|
||||
REGISTER_FILTER(ANEQUALIZER, anequalizer, af);
|
||||
REGISTER_FILTER(ANULL, anull, af);
|
||||
REGISTER_FILTER(APAD, apad, af);
|
||||
REGISTER_FILTER(APERMS, aperms, af);
|
||||
|
||||
@ -30,8 +30,8 @@
|
||||
#include "libavutil/version.h"
|
||||
|
||||
#define LIBAVFILTER_VERSION_MAJOR 6
|
||||
#define LIBAVFILTER_VERSION_MINOR 21
|
||||
#define LIBAVFILTER_VERSION_MICRO 101
|
||||
#define LIBAVFILTER_VERSION_MINOR 22
|
||||
#define LIBAVFILTER_VERSION_MICRO 100
|
||||
|
||||
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
|
||||
LIBAVFILTER_VERSION_MINOR, \
|
||||
|
||||
Loading…
Reference in New Issue
Block a user