avfilter: add high-order parametric multiband equalizer filter

Signed-off-by: Paul B Mahol <onemda@gmail.com>

1 files changed, 751 insertions, 0 deletions

diff --git a/libavfilter/af_anequalizer.c b/libavfilter/af_anequalizer.c
new file mode 100644
index 0000000000..a437e2b9dc
--- /dev/null
+++ b/libavfilter/af_anequalizer.c
@@ -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,
+};