@@ -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:

@@ -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"

@@ -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.

@@ -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

new file mode 100644

@@ -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,