libavfilter/af_superequalizer.c - third_party/ffmpeg - Git at Google

 /*
  * Copyright (c) 2002 Naoki Shibata
  * Copyright (c) 2017 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/opt.h"

 #include "libavcodec/avfft.h"

 #include "audio.h"
 #include "avfilter.h"
 #include "filters.h"
 #include "internal.h"

 #define NBANDS 17
 #define M 15

 typedef struct EqParameter {
     float lower, upper, gain;
 } EqParameter;

 typedef struct SuperEqualizerContext {
     const AVClass *class;

     EqParameter params[NBANDS + 1];

     float gains[NBANDS + 1];

     float fact[M + 1];
     float aa;
     float iza;
     float *ires, *irest;
     float *fsamples;
     int winlen, tabsize;

     AVFrame *in, *out;
     RDFTContext *rdft, *irdft;
 } SuperEqualizerContext;

 static const float bands[] = {
     65.406392, 92.498606, 130.81278, 184.99721, 261.62557, 369.99442, 523.25113, 739.9884, 1046.5023,
     1479.9768, 2093.0045, 2959.9536, 4186.0091, 5919.9072, 8372.0181, 11839.814, 16744.036
 };

 static float izero(SuperEqualizerContext *s, float x)
 {
     float ret = 1;
     int m;

     for (m = 1; m <= M; m++) {
         float t;

         t = pow(x / 2, m) / s->fact[m];
         ret += t*t;
     }

     return ret;
 }

 static float hn_lpf(int n, float f, float fs)
 {
     float t = 1 / fs;
     float omega = 2 * M_PI * f;

     if (n * omega * t == 0)
         return 2 * f * t;
     return 2 * f * t * sinf(n * omega * t) / (n * omega * t);
 }

 static float hn_imp(int n)
 {
     return n == 0 ? 1.f : 0.f;
 }

 static float hn(int n, EqParameter *param, float fs)
 {
     float ret, lhn;
     int i;

     lhn = hn_lpf(n, param[0].upper, fs);
     ret = param[0].gain*lhn;

     for (i = 1; i < NBANDS + 1 && param[i].upper < fs / 2; i++) {
         float lhn2 = hn_lpf(n, param[i].upper, fs);
         ret += param[i].gain * (lhn2 - lhn);
         lhn = lhn2;
     }

     ret += param[i].gain * (hn_imp(n) - lhn);

     return ret;
 }

 static float alpha(float a)
 {
     if (a <= 21)
         return 0;
     if (a <= 50)
         return .5842f * pow(a - 21, 0.4f) + 0.07886f * (a - 21);
     return .1102f * (a - 8.7f);
 }

 static float win(SuperEqualizerContext *s, float n, int N)
 {
     return izero(s, alpha(s->aa) * sqrtf(1 - 4 * n * n / ((N - 1) * (N - 1)))) / s->iza;
 }

 static void process_param(float *bc, EqParameter *param, float fs)
 {
     int i;

     for (i = 0; i <= NBANDS; i++) {
         param[i].lower = i == 0 ? 0 : bands[i - 1];
         param[i].upper = i == NBANDS ? fs : bands[i];
         param[i].gain  = bc[i];
     }
 }

 static int equ_init(SuperEqualizerContext *s, int wb)
 {
     int i,j;

     s->rdft  = av_rdft_init(wb, DFT_R2C);
     s->irdft = av_rdft_init(wb, IDFT_C2R);
     if (!s->rdft || !s->irdft)
         return AVERROR(ENOMEM);

     s->aa = 96;
     s->winlen = (1 << (wb-1))-1;
     s->tabsize  = 1 << wb;

     s->ires     = av_calloc(s->tabsize, sizeof(float));
     s->irest    = av_calloc(s->tabsize, sizeof(float));
     s->fsamples = av_calloc(s->tabsize, sizeof(float));

     for (i = 0; i <= M; i++) {
         s->fact[i] = 1;
         for (j = 1; j <= i; j++)
             s->fact[i] *= j;
     }

     s->iza = izero(s, alpha(s->aa));

     return 0;
 }

 static void make_fir(SuperEqualizerContext *s, float *lbc, float *rbc, EqParameter *param, float fs)
 {
     const int winlen = s->winlen;
     const int tabsize = s->tabsize;
     float *nires;
     int i;

     if (fs <= 0)
         return;

     process_param(lbc, param, fs);
     for (i = 0; i < winlen; i++)
         s->irest[i] = hn(i - winlen / 2, param, fs) * win(s, i - winlen / 2, winlen);
     for (; i < tabsize; i++)
         s->irest[i] = 0;

     av_rdft_calc(s->rdft, s->irest);
     nires = s->ires;
     for (i = 0; i < tabsize; i++)
         nires[i] = s->irest[i];
 }

 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 {
     AVFilterContext *ctx = inlink->dst;
     SuperEqualizerContext *s = ctx->priv;
     AVFilterLink *outlink = ctx->outputs[0];
     const float *ires = s->ires;
     float *fsamples = s->fsamples;
     int ch, i;

     AVFrame *out = ff_get_audio_buffer(outlink, s->winlen);
     float *src, *dst, *ptr;

     if (!out) {
         av_frame_free(&in);
         return AVERROR(ENOMEM);
     }

     for (ch = 0; ch < in->channels; ch++) {
         ptr = (float *)out->extended_data[ch];
         dst = (float *)s->out->extended_data[ch];
         src = (float *)in->extended_data[ch];

         for (i = 0; i < in->nb_samples; i++)
             fsamples[i] = src[i];
         for (; i < s->tabsize; i++)
             fsamples[i] = 0;

         av_rdft_calc(s->rdft, fsamples);

         fsamples[0] = ires[0] * fsamples[0];
         fsamples[1] = ires[1] * fsamples[1];
         for (i = 1; i < s->tabsize / 2; i++) {
             float re, im;

             re = ires[i*2  ] * fsamples[i*2] - ires[i*2+1] * fsamples[i*2+1];
             im = ires[i*2+1] * fsamples[i*2] + ires[i*2  ] * fsamples[i*2+1];

             fsamples[i*2  ] = re;
             fsamples[i*2+1] = im;
         }

         av_rdft_calc(s->irdft, fsamples);

         for (i = 0; i < s->winlen; i++)
             dst[i] += fsamples[i] / s->tabsize * 2;
         for (i = s->winlen; i < s->tabsize; i++)
             dst[i]  = fsamples[i] / s->tabsize * 2;
         for (i = 0; i < s->winlen; i++)
             ptr[i] = dst[i];
         for (i = 0; i < s->winlen; i++)
             dst[i] = dst[i+s->winlen];
     }

     out->pts = in->pts;
     av_frame_free(&in);

     return ff_filter_frame(outlink, out);
 }

 static int activate(AVFilterContext *ctx)
 {
     AVFilterLink *inlink = ctx->inputs[0];
     AVFilterLink *outlink = ctx->outputs[0];
     SuperEqualizerContext *s = ctx->priv;
     AVFrame *in = NULL;
     int ret;

     FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);

     ret = ff_inlink_consume_samples(inlink, s->winlen, s->winlen, &in);
     if (ret < 0)
         return ret;
     if (ret > 0)
         return filter_frame(inlink, in);

     FF_FILTER_FORWARD_STATUS(inlink, outlink);
     FF_FILTER_FORWARD_WANTED(outlink, inlink);

     return FFERROR_NOT_READY;
 }

 static av_cold int init(AVFilterContext *ctx)
 {
     SuperEqualizerContext *s = ctx->priv;

     return equ_init(s, 14);
 }

 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterFormats *formats;
     AVFilterChannelLayouts *layouts;
     static const enum AVSampleFormat sample_fmts[] = {
         AV_SAMPLE_FMT_FLTP,
         AV_SAMPLE_FMT_NONE
     };
     int ret;

     layouts = ff_all_channel_counts();
     if (!layouts)
         return AVERROR(ENOMEM);
     ret = ff_set_common_channel_layouts(ctx, layouts);
     if (ret < 0)
         return ret;

     formats = ff_make_format_list(sample_fmts);
     if ((ret = ff_set_common_formats(ctx, formats)) < 0)
         return ret;

     formats = ff_all_samplerates();
     return ff_set_common_samplerates(ctx, formats);
 }

 static int config_input(AVFilterLink *inlink)
 {
     AVFilterContext *ctx = inlink->dst;
     SuperEqualizerContext *s = ctx->priv;

     s->out = ff_get_audio_buffer(inlink, s->tabsize);
     if (!s->out)
         return AVERROR(ENOMEM);

     return 0;
 }

 static int config_output(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     SuperEqualizerContext *s = ctx->priv;

     make_fir(s, s->gains, s->gains, s->params, outlink->sample_rate);

     return 0;
 }

 static av_cold void uninit(AVFilterContext *ctx)
 {
     SuperEqualizerContext *s = ctx->priv;

     av_frame_free(&s->out);
     av_freep(&s->irest);
     av_freep(&s->ires);
     av_freep(&s->fsamples);
     av_rdft_end(s->rdft);
     av_rdft_end(s->irdft);
 }

 static const AVFilterPad superequalizer_inputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .config_props = config_input,
     },
     { NULL }
 };

 static const AVFilterPad superequalizer_outputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .config_props = config_output,
     },
     { NULL }
 };

 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
 #define OFFSET(x) offsetof(SuperEqualizerContext, x)

 static const AVOption superequalizer_options[] = {
     {  "1b", "set 65Hz band gain",    OFFSET(gains [0]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
     {  "2b", "set 92Hz band gain",    OFFSET(gains [1]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
     {  "3b", "set 131Hz band gain",   OFFSET(gains [2]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
     {  "4b", "set 185Hz band gain",   OFFSET(gains [3]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
     {  "5b", "set 262Hz band gain",   OFFSET(gains [4]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
     {  "6b", "set 370Hz band gain",   OFFSET(gains [5]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
     {  "7b", "set 523Hz band gain",   OFFSET(gains [6]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
     {  "8b", "set 740Hz band gain",   OFFSET(gains [7]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
     {  "9b", "set 1047Hz band gain",  OFFSET(gains [8]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
     { "10b", "set 1480Hz band gain",  OFFSET(gains [9]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
     { "11b", "set 2093Hz band gain",  OFFSET(gains[10]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
     { "12b", "set 2960Hz band gain",  OFFSET(gains[11]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
     { "13b", "set 4186Hz band gain",  OFFSET(gains[12]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
     { "14b", "set 5920Hz band gain",  OFFSET(gains[13]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
     { "15b", "set 8372Hz band gain",  OFFSET(gains[14]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
     { "16b", "set 11840Hz band gain", OFFSET(gains[15]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
     { "17b", "set 16744Hz band gain", OFFSET(gains[16]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
     { "18b", "set 20000Hz band gain", OFFSET(gains[17]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
     { NULL }
 };

 AVFILTER_DEFINE_CLASS(superequalizer);

 AVFilter ff_af_superequalizer = {
     .name          = "superequalizer",
     .description   = NULL_IF_CONFIG_SMALL("Apply 18 band equalization filter."),
     .priv_size     = sizeof(SuperEqualizerContext),
     .priv_class    = &superequalizer_class,
     .query_formats = query_formats,
     .init          = init,
     .activate      = activate,
     .uninit        = uninit,
     .inputs        = superequalizer_inputs,
     .outputs       = superequalizer_outputs,
 };
	/*
	* Copyright (c) 2002 Naoki Shibata
	* Copyright (c) 2017 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/opt.h"

	#include "libavcodec/avfft.h"

	#include "audio.h"
	#include "avfilter.h"
	#include "filters.h"
	#include "internal.h"

	#define NBANDS 17
	#define M 15

	typedef struct EqParameter {
	float lower, upper, gain;
	} EqParameter;

	typedef struct SuperEqualizerContext {
	const AVClass *class;

	EqParameter params[NBANDS + 1];

	float gains[NBANDS + 1];

	float fact[M + 1];
	float aa;
	float iza;
	float ires, irest;
	float *fsamples;
	int winlen, tabsize;

	AVFrame in, out;
	RDFTContext rdft, irdft;
	} SuperEqualizerContext;

	static const float bands[] = {
	65.406392, 92.498606, 130.81278, 184.99721, 261.62557, 369.99442, 523.25113, 739.9884, 1046.5023,
	1479.9768, 2093.0045, 2959.9536, 4186.0091, 5919.9072, 8372.0181, 11839.814, 16744.036
	};

	static float izero(SuperEqualizerContext *s, float x)
	{
	float ret = 1;
	int m;

	for (m = 1; m <= M; m++) {
	float t;

	t = pow(x / 2, m) / s->fact[m];
	ret += t*t;
	}

	return ret;
	}

	static float hn_lpf(int n, float f, float fs)
	{
	float t = 1 / fs;
	float omega = 2 * M_PI * f;

	if (n * omega * t == 0)
	return 2 * f * t;
	return 2 * f * t * sinf(n * omega * t) / (n * omega * t);
	}

	static float hn_imp(int n)
	{
	return n == 0 ? 1.f : 0.f;
	}

	static float hn(int n, EqParameter *param, float fs)
	{
	float ret, lhn;
	int i;

	lhn = hn_lpf(n, param[0].upper, fs);
	ret = param[0].gain*lhn;

	for (i = 1; i < NBANDS + 1 && param[i].upper < fs / 2; i++) {
	float lhn2 = hn_lpf(n, param[i].upper, fs);
	ret += param[i].gain * (lhn2 - lhn);
	lhn = lhn2;
	}

	ret += param[i].gain * (hn_imp(n) - lhn);

	return ret;
	}

	static float alpha(float a)
	{
	if (a <= 21)
	return 0;
	if (a <= 50)
	return .5842f * pow(a - 21, 0.4f) + 0.07886f * (a - 21);
	return .1102f * (a - 8.7f);
	}

	static float win(SuperEqualizerContext *s, float n, int N)
	{
	return izero(s, alpha(s->aa) * sqrtf(1 - 4 * n * n / ((N - 1) * (N - 1)))) / s->iza;
	}

	static void process_param(float bc, EqParameter param, float fs)
	{
	int i;

	for (i = 0; i <= NBANDS; i++) {
	param[i].lower = i == 0 ? 0 : bands[i - 1];
	param[i].upper = i == NBANDS ? fs : bands[i];
	param[i].gain = bc[i];
	}
	}

	static int equ_init(SuperEqualizerContext *s, int wb)
	{
	int i,j;

	s->rdft = av_rdft_init(wb, DFT_R2C);
	s->irdft = av_rdft_init(wb, IDFT_C2R);
	if (!s->rdft \|\| !s->irdft)
	return AVERROR(ENOMEM);

	s->aa = 96;
	s->winlen = (1 << (wb-1))-1;
	s->tabsize = 1 << wb;

	s->ires = av_calloc(s->tabsize, sizeof(float));
	s->irest = av_calloc(s->tabsize, sizeof(float));
	s->fsamples = av_calloc(s->tabsize, sizeof(float));

	for (i = 0; i <= M; i++) {
	s->fact[i] = 1;
	for (j = 1; j <= i; j++)
	s->fact[i] *= j;
	}

	s->iza = izero(s, alpha(s->aa));

	return 0;
	}

	static void make_fir(SuperEqualizerContext s, float lbc, float rbc, EqParameter param, float fs)
	{
	const int winlen = s->winlen;
	const int tabsize = s->tabsize;
	float *nires;
	int i;

	if (fs <= 0)
	return;

	process_param(lbc, param, fs);
	for (i = 0; i < winlen; i++)
	s->irest[i] = hn(i - winlen / 2, param, fs) * win(s, i - winlen / 2, winlen);
	for (; i < tabsize; i++)
	s->irest[i] = 0;

	av_rdft_calc(s->rdft, s->irest);
	nires = s->ires;
	for (i = 0; i < tabsize; i++)
	nires[i] = s->irest[i];
	}

	static int filter_frame(AVFilterLink inlink, AVFrame in)
	{
	AVFilterContext *ctx = inlink->dst;
	SuperEqualizerContext *s = ctx->priv;
	AVFilterLink *outlink = ctx->outputs[0];
	const float *ires = s->ires;
	float *fsamples = s->fsamples;
	int ch, i;

	AVFrame *out = ff_get_audio_buffer(outlink, s->winlen);
	float src, dst, *ptr;

	if (!out) {
	av_frame_free(&in);
	return AVERROR(ENOMEM);
	}

	for (ch = 0; ch < in->channels; ch++) {
	ptr = (float *)out->extended_data[ch];
	dst = (float *)s->out->extended_data[ch];
	src = (float *)in->extended_data[ch];

	for (i = 0; i < in->nb_samples; i++)
	fsamples[i] = src[i];
	for (; i < s->tabsize; i++)
	fsamples[i] = 0;

	av_rdft_calc(s->rdft, fsamples);

	fsamples[0] = ires[0] * fsamples[0];
	fsamples[1] = ires[1] * fsamples[1];
	for (i = 1; i < s->tabsize / 2; i++) {
	float re, im;

	re = ires[i2 ] fsamples[i2] - ires[i2+1] * fsamples[i*2+1];
	im = ires[i2+1] fsamples[i2] + ires[i2 ] * fsamples[i*2+1];

	fsamples[i*2 ] = re;
	fsamples[i*2+1] = im;
	}

	av_rdft_calc(s->irdft, fsamples);

	for (i = 0; i < s->winlen; i++)
	dst[i] += fsamples[i] / s->tabsize * 2;
	for (i = s->winlen; i < s->tabsize; i++)
	dst[i] = fsamples[i] / s->tabsize * 2;
	for (i = 0; i < s->winlen; i++)
	ptr[i] = dst[i];
	for (i = 0; i < s->winlen; i++)
	dst[i] = dst[i+s->winlen];
	}

	out->pts = in->pts;
	av_frame_free(&in);

	return ff_filter_frame(outlink, out);
	}

	static int activate(AVFilterContext *ctx)
	{
	AVFilterLink *inlink = ctx->inputs[0];
	AVFilterLink *outlink = ctx->outputs[0];
	SuperEqualizerContext *s = ctx->priv;
	AVFrame *in = NULL;
	int ret;

	FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);

	ret = ff_inlink_consume_samples(inlink, s->winlen, s->winlen, &in);
	if (ret < 0)
	return ret;
	if (ret > 0)
	return filter_frame(inlink, in);

	FF_FILTER_FORWARD_STATUS(inlink, outlink);
	FF_FILTER_FORWARD_WANTED(outlink, inlink);

	return FFERROR_NOT_READY;
	}

	static av_cold int init(AVFilterContext *ctx)
	{
	SuperEqualizerContext *s = ctx->priv;

	return equ_init(s, 14);
	}

	static int query_formats(AVFilterContext *ctx)
	{
	AVFilterFormats *formats;
	AVFilterChannelLayouts *layouts;
	static const enum AVSampleFormat sample_fmts[] = {
	AV_SAMPLE_FMT_FLTP,
	AV_SAMPLE_FMT_NONE
	};
	int ret;

	layouts = ff_all_channel_counts();
	if (!layouts)
	return AVERROR(ENOMEM);
	ret = ff_set_common_channel_layouts(ctx, layouts);
	if (ret < 0)
	return ret;

	formats = ff_make_format_list(sample_fmts);
	if ((ret = ff_set_common_formats(ctx, formats)) < 0)
	return ret;

	formats = ff_all_samplerates();
	return ff_set_common_samplerates(ctx, formats);
	}

	static int config_input(AVFilterLink *inlink)
	{
	AVFilterContext *ctx = inlink->dst;
	SuperEqualizerContext *s = ctx->priv;

	s->out = ff_get_audio_buffer(inlink, s->tabsize);
	if (!s->out)
	return AVERROR(ENOMEM);

	return 0;
	}

	static int config_output(AVFilterLink *outlink)
	{
	AVFilterContext *ctx = outlink->src;
	SuperEqualizerContext *s = ctx->priv;

	make_fir(s, s->gains, s->gains, s->params, outlink->sample_rate);

	return 0;
	}

	static av_cold void uninit(AVFilterContext *ctx)
	{
	SuperEqualizerContext *s = ctx->priv;

	av_frame_free(&s->out);
	av_freep(&s->irest);
	av_freep(&s->ires);
	av_freep(&s->fsamples);
	av_rdft_end(s->rdft);
	av_rdft_end(s->irdft);
	}

	static const AVFilterPad superequalizer_inputs[] = {
	{
	.name = "default",
	.type = AVMEDIA_TYPE_AUDIO,
	.config_props = config_input,
	},
	{ NULL }
	};

	static const AVFilterPad superequalizer_outputs[] = {
	{
	.name = "default",
	.type = AVMEDIA_TYPE_AUDIO,
	.config_props = config_output,
	},
	{ NULL }
	};

	#define AF AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM
	#define OFFSET(x) offsetof(SuperEqualizerContext, x)

	static const AVOption superequalizer_options[] = {
	{ "1b", "set 65Hz band gain", OFFSET(gains [0]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
	{ "2b", "set 92Hz band gain", OFFSET(gains [1]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
	{ "3b", "set 131Hz band gain", OFFSET(gains [2]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
	{ "4b", "set 185Hz band gain", OFFSET(gains [3]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
	{ "5b", "set 262Hz band gain", OFFSET(gains [4]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
	{ "6b", "set 370Hz band gain", OFFSET(gains [5]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
	{ "7b", "set 523Hz band gain", OFFSET(gains [6]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
	{ "8b", "set 740Hz band gain", OFFSET(gains [7]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
	{ "9b", "set 1047Hz band gain", OFFSET(gains [8]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
	{ "10b", "set 1480Hz band gain", OFFSET(gains [9]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
	{ "11b", "set 2093Hz band gain", OFFSET(gains[10]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
	{ "12b", "set 2960Hz band gain", OFFSET(gains[11]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
	{ "13b", "set 4186Hz band gain", OFFSET(gains[12]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
	{ "14b", "set 5920Hz band gain", OFFSET(gains[13]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
	{ "15b", "set 8372Hz band gain", OFFSET(gains[14]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
	{ "16b", "set 11840Hz band gain", OFFSET(gains[15]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
	{ "17b", "set 16744Hz band gain", OFFSET(gains[16]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
	{ "18b", "set 20000Hz band gain", OFFSET(gains[17]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
	{ NULL }
	};

	AVFILTER_DEFINE_CLASS(superequalizer);

	AVFilter ff_af_superequalizer = {
	.name = "superequalizer",
	.description = NULL_IF_CONFIG_SMALL("Apply 18 band equalization filter."),
	.priv_size = sizeof(SuperEqualizerContext),
	.priv_class = &superequalizer_class,
	.query_formats = query_formats,
	.init = init,
	.activate = activate,
	.uninit = uninit,
	.inputs = superequalizer_inputs,
	.outputs = superequalizer_outputs,
	};