libavfilter/af_aexciter.c - manifest_repos/ffmpeg - Git at Google

 /*
  * Copyright (c) Markus Schmidt and Christian Holschuh
  *
  * 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 ChannelParams {
     double blend_old, drive_old;
     double rdrive, rbdr, kpa, kpb, kna, knb, ap,
            an, imr, kc, srct, sq, pwrq;
     double prev_med, prev_out;

     double hp[5], lp[5];
     double hw[4][2], lw[2][2];
 } ChannelParams;

 typedef struct AExciterContext {
     const AVClass *class;

     double level_in;
     double level_out;
     double amount;
     double drive;
     double blend;
     double freq;
     double ceil;
     int listen;

     ChannelParams *cp;
 } AExciterContext;

 #define OFFSET(x) offsetof(AExciterContext, x)
 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM

 static const AVOption aexciter_options[] = {
     { "level_in",  "set level in",    OFFSET(level_in),  AV_OPT_TYPE_DOUBLE, {.dbl=1},           0, 64, A },
     { "level_out", "set level out",   OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1},           0, 64, A },
     { "amount", "set amount",         OFFSET(amount),    AV_OPT_TYPE_DOUBLE, {.dbl=1},           0, 64, A },
     { "drive", "set harmonics",       OFFSET(drive),     AV_OPT_TYPE_DOUBLE, {.dbl=8.5},       0.1, 10, A },
     { "blend", "set blend harmonics", OFFSET(blend),     AV_OPT_TYPE_DOUBLE, {.dbl=0},         -10, 10, A },
     { "freq", "set scope",            OFFSET(freq),      AV_OPT_TYPE_DOUBLE, {.dbl=7500},  2000, 12000, A },
     { "ceil", "set ceiling",          OFFSET(ceil),      AV_OPT_TYPE_DOUBLE, {.dbl=9999},  9999, 20000, A },
     { "listen", "enable listen mode", OFFSET(listen),    AV_OPT_TYPE_BOOL,   {.i64=0},        0,     1, A },
     { NULL }
 };

 AVFILTER_DEFINE_CLASS(aexciter);

 static inline double M(double x)
 {
     return (fabs(x) > 0.00000001) ? x : 0.0;
 }

 static inline double D(double x)
 {
     x = fabs(x);

     return (x > 0.00000001) ? sqrt(x) : 0.0;
 }

 static void set_params(ChannelParams *p,
                        double blend, double drive,
                        double srate, double freq,
                        double ceil)
 {
     double a0, a1, a2, b0, b1, b2, w0, alpha;

     p->rdrive = 12.0 / drive;
     p->rbdr = p->rdrive / (10.5 - blend) * 780.0 / 33.0;
     p->kpa = D(2.0 * (p->rdrive*p->rdrive) - 1.0) + 1.0;
     p->kpb = (2.0 - p->kpa) / 2.0;
     p->ap = ((p->rdrive*p->rdrive) - p->kpa + 1.0) / 2.0;
     p->kc = p->kpa / D(2.0 * D(2.0 * (p->rdrive*p->rdrive) - 1.0) - 2.0 * p->rdrive*p->rdrive);

     p->srct = (0.1 * srate) / (0.1 * srate + 1.0);
     p->sq = p->kc*p->kc + 1.0;
     p->knb = -1.0 * p->rbdr / D(p->sq);
     p->kna = 2.0 * p->kc * p->rbdr / D(p->sq);
     p->an = p->rbdr*p->rbdr / p->sq;
     p->imr = 2.0 * p->knb + D(2.0 * p->kna + 4.0 * p->an - 1.0);
     p->pwrq = 2.0 / (p->imr + 1.0);

     w0 = 2 * M_PI * freq / srate;
     alpha = sin(w0) / (2. * 0.707);
     a0 =   1 + alpha;
     a1 =  -2 * cos(w0);
     a2 =   1 - alpha;
     b0 =  (1 + cos(w0)) / 2;
     b1 = -(1 + cos(w0));
     b2 =  (1 + cos(w0)) / 2;

     p->hp[0] =-a1 / a0;
     p->hp[1] =-a2 / a0;
     p->hp[2] = b0 / a0;
     p->hp[3] = b1 / a0;
     p->hp[4] = b2 / a0;

     w0 = 2 * M_PI * ceil / srate;
     alpha = sin(w0) / (2. * 0.707);
     a0 =  1 + alpha;
     a1 = -2 * cos(w0);
     a2 =  1 - alpha;
     b0 = (1 - cos(w0)) / 2;
     b1 =  1 - cos(w0);
     b2 = (1 - cos(w0)) / 2;

     p->lp[0] =-a1 / a0;
     p->lp[1] =-a2 / a0;
     p->lp[2] = b0 / a0;
     p->lp[3] = b1 / a0;
     p->lp[4] = b2 / a0;
 }

 static double bprocess(double in, const double *const c,
                        double *w1, double *w2)
 {
     double out = c[2] * in + *w1;

     *w1 = c[3] * in + *w2 + c[0] * out;
     *w2 = c[4] * in + c[1] * out;

     return out;
 }

 static double distortion_process(AExciterContext *s, ChannelParams *p, double in)
 {
     double proc = in, med;

     proc = bprocess(proc, p->hp, &p->hw[0][0], &p->hw[0][1]);
     proc = bprocess(proc, p->hp, &p->hw[1][0], &p->hw[1][1]);

     if (proc >= 0.0) {
         med = (D(p->ap + proc * (p->kpa - proc)) + p->kpb) * p->pwrq;
     } else {
         med = (D(p->an - proc * (p->kna + proc)) + p->knb) * p->pwrq * -1.0;
     }

     proc = p->srct * (med - p->prev_med + p->prev_out);
     p->prev_med = M(med);
     p->prev_out = M(proc);

     proc = bprocess(proc, p->hp, &p->hw[2][0], &p->hw[2][1]);
     proc = bprocess(proc, p->hp, &p->hw[3][0], &p->hw[3][1]);

     if (s->ceil >= 10000.) {
         proc = bprocess(proc, p->lp, &p->lw[0][0], &p->lw[0][1]);
         proc = bprocess(proc, p->lp, &p->lw[1][0], &p->lw[1][1]);
     }

     return proc;
 }

 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 {
     AVFilterContext *ctx = inlink->dst;
     AExciterContext *s = ctx->priv;
     AVFilterLink *outlink = ctx->outputs[0];
     AVFrame *out;
     const double *src = (const double *)in->data[0];
     const double level_in = s->level_in;
     const double level_out = s->level_out;
     const double amount = s->amount;
     const double listen = 1.0 - s->listen;
     double *dst;

     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 (int n = 0; n < in->nb_samples; n++) {
         for (int c = 0; c < inlink->channels; c++) {
             double sample = src[c] * level_in;

             sample = distortion_process(s, &s->cp[c], sample);
             sample = sample * amount + listen * src[c];

             sample *= level_out;
             if (ctx->is_disabled)
                 dst[c] = src[c];
             else
                 dst[c] = sample;
         }

         src += inlink->channels;
         dst += inlink->channels;
     }

     if (in != out)
         av_frame_free(&in);

     return ff_filter_frame(outlink, out);
 }

 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterFormats *formats;
     AVFilterChannelLayouts *layouts;
     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 av_cold void uninit(AVFilterContext *ctx)
 {
     AExciterContext *s = ctx->priv;

     av_freep(&s->cp);
 }

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

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

     for (int i = 0; i < inlink->channels; i++)
         set_params(&s->cp[i], s->blend, s->drive, inlink->sample_rate,
                    s->freq, s->ceil);

     return 0;
 }

 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
                            char *res, int res_len, int flags)
 {
     AVFilterLink *inlink = ctx->inputs[0];
     int ret;

     ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
     if (ret < 0)
         return ret;

     return config_input(inlink);
 }

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

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

 AVFilter ff_af_aexciter = {
     .name          = "aexciter",
     .description   = NULL_IF_CONFIG_SMALL("Enhance high frequency part of audio."),
     .priv_size     = sizeof(AExciterContext),
     .priv_class    = &aexciter_class,
     .uninit        = uninit,
     .query_formats = query_formats,
     .inputs        = avfilter_af_aexciter_inputs,
     .outputs       = avfilter_af_aexciter_outputs,
     .process_command = process_command,
     .flags         = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
 };
	/*
	* Copyright (c) Markus Schmidt and Christian Holschuh
	*
	* 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 ChannelParams {
	double blend_old, drive_old;
	double rdrive, rbdr, kpa, kpb, kna, knb, ap,
	an, imr, kc, srct, sq, pwrq;
	double prev_med, prev_out;

	double hp[5], lp[5];
	double hw[4][2], lw[2][2];
	} ChannelParams;

	typedef struct AExciterContext {
	const AVClass *class;

	double level_in;
	double level_out;
	double amount;
	double drive;
	double blend;
	double freq;
	double ceil;
	int listen;

	ChannelParams *cp;
	} AExciterContext;

	#define OFFSET(x) offsetof(AExciterContext, x)
	#define A AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM\|AV_OPT_FLAG_RUNTIME_PARAM

	static const AVOption aexciter_options[] = {
	{ "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A },
	{ "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A },
	{ "amount", "set amount", OFFSET(amount), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A },
	{ "drive", "set harmonics", OFFSET(drive), AV_OPT_TYPE_DOUBLE, {.dbl=8.5}, 0.1, 10, A },
	{ "blend", "set blend harmonics", OFFSET(blend), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -10, 10, A },
	{ "freq", "set scope", OFFSET(freq), AV_OPT_TYPE_DOUBLE, {.dbl=7500}, 2000, 12000, A },
	{ "ceil", "set ceiling", OFFSET(ceil), AV_OPT_TYPE_DOUBLE, {.dbl=9999}, 9999, 20000, A },
	{ "listen", "enable listen mode", OFFSET(listen), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
	{ NULL }
	};

	AVFILTER_DEFINE_CLASS(aexciter);

	static inline double M(double x)
	{
	return (fabs(x) > 0.00000001) ? x : 0.0;
	}

	static inline double D(double x)
	{
	x = fabs(x);

	return (x > 0.00000001) ? sqrt(x) : 0.0;
	}

	static void set_params(ChannelParams *p,
	double blend, double drive,
	double srate, double freq,
	double ceil)
	{
	double a0, a1, a2, b0, b1, b2, w0, alpha;

	p->rdrive = 12.0 / drive;
	p->rbdr = p->rdrive / (10.5 - blend) * 780.0 / 33.0;
	p->kpa = D(2.0 * (p->rdrive*p->rdrive) - 1.0) + 1.0;
	p->kpb = (2.0 - p->kpa) / 2.0;
	p->ap = ((p->rdrive*p->rdrive) - p->kpa + 1.0) / 2.0;
	p->kc = p->kpa / D(2.0 * D(2.0 * (p->rdrivep->rdrive) - 1.0) - 2.0 p->rdrive*p->rdrive);

	p->srct = (0.1 * srate) / (0.1 * srate + 1.0);
	p->sq = p->kc*p->kc + 1.0;
	p->knb = -1.0 * p->rbdr / D(p->sq);
	p->kna = 2.0 * p->kc * p->rbdr / D(p->sq);
	p->an = p->rbdr*p->rbdr / p->sq;
	p->imr = 2.0 * p->knb + D(2.0 * p->kna + 4.0 * p->an - 1.0);
	p->pwrq = 2.0 / (p->imr + 1.0);

	w0 = 2 * M_PI * freq / srate;
	alpha = sin(w0) / (2. * 0.707);
	a0 = 1 + alpha;
	a1 = -2 * cos(w0);
	a2 = 1 - alpha;
	b0 = (1 + cos(w0)) / 2;
	b1 = -(1 + cos(w0));
	b2 = (1 + cos(w0)) / 2;

	p->hp[0] =-a1 / a0;
	p->hp[1] =-a2 / a0;
	p->hp[2] = b0 / a0;
	p->hp[3] = b1 / a0;
	p->hp[4] = b2 / a0;

	w0 = 2 * M_PI * ceil / srate;
	alpha = sin(w0) / (2. * 0.707);
	a0 = 1 + alpha;
	a1 = -2 * cos(w0);
	a2 = 1 - alpha;
	b0 = (1 - cos(w0)) / 2;
	b1 = 1 - cos(w0);
	b2 = (1 - cos(w0)) / 2;

	p->lp[0] =-a1 / a0;
	p->lp[1] =-a2 / a0;
	p->lp[2] = b0 / a0;
	p->lp[3] = b1 / a0;
	p->lp[4] = b2 / a0;
	}

	static double bprocess(double in, const double *const c,
	double w1, double w2)
	{
	double out = c[2] * in + *w1;

	w1 = c[3] in + w2 + c[0] out;
	w2 = c[4] in + c[1] * out;

	return out;
	}

	static double distortion_process(AExciterContext s, ChannelParams p, double in)
	{
	double proc = in, med;

	proc = bprocess(proc, p->hp, &p->hw[0][0], &p->hw[0][1]);
	proc = bprocess(proc, p->hp, &p->hw[1][0], &p->hw[1][1]);

	if (proc >= 0.0) {
	med = (D(p->ap + proc * (p->kpa - proc)) + p->kpb) * p->pwrq;
	} else {
	med = (D(p->an - proc * (p->kna + proc)) + p->knb) * p->pwrq * -1.0;
	}

	proc = p->srct * (med - p->prev_med + p->prev_out);
	p->prev_med = M(med);
	p->prev_out = M(proc);

	proc = bprocess(proc, p->hp, &p->hw[2][0], &p->hw[2][1]);
	proc = bprocess(proc, p->hp, &p->hw[3][0], &p->hw[3][1]);

	if (s->ceil >= 10000.) {
	proc = bprocess(proc, p->lp, &p->lw[0][0], &p->lw[0][1]);
	proc = bprocess(proc, p->lp, &p->lw[1][0], &p->lw[1][1]);
	}

	return proc;
	}

	static int filter_frame(AVFilterLink inlink, AVFrame in)
	{
	AVFilterContext *ctx = inlink->dst;
	AExciterContext *s = ctx->priv;
	AVFilterLink *outlink = ctx->outputs[0];
	AVFrame *out;
	const double src = (const double )in->data[0];
	const double level_in = s->level_in;
	const double level_out = s->level_out;
	const double amount = s->amount;
	const double listen = 1.0 - s->listen;
	double *dst;

	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 (int n = 0; n < in->nb_samples; n++) {
	for (int c = 0; c < inlink->channels; c++) {
	double sample = src[c] * level_in;

	sample = distortion_process(s, &s->cp[c], sample);
	sample = sample * amount + listen * src[c];

	sample *= level_out;
	if (ctx->is_disabled)
	dst[c] = src[c];
	else
	dst[c] = sample;
	}

	src += inlink->channels;
	dst += inlink->channels;
	}

	if (in != out)
	av_frame_free(&in);

	return ff_filter_frame(outlink, out);
	}

	static int query_formats(AVFilterContext *ctx)
	{
	AVFilterFormats *formats;
	AVFilterChannelLayouts *layouts;
	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 av_cold void uninit(AVFilterContext *ctx)
	{
	AExciterContext *s = ctx->priv;

	av_freep(&s->cp);
	}

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

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

	for (int i = 0; i < inlink->channels; i++)
	set_params(&s->cp[i], s->blend, s->drive, inlink->sample_rate,
	s->freq, s->ceil);

	return 0;
	}

	static int process_command(AVFilterContext ctx, const char cmd, const char *args,
	char *res, int res_len, int flags)
	{
	AVFilterLink *inlink = ctx->inputs[0];
	int ret;

	ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
	if (ret < 0)
	return ret;

	return config_input(inlink);
	}

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

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

	AVFilter ff_af_aexciter = {
	.name = "aexciter",
	.description = NULL_IF_CONFIG_SMALL("Enhance high frequency part of audio."),
	.priv_size = sizeof(AExciterContext),
	.priv_class = &aexciter_class,
	.uninit = uninit,
	.query_formats = query_formats,
	.inputs = avfilter_af_aexciter_inputs,
	.outputs = avfilter_af_aexciter_outputs,
	.process_command = process_command,
	.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
	};