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 2015-12-20 09:50:35 +01:00
parent 80508178e1
commit 56c7d2b4da
7 changed files with 851 additions and 2 deletions

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@ -47,6 +47,7 @@ version <next>:
- DXVA2-accelerated VP9 decoding
- SOFAlizer: virtual binaural acoustics filter
- VAAPI VP9 hwaccel
- audio high-order multiband parametric equalizer
version 2.8:

1
configure vendored
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@ -2841,6 +2841,7 @@ unix_protocol_select="network"
# filters
amovie_filter_deps="avcodec avformat"
anequalizer_filter_deps="cabs cexp"
aresample_filter_deps="swresample"
ass_filter_deps="libass"
asyncts_filter_deps="avresample"

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@ -992,6 +992,100 @@ stream ends. The default value is 2 seconds.
@end table
@section anequalizer
High-order parametric multiband equalizer for each channel.
It accepts the following parameters:
@table @option
@item params
This option string is in format:
"c@var{chn} f=@var{cf} w=@var{w} g=@var{g} t=@var{f} | ..."
Each equalizer band is separated by '|'.
@table @option
@item chn
Set channel number to which equalization will be applied.
If input doesn't have that channel the entry is ignored.
@item cf
Set central frequency for band.
If input doesn't have that frequency the entry is ignored.
@item w
Set band width in hertz.
@item g
Set band gain in dB.
@item f
Set filter type for band, optional, can be:
@table @samp
@item 0
Butterworth, this is default.
@item 1
Chebyshev type 1.
@item 2
Chebyshev type 2.
@end table
@end table
@item curves
With this option activated frequency response of anequalizer is displayed
in video stream.
@item size
Set video stream size. Only useful if curves option is activated.
@item mgain
Set max gain that will be displayed. Only useful if curves option is activated.
Setting this to reasonable value allows to display gain which is derived from
neighbour bands which are too close to each other and thus produce higher gain
when both are activated.
@item fscale
Set frequency scale used to draw frequency response in video output.
Can be linear or logarithmic. Default is logarithmic.
@item colors
Set color for each channel curve which is going to be displayed in video stream.
This is list of color names separated by space or by '|'.
Unrecognised or missing colors will be replaced by white color.
@end table
@subsection Examples
@itemize
@item
Lower gain by 10 of central frequency 200Hz and width 100 Hz
for first 2 channels using Chebyshev type 1 filter:
@example
anequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1
@end example
@end itemize
@subsection Commands
This filter supports the following commands:
@table @option
@item change
Alter existing filter parameters.
Syntax for the commands is : "@var{fN}|f=@var{freq}|w=@var{width}|g=@var{gain}"
@var{fN} is existing filter number, starting from 0, if no such filter is available
error is returned.
@var{freq} set new frequency parameter.
@var{width} set new width parameter in herz.
@var{gain} set new gain parameter in dB.
Full filter invocation with asendcmd may look like this:
asendcmd=c='4.0 anequalizer change 0|f=200|w=50|g=1',anequalizer=...
@end table
@section anull
Pass the audio source unchanged to the output.

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@ -29,6 +29,7 @@ OBJS-$(CONFIG_ACROSSFADE_FILTER) += af_afade.o
OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.o
OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o
OBJS-$(CONFIG_ANEQUALIZER_FILTER) += af_anequalizer.o
OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o
OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o

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@ -0,0 +1,751 @@
/*
* Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
* Copyright (c) 2015 Paul B Mahol
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <complex.h>
#include "libavutil/intreadwrite.h"
#include "libavutil/avstring.h"
#include "libavutil/opt.h"
#include "libavutil/parseutils.h"
#include "avfilter.h"
#include "internal.h"
#include "audio.h"
#define FILTER_ORDER 4
enum FilterType {
BUTTERWORTH,
CHEBYSHEV1,
CHEBYSHEV2,
NB_TYPES
};
typedef struct FoSection {
double a0, a1, a2, a3, a4;
double b0, b1, b2, b3, b4;
double num[4];
double denum[4];
} FoSection;
typedef struct EqualizatorFilter {
int ignore;
int channel;
int type;
double freq;
double gain;
double width;
FoSection section[2];
} EqualizatorFilter;
typedef struct AudioNEqualizerContext {
const AVClass *class;
char *args;
char *colors;
int draw_curves;
int w, h;
double mag;
int fscale;
int nb_filters;
int nb_allocated;
EqualizatorFilter *filters;
AVFrame *video;
} AudioNEqualizerContext;
#define OFFSET(x) offsetof(AudioNEqualizerContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM
#define V AV_OPT_FLAG_VIDEO_PARAM
#define F AV_OPT_FLAG_FILTERING_PARAM
static const AVOption anequalizer_options[] = {
{ "params", NULL, OFFSET(args), AV_OPT_TYPE_STRING, {.str=""}, 0, 0, A|F },
{ "curves", "draw frequency response curves", OFFSET(draw_curves), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, V|F },
{ "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, V|F },
{ "mgain", "set max gain", OFFSET(mag), AV_OPT_TYPE_DOUBLE, {.dbl=60}, -900, 900, V|F },
{ "fscale", "set frequency scale", OFFSET(fscale), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, V|F, "fscale" },
{ "lin", "linear", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, V|F, "fscale" },
{ "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, V|F, "fscale" },
{ "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 },
{ NULL }
};
AVFILTER_DEFINE_CLASS(anequalizer);
static void draw_curves(AVFilterContext *ctx, AVFilterLink *inlink, AVFrame *out)
{
AudioNEqualizerContext *s = ctx->priv;
char *colors, *color, *saveptr = NULL;
int ch, i, n;
colors = av_strdup(s->colors);
if (!colors)
return;
memset(out->data[0], 0, s->h * out->linesize[0]);
for (ch = 0; ch < inlink->channels; ch++) {
uint8_t fg[4] = { 0xff, 0xff, 0xff, 0xff };
int prev_v = -1;
double f;
color = av_strtok(ch == 0 ? colors : NULL, " |", &saveptr);
if (color)
av_parse_color(fg, color, -1, ctx);
for (f = 0; f < s->w; f++) {
double complex z;
double complex H = 1;
double w;
int v, y, x;
w = M_PI * (s->fscale ? pow(s->w - 1, f / s->w) : f) / (s->w - 1);
z = 1. / cexp(I * w);
for (n = 0; n < s->nb_filters; n++) {
if (s->filters[n].channel != ch ||
s->filters[n].ignore)
continue;
for (i = 0; i < FILTER_ORDER / 2; i++) {
FoSection *S = &s->filters[n].section[i];
H *= (((((S->b4 * z + S->b3) * z + S->b2) * z + S->b1) * z + S->b0) /
((((S->a4 * z + S->a3) * z + S->a2) * z + S->a1) * z + S->a0));
}
}
v = av_clip((1. + -20 * log10(cabs(H)) / s->mag) * s->h / 2, 0, s->h - 1);
x = lrint(f);
if (prev_v == -1)
prev_v = v;
if (v <= prev_v) {
for (y = v; y <= prev_v; y++)
AV_WL32(out->data[0] + y * out->linesize[0] + x * 4, AV_RL32(fg));
} else {
for (y = prev_v; y <= v; y++)
AV_WL32(out->data[0] + y * out->linesize[0] + x * 4, AV_RL32(fg));
}
prev_v = v;
}
}
av_free(colors);
}
static int config_video(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioNEqualizerContext *s = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
AVFrame *out;
outlink->w = s->w;
outlink->h = s->h;
av_frame_free(&s->video);
s->video = out = ff_get_video_buffer(outlink, outlink->w, outlink->h);
if (!out)
return AVERROR(ENOMEM);
outlink->sample_aspect_ratio = (AVRational){1,1};
draw_curves(ctx, inlink, out);
return 0;
}
static av_cold int init(AVFilterContext *ctx)
{
AudioNEqualizerContext *s = ctx->priv;
AVFilterPad pad, vpad;
pad = (AVFilterPad){
.name = av_strdup("out0"),
.type = AVMEDIA_TYPE_AUDIO,
};
if (!pad.name)
return AVERROR(ENOMEM);
if (s->draw_curves) {
vpad = (AVFilterPad){
.name = av_strdup("out1"),
.type = AVMEDIA_TYPE_VIDEO,
.config_props = config_video,
};
if (!vpad.name)
return AVERROR(ENOMEM);
}
ff_insert_outpad(ctx, 0, &pad);
if (s->draw_curves)
ff_insert_outpad(ctx, 1, &vpad);
return 0;
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
AudioNEqualizerContext *s = ctx->priv;
AVFilterFormats *formats;
AVFilterChannelLayouts *layouts;
static const enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_RGBA, AV_PIX_FMT_NONE };
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
int ret;
if (s->draw_curves) {
AVFilterLink *videolink = ctx->outputs[1];
formats = ff_make_format_list(pix_fmts);
if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0)
return ret;
}
formats = ff_make_format_list(sample_fmts);
if ((ret = ff_formats_ref(formats, &inlink->out_formats)) < 0 ||
(ret = ff_formats_ref(formats, &outlink->in_formats)) < 0)
return ret;
layouts = ff_all_channel_counts();
if ((ret = ff_channel_layouts_ref(layouts, &inlink->out_channel_layouts)) < 0 ||
(ret = ff_channel_layouts_ref(layouts, &outlink->in_channel_layouts)) < 0)
return ret;
formats = ff_all_samplerates();
if ((ret = ff_formats_ref(formats, &inlink->out_samplerates)) < 0 ||
(ret = ff_formats_ref(formats, &outlink->in_samplerates)) < 0)
return ret;
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioNEqualizerContext *s = ctx->priv;
av_freep(&ctx->output_pads[0].name);
if (s->draw_curves)
av_freep(&ctx->output_pads[1].name);
av_frame_free(&s->video);
av_freep(&s->filters);
s->nb_filters = 0;
s->nb_allocated = 0;
}
static void butterworth_fo_section(FoSection *S, double beta,
double si, double g, double g0,
double D, double c0)
{
if (c0 == 1 || c0 == -1) {
S->b0 = (g*g*beta*beta + 2*g*g0*si*beta + g0*g0)/D;
S->b1 = 2*c0*(g*g*beta*beta - g0*g0)/D;
S->b2 = (g*g*beta*beta - 2*g0*g*beta*si + g0*g0)/D;
S->b3 = 0;
S->b4 = 0;
S->a0 = 1;
S->a1 = 2*c0*(beta*beta - 1)/D;
S->a2 = (beta*beta - 2*beta*si + 1)/D;
S->a3 = 0;
S->a4 = 0;
} else {
S->b0 = (g*g*beta*beta + 2*g*g0*si*beta + g0*g0)/D;
S->b1 = -4*c0*(g0*g0 + g*g0*si*beta)/D;
S->b2 = 2*(g0*g0*(1 + 2*c0*c0) - g*g*beta*beta)/D;
S->b3 = -4*c0*(g0*g0 - g*g0*si*beta)/D;
S->b4 = (g*g*beta*beta - 2*g*g0*si*beta + g0*g0)/D;
S->a0 = 1;
S->a1 = -4*c0*(1 + si*beta)/D;
S->a2 = 2*(1 + 2*c0*c0 - beta*beta)/D;
S->a3 = -4*c0*(1 - si*beta)/D;
S->a4 = (beta*beta - 2*si*beta + 1)/D;
}
}
static void butterworth_bp_filter(EqualizatorFilter *f,
int N, double w0, double wb,
double G, double Gb, double G0)
{
double g, c0, g0, beta;
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));
g = pow(G, 1.0 / N);
g0 = pow(G0, 1.0 / N);
beta = pow(epsilon, -1.0 / N) * tan(wb/2);
c0 = cos(w0);
for (i = 1; i <= L; i++) {
double ui = (2.0 * i - 1) / N;
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,
};

View File

@ -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);

View File

@ -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, \