libavfilter/af_aemphasis.c - nest-android-app/ffmpeg - Git at Google

 /*
  * Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen, Damien Zammit and others
  *
  * 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 "avfilter.h"
 #include "internal.h"
 #include "audio.h"

 typedef struct BiquadCoeffs {
     double a0, a1, a2, b1, b2;
 } BiquadCoeffs;

 typedef struct BiquadD2 {
     double a0, a1, a2, b1, b2, w1, w2;
 } BiquadD2;

 typedef struct RIAACurve {
     BiquadD2 r1;
     BiquadD2 brickw;
     int use_brickw;
 } RIAACurve;

 typedef struct AudioEmphasisContext {
     const AVClass *class;
     int mode, type;
     double level_in, level_out;

     RIAACurve *rc;
 } AudioEmphasisContext;

 #define OFFSET(x) offsetof(AudioEmphasisContext, x)
 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

 static const AVOption aemphasis_options[] = {
     { "level_in",      "set input gain", OFFSET(level_in),  AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, FLAGS },
     { "level_out",    "set output gain", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, FLAGS },
     { "mode",         "set filter mode", OFFSET(mode), AV_OPT_TYPE_INT,   {.i64=0}, 0, 1, FLAGS, "mode" },
     { "reproduction",              NULL,            0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "mode" },
     { "production",                NULL,            0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "mode" },
     { "type",         "set filter type", OFFSET(type), AV_OPT_TYPE_INT,   {.i64=4}, 0, 8, FLAGS, "type" },
     { "col",                 "Columbia",            0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "type" },
     { "emi",                      "EMI",            0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "type" },
     { "bsi",              "BSI (78RPM)",            0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, "type" },
     { "riaa",                    "RIAA",            0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, FLAGS, "type" },
     { "cd",         "Compact Disc (CD)",            0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, FLAGS, "type" },
     { "50fm",               "50µs (FM)",            0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, FLAGS, "type" },
     { "75fm",               "75µs (FM)",            0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, FLAGS, "type" },
     { "50kf",            "50µs (FM-KF)",            0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, FLAGS, "type" },
     { "75kf",            "75µs (FM-KF)",            0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, FLAGS, "type" },
     { NULL }
 };

 AVFILTER_DEFINE_CLASS(aemphasis);

 static inline double biquad(BiquadD2 *bq, double in)
 {
     double n = in;
     double tmp = n - bq->w1 * bq->b1 - bq->w2 * bq->b2;
     double out = tmp * bq->a0 + bq->w1 * bq->a1 + bq->w2 * bq->a2;

     bq->w2 = bq->w1;
     bq->w1 = tmp;

     return out;
 }

 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 {
     AVFilterContext *ctx = inlink->dst;
     AVFilterLink *outlink = ctx->outputs[0];
     AudioEmphasisContext *s = ctx->priv;
     const double *src = (const double *)in->data[0];
     const double level_out = s->level_out;
     const double level_in = s->level_in;
     AVFrame *out;
     double *dst;
     int n, c;

     if (av_frame_is_writable(in)) {
         out = in;
     } else {
         out = ff_get_audio_buffer(inlink, in->nb_samples);
         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);
         }
         av_frame_copy_props(out, in);
     }
     dst = (double *)out->data[0];

     for (n = 0; n < in->nb_samples; n++) {
         for (c = 0; c < inlink->channels; c++)
             dst[c] = level_out * biquad(&s->rc[c].r1, s->rc[c].use_brickw ? biquad(&s->rc[c].brickw, src[c] * level_in) : src[c] * level_in);
         dst += inlink->channels;
         src += inlink->channels;
     }

     if (in != out)
         av_frame_free(&in);
     return ff_filter_frame(outlink, out);
 }

 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterChannelLayouts *layouts;
     AVFilterFormats *formats;
     static const enum AVSampleFormat sample_fmts[] = {
         AV_SAMPLE_FMT_DBL,
         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 (!formats)
         return AVERROR(ENOMEM);
     ret = ff_set_common_formats(ctx, formats);
     if (ret < 0)
         return ret;

     formats = ff_all_samplerates();
     if (!formats)
         return AVERROR(ENOMEM);
     return ff_set_common_samplerates(ctx, formats);
 }

 static inline void set_highshelf_rbj(BiquadD2 *bq, double freq, double q, double peak, double sr)
 {
     double A = sqrt(peak);
     double w0 = freq * 2 * M_PI / sr;
     double alpha = sin(w0) / (2 * q);
     double cw0 = cos(w0);
     double tmp = 2 * sqrt(A) * alpha;
     double b0 = 0, ib0 = 0;

     bq->a0 =    A*( (A+1) + (A-1)*cw0 + tmp);
     bq->a1 = -2*A*( (A-1) + (A+1)*cw0);
     bq->a2 =    A*( (A+1) + (A-1)*cw0 - tmp);
         b0 =        (A+1) - (A-1)*cw0 + tmp;
     bq->b1 =    2*( (A-1) - (A+1)*cw0);
     bq->b2 =        (A+1) - (A-1)*cw0 - tmp;

     ib0     = 1 / b0;
     bq->b1 *= ib0;
     bq->b2 *= ib0;
     bq->a0 *= ib0;
     bq->a1 *= ib0;
     bq->a2 *= ib0;
 }

 static inline void set_lp_rbj(BiquadD2 *bq, double fc, double q, double sr, double gain)
 {
     double omega = 2.0 * M_PI * fc / sr;
     double sn = sin(omega);
     double cs = cos(omega);
     double alpha = sn/(2 * q);
     double inv = 1.0/(1.0 + alpha);

     bq->a2 = bq->a0 = gain * inv * (1.0 - cs) * 0.5;
     bq->a1 = bq->a0 + bq->a0;
     bq->b1 = (-2.0 * cs * inv);
     bq->b2 = ((1.0 - alpha) * inv);
 }

 static double freq_gain(BiquadCoeffs *c, double freq, double sr)
 {
     double zr, zi;

     freq *= 2.0 * M_PI / sr;
     zr = cos(freq);
     zi = -sin(freq);

     /* |(a0 + a1*z + a2*z^2)/(1 + b1*z + b2*z^2)| */
     return hypot(c->a0 + c->a1*zr + c->a2*(zr*zr-zi*zi), c->a1*zi + 2*c->a2*zr*zi) /
            hypot(1 + c->b1*zr + c->b2*(zr*zr-zi*zi), c->b1*zi + 2*c->b2*zr*zi);
 }

 static int config_input(AVFilterLink *inlink)
 {
     double i, j, k, g, t, a0, a1, a2, b1, b2, tau1, tau2, tau3;
     double cutfreq, gain1kHz, gc, sr = inlink->sample_rate;
     AVFilterContext *ctx = inlink->dst;
     AudioEmphasisContext *s = ctx->priv;
     BiquadCoeffs coeffs;
     int ch;

     s->rc = av_calloc(inlink->channels, sizeof(*s->rc));
     if (!s->rc)
         return AVERROR(ENOMEM);

     switch (s->type) {
     case 0: //"Columbia"
         i = 100.;
         j = 500.;
         k = 1590.;
         break;
     case 1: //"EMI"
         i = 70.;
         j = 500.;
         k = 2500.;
         break;
     case 2: //"BSI(78rpm)"
         i = 50.;
         j = 353.;
         k = 3180.;
         break;
     case 3: //"RIAA"
     default:
         tau1 = 0.003180;
         tau2 = 0.000318;
         tau3 = 0.000075;
         i = 1. / (2. * M_PI * tau1);
         j = 1. / (2. * M_PI * tau2);
         k = 1. / (2. * M_PI * tau3);
         break;
     case 4: //"CD Mastering"
         tau1 = 0.000050;
         tau2 = 0.000015;
         tau3 = 0.0000001;// 1.6MHz out of audible range for null impact
         i = 1. / (2. * M_PI * tau1);
         j = 1. / (2. * M_PI * tau2);
         k = 1. / (2. * M_PI * tau3);
         break;
     case 5: //"50µs FM (Europe)"
         tau1 = 0.000050;
         tau2 = tau1 / 20;// not used
         tau3 = tau1 / 50;//
         i = 1. / (2. * M_PI * tau1);
         j = 1. / (2. * M_PI * tau2);
         k = 1. / (2. * M_PI * tau3);
         break;
     case 6: //"75µs FM (US)"
         tau1 = 0.000075;
         tau2 = tau1 / 20;// not used
         tau3 = tau1 / 50;//
         i = 1. / (2. * M_PI * tau1);
         j = 1. / (2. * M_PI * tau2);
         k = 1. / (2. * M_PI * tau3);
         break;
     }

     i *= 2 * M_PI;
     j *= 2 * M_PI;
     k *= 2 * M_PI;

     t = 1. / sr;

     //swap a1 b1, a2 b2
     if (s->type == 7 || s->type == 8) {
         double tau = (s->type == 7 ? 0.000050 : 0.000075);
         double f = 1.0 / (2 * M_PI * tau);
         double nyq = sr * 0.5;
         double gain = sqrt(1.0 + nyq * nyq / (f * f)); // gain at Nyquist
         double cfreq = sqrt((gain - 1.0) * f * f); // frequency
         double q = 1.0;

         if (s->type == 8)
             q = pow((sr / 3269.0) + 19.5, -0.25); // somewhat poor curve-fit
         if (s->type == 7)
             q = pow((sr / 4750.0) + 19.5, -0.25);
         if (s->mode == 0)
             set_highshelf_rbj(&s->rc[0].r1, cfreq, q, 1. / gain, sr);
         else
             set_highshelf_rbj(&s->rc[0].r1, cfreq, q, gain, sr);
         s->rc[0].use_brickw = 0;
     } else {
         s->rc[0].use_brickw = 1;
         if (s->mode == 0) { // Reproduction
             g  = 1. / (4.+2.*i*t+2.*k*t+i*k*t*t);
             a0 = (2.*t+j*t*t)*g;
             a1 = (2.*j*t*t)*g;
             a2 = (-2.*t+j*t*t)*g;
             b1 = (-8.+2.*i*k*t*t)*g;
             b2 = (4.-2.*i*t-2.*k*t+i*k*t*t)*g;
         } else {  // Production
             g  = 1. / (2.*t+j*t*t);
             a0 = (4.+2.*i*t+2.*k*t+i*k*t*t)*g;
             a1 = (-8.+2.*i*k*t*t)*g;
             a2 = (4.-2.*i*t-2.*k*t+i*k*t*t)*g;
             b1 = (2.*j*t*t)*g;
             b2 = (-2.*t+j*t*t)*g;
         }

         coeffs.a0 = a0;
         coeffs.a1 = a1;
         coeffs.a2 = a2;
         coeffs.b1 = b1;
         coeffs.b2 = b2;

         // the coeffs above give non-normalized value, so it should be normalized to produce 0dB at 1 kHz
         // find actual gain
         // Note: for FM emphasis, use 100 Hz for normalization instead
         gain1kHz = freq_gain(&coeffs, 1000.0, sr);
         // divide one filter's x[n-m] coefficients by that value
         gc = 1.0 / gain1kHz;
         s->rc[0].r1.a0 = coeffs.a0 * gc;
         s->rc[0].r1.a1 = coeffs.a1 * gc;
         s->rc[0].r1.a2 = coeffs.a2 * gc;
         s->rc[0].r1.b1 = coeffs.b1;
         s->rc[0].r1.b2 = coeffs.b2;
     }

     cutfreq = FFMIN(0.45 * sr, 21000.);
     set_lp_rbj(&s->rc[0].brickw, cutfreq, 0.707, sr, 1.);

     for (ch = 1; ch < inlink->channels; ch++) {
         memcpy(&s->rc[ch], &s->rc[0], sizeof(RIAACurve));
     }

     return 0;
 }

 static av_cold void uninit(AVFilterContext *ctx)
 {
     AudioEmphasisContext *s = ctx->priv;
     av_freep(&s->rc);
 }

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

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

 AVFilter ff_af_aemphasis = {
     .name          = "aemphasis",
     .description   = NULL_IF_CONFIG_SMALL("Audio emphasis."),
     .priv_size     = sizeof(AudioEmphasisContext),
     .priv_class    = &aemphasis_class,
     .uninit        = uninit,
     .query_formats = query_formats,
     .inputs        = avfilter_af_aemphasis_inputs,
     .outputs       = avfilter_af_aemphasis_outputs,
 };
	/*
	* Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen, Damien Zammit and others
	*
	* 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 "avfilter.h"
	#include "internal.h"
	#include "audio.h"

	typedef struct BiquadCoeffs {
	double a0, a1, a2, b1, b2;
	} BiquadCoeffs;

	typedef struct BiquadD2 {
	double a0, a1, a2, b1, b2, w1, w2;
	} BiquadD2;

	typedef struct RIAACurve {
	BiquadD2 r1;
	BiquadD2 brickw;
	int use_brickw;
	} RIAACurve;

	typedef struct AudioEmphasisContext {
	const AVClass *class;
	int mode, type;
	double level_in, level_out;

	RIAACurve *rc;
	} AudioEmphasisContext;

	#define OFFSET(x) offsetof(AudioEmphasisContext, x)
	#define FLAGS AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM

	static const AVOption aemphasis_options[] = {
	{ "level_in", "set input gain", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, FLAGS },
	{ "level_out", "set output gain", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, FLAGS },
	{ "mode", "set filter mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS, "mode" },
	{ "reproduction", NULL, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "mode" },
	{ "production", NULL, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "mode" },
	{ "type", "set filter type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=4}, 0, 8, FLAGS, "type" },
	{ "col", "Columbia", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "type" },
	{ "emi", "EMI", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "type" },
	{ "bsi", "BSI (78RPM)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, "type" },
	{ "riaa", "RIAA", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, FLAGS, "type" },
	{ "cd", "Compact Disc (CD)", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, FLAGS, "type" },
	{ "50fm", "50µs (FM)", 0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, FLAGS, "type" },
	{ "75fm", "75µs (FM)", 0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, FLAGS, "type" },
	{ "50kf", "50µs (FM-KF)", 0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, FLAGS, "type" },
	{ "75kf", "75µs (FM-KF)", 0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, FLAGS, "type" },
	{ NULL }
	};

	AVFILTER_DEFINE_CLASS(aemphasis);

	static inline double biquad(BiquadD2 *bq, double in)
	{
	double n = in;
	double tmp = n - bq->w1 * bq->b1 - bq->w2 * bq->b2;
	double out = tmp * bq->a0 + bq->w1 * bq->a1 + bq->w2 * bq->a2;

	bq->w2 = bq->w1;
	bq->w1 = tmp;

	return out;
	}

	static int filter_frame(AVFilterLink inlink, AVFrame in)
	{
	AVFilterContext *ctx = inlink->dst;
	AVFilterLink *outlink = ctx->outputs[0];
	AudioEmphasisContext *s = ctx->priv;
	const double src = (const double )in->data[0];
	const double level_out = s->level_out;
	const double level_in = s->level_in;
	AVFrame *out;
	double *dst;
	int n, c;

	if (av_frame_is_writable(in)) {
	out = in;
	} else {
	out = ff_get_audio_buffer(inlink, in->nb_samples);
	if (!out) {
	av_frame_free(&in);
	return AVERROR(ENOMEM);
	}
	av_frame_copy_props(out, in);
	}
	dst = (double *)out->data[0];

	for (n = 0; n < in->nb_samples; n++) {
	for (c = 0; c < inlink->channels; c++)
	dst[c] = level_out * biquad(&s->rc[c].r1, s->rc[c].use_brickw ? biquad(&s->rc[c].brickw, src[c] * level_in) : src[c] * level_in);
	dst += inlink->channels;
	src += inlink->channels;
	}

	if (in != out)
	av_frame_free(&in);
	return ff_filter_frame(outlink, out);
	}

	static int query_formats(AVFilterContext *ctx)
	{
	AVFilterChannelLayouts *layouts;
	AVFilterFormats *formats;
	static const enum AVSampleFormat sample_fmts[] = {
	AV_SAMPLE_FMT_DBL,
	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 (!formats)
	return AVERROR(ENOMEM);
	ret = ff_set_common_formats(ctx, formats);
	if (ret < 0)
	return ret;

	formats = ff_all_samplerates();
	if (!formats)
	return AVERROR(ENOMEM);
	return ff_set_common_samplerates(ctx, formats);
	}

	static inline void set_highshelf_rbj(BiquadD2 *bq, double freq, double q, double peak, double sr)
	{
	double A = sqrt(peak);
	double w0 = freq * 2 * M_PI / sr;
	double alpha = sin(w0) / (2 * q);
	double cw0 = cos(w0);
	double tmp = 2 * sqrt(A) * alpha;
	double b0 = 0, ib0 = 0;

	bq->a0 = A( (A+1) + (A-1)cw0 + tmp);
	bq->a1 = -2A( (A-1) + (A+1)*cw0);
	bq->a2 = A( (A+1) + (A-1)cw0 - tmp);
	b0 = (A+1) - (A-1)*cw0 + tmp;
	bq->b1 = 2( (A-1) - (A+1)cw0);
	bq->b2 = (A+1) - (A-1)*cw0 - tmp;

	ib0 = 1 / b0;
	bq->b1 *= ib0;
	bq->b2 *= ib0;
	bq->a0 *= ib0;
	bq->a1 *= ib0;
	bq->a2 *= ib0;
	}

	static inline void set_lp_rbj(BiquadD2 *bq, double fc, double q, double sr, double gain)
	{
	double omega = 2.0 * M_PI * fc / sr;
	double sn = sin(omega);
	double cs = cos(omega);
	double alpha = sn/(2 * q);
	double inv = 1.0/(1.0 + alpha);

	bq->a2 = bq->a0 = gain * inv * (1.0 - cs) * 0.5;
	bq->a1 = bq->a0 + bq->a0;
	bq->b1 = (-2.0 * cs * inv);
	bq->b2 = ((1.0 - alpha) * inv);
	}

	static double freq_gain(BiquadCoeffs *c, double freq, double sr)
	{
	double zr, zi;

	freq = 2.0 M_PI / sr;
	zr = cos(freq);
	zi = -sin(freq);

	/* \|(a0 + a1z + a2z^2)/(1 + b1z + b2z^2)\| */
	return hypot(c->a0 + c->a1zr + c->a2(zrzr-zizi), c->a1zi + 2c->a2zrzi) /
	hypot(1 + c->b1zr + c->b2(zrzr-zizi), c->b1zi + 2c->b2zrzi);
	}

	static int config_input(AVFilterLink *inlink)
	{
	double i, j, k, g, t, a0, a1, a2, b1, b2, tau1, tau2, tau3;
	double cutfreq, gain1kHz, gc, sr = inlink->sample_rate;
	AVFilterContext *ctx = inlink->dst;
	AudioEmphasisContext *s = ctx->priv;
	BiquadCoeffs coeffs;
	int ch;

	s->rc = av_calloc(inlink->channels, sizeof(*s->rc));
	if (!s->rc)
	return AVERROR(ENOMEM);

	switch (s->type) {
	case 0: //"Columbia"
	i = 100.;
	j = 500.;
	k = 1590.;
	break;
	case 1: //"EMI"
	i = 70.;
	j = 500.;
	k = 2500.;
	break;
	case 2: //"BSI(78rpm)"
	i = 50.;
	j = 353.;
	k = 3180.;
	break;
	case 3: //"RIAA"
	default:
	tau1 = 0.003180;
	tau2 = 0.000318;
	tau3 = 0.000075;
	i = 1. / (2. * M_PI * tau1);
	j = 1. / (2. * M_PI * tau2);
	k = 1. / (2. * M_PI * tau3);
	break;
	case 4: //"CD Mastering"
	tau1 = 0.000050;
	tau2 = 0.000015;
	tau3 = 0.0000001;// 1.6MHz out of audible range for null impact
	i = 1. / (2. * M_PI * tau1);
	j = 1. / (2. * M_PI * tau2);
	k = 1. / (2. * M_PI * tau3);
	break;
	case 5: //"50µs FM (Europe)"
	tau1 = 0.000050;
	tau2 = tau1 / 20;// not used
	tau3 = tau1 / 50;//
	i = 1. / (2. * M_PI * tau1);
	j = 1. / (2. * M_PI * tau2);
	k = 1. / (2. * M_PI * tau3);
	break;
	case 6: //"75µs FM (US)"
	tau1 = 0.000075;
	tau2 = tau1 / 20;// not used
	tau3 = tau1 / 50;//
	i = 1. / (2. * M_PI * tau1);
	j = 1. / (2. * M_PI * tau2);
	k = 1. / (2. * M_PI * tau3);
	break;
	}

	i = 2 M_PI;
	j = 2 M_PI;
	k = 2 M_PI;

	t = 1. / sr;

	//swap a1 b1, a2 b2
	if (s->type == 7 \|\| s->type == 8) {
	double tau = (s->type == 7 ? 0.000050 : 0.000075);
	double f = 1.0 / (2 * M_PI * tau);
	double nyq = sr * 0.5;
	double gain = sqrt(1.0 + nyq * nyq / (f * f)); // gain at Nyquist
	double cfreq = sqrt((gain - 1.0) * f * f); // frequency
	double q = 1.0;

	if (s->type == 8)
	q = pow((sr / 3269.0) + 19.5, -0.25); // somewhat poor curve-fit
	if (s->type == 7)
	q = pow((sr / 4750.0) + 19.5, -0.25);
	if (s->mode == 0)
	set_highshelf_rbj(&s->rc[0].r1, cfreq, q, 1. / gain, sr);
	else
	set_highshelf_rbj(&s->rc[0].r1, cfreq, q, gain, sr);
	s->rc[0].use_brickw = 0;
	} else {
	s->rc[0].use_brickw = 1;
	if (s->mode == 0) { // Reproduction
	g = 1. / (4.+2.it+2.kt+ikt*t);
	a0 = (2.t+jtt)g;
	a1 = (2.jtt)g;
	a2 = (-2.t+jtt)g;
	b1 = (-8.+2.iktt)*g;
	b2 = (4.-2.it-2.kt+iktt)g;
	} else { // Production
	g = 1. / (2.t+jt*t);
	a0 = (4.+2.it+2.kt+iktt)g;
	a1 = (-8.+2.iktt)*g;
	a2 = (4.-2.it-2.kt+iktt)g;
	b1 = (2.jtt)g;
	b2 = (-2.t+jtt)g;
	}

	coeffs.a0 = a0;
	coeffs.a1 = a1;
	coeffs.a2 = a2;
	coeffs.b1 = b1;
	coeffs.b2 = b2;

	// the coeffs above give non-normalized value, so it should be normalized to produce 0dB at 1 kHz
	// find actual gain
	// Note: for FM emphasis, use 100 Hz for normalization instead
	gain1kHz = freq_gain(&coeffs, 1000.0, sr);
	// divide one filter's x[n-m] coefficients by that value
	gc = 1.0 / gain1kHz;
	s->rc[0].r1.a0 = coeffs.a0 * gc;
	s->rc[0].r1.a1 = coeffs.a1 * gc;
	s->rc[0].r1.a2 = coeffs.a2 * gc;
	s->rc[0].r1.b1 = coeffs.b1;
	s->rc[0].r1.b2 = coeffs.b2;
	}

	cutfreq = FFMIN(0.45 * sr, 21000.);
	set_lp_rbj(&s->rc[0].brickw, cutfreq, 0.707, sr, 1.);

	for (ch = 1; ch < inlink->channels; ch++) {
	memcpy(&s->rc[ch], &s->rc[0], sizeof(RIAACurve));
	}

	return 0;
	}

	static av_cold void uninit(AVFilterContext *ctx)
	{
	AudioEmphasisContext *s = ctx->priv;
	av_freep(&s->rc);
	}

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

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

	AVFilter ff_af_aemphasis = {
	.name = "aemphasis",
	.description = NULL_IF_CONFIG_SMALL("Audio emphasis."),
	.priv_size = sizeof(AudioEmphasisContext),
	.priv_class = &aemphasis_class,
	.uninit = uninit,
	.query_formats = query_formats,
	.inputs = avfilter_af_aemphasis_inputs,
	.outputs = avfilter_af_aemphasis_outputs,
	};