libavfilter/af_anequalizer.c
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 /*
  * 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 "libavutil/intreadwrite.h"
 #include "libavutil/avstring.h"
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 #include "libavutil/ffmath.h"
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 #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++) {
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             double zr, zi, zr2, zi2;
             double Hr, Hi;
             double Hmag = 1;
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             double w;
             int v, y, x;
 
             w = M_PI * (s->fscale ? pow(s->w - 1, f / s->w) : f) / (s->w - 1);
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             zr = cos(w);
             zr2 = zr * zr;
             zi = -sin(w);
             zi2 = zi * zi;
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             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];
 
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                     /* 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)); */
 
                     Hr = S->b4*(1-8*zr2*zi2) + S->b2*(zr2-zi2) + zr*(S->b1+S->b3*(zr2-3*zi2))+ S->b0;
                     Hi = zi*(S->b3*(3*zr2-zi2) + S->b1 + 2*zr*(2*S->b4*(zr2-zi2) + S->b2));
                     Hmag *= hypot(Hr, Hi);
                     Hr = S->a4*(1-8*zr2*zi2) + S->a2*(zr2-zi2) + zr*(S->a1+S->a3*(zr2-3*zi2))+ S->a0;
                     Hi = zi*(S->a3*(3*zr2-zi2) + S->a1 + 2*zr*(2*S->a4*(zr2-zi2) + S->a2));
                     Hmag /= hypot(Hr, Hi);
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                 }
             }
 
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             v = av_clip((1. + -20 * log10(Hmag) / s->mag) * s->h / 2, 0, s->h - 1);
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             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;
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     int ret;
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     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,
         };
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         if (!vpad.name) {
             av_freep(&pad.name);
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             return AVERROR(ENOMEM);
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         }
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     }
 
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     ret = ff_insert_outpad(ctx, 0, &pad);
     if (ret < 0) {
         av_freep(&pad.name);
         return ret;
     }
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     if (s->draw_curves) {
         ret = ff_insert_outpad(ctx, 1, &vpad);
         if (ret < 0) {
             av_freep(&vpad.name);
             return ret;
         }
     }
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     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;
 
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     for (int i = 0; i < ctx->nb_outputs; i++)
         av_freep(&ctx->output_pads[i].name);
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     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;
     }
 
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     G  = ff_exp10(G/20);
     Gb = ff_exp10(Gb/20);
     G0 = ff_exp10(G0/20);
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     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;
     }
 
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     G  = ff_exp10(G/20);
     Gb = ff_exp10(Gb/20);
     G0 = ff_exp10(G0/20);
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     epsilon = sqrt((G*G - Gb*Gb) / (Gb*Gb - G0*G0));
     g0 = pow(G0,1.0/N);
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     alfa = pow(1.0/epsilon    + sqrt(1 + 1/(epsilon*epsilon)), 1.0/N);
     beta = pow(G/epsilon + Gb * sqrt(1 + 1/(epsilon*epsilon)), 1.0/N);
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     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;
     }
 
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     G  = ff_exp10(G/20);
     Gb = ff_exp10(Gb/20);
     G0 = ff_exp10(G0/20);
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     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);
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     if (s->nb_filters >= s->nb_allocated - 1) {
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         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;
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         av_free(args);
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         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 ||
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             s->filters[s->nb_filters].freq > inlink->sample_rate / 2.0)
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             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;
 
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         s->filters[s->nb_filters].type = av_clip(s->filters[s->nb_filters].type, 0, NB_TYPES - 1);
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         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);
 
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         if (freq < 0 || freq > inlink->sample_rate / 2.0)
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             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) {
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         AVFrame *clone;
 
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         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;
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         clone = av_frame_clone(s->video);
         if (!clone)
             return AVERROR(ENOMEM);
         ret = ff_filter_frame(ctx->outputs[1], clone);
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         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,
 };