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

 /*
  * Copyright (c) Paul B Mahol
  * Copyright (c) Laurent de Soras, 2005
  *
  * 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/channel_layout.h"
 #include "libavutil/ffmath.h"
 #include "libavutil/opt.h"
 #include "avfilter.h"
 #include "audio.h"
 #include "formats.h"

 #define NB_COEFS 16

 typedef struct AFreqShift {
     const AVClass *class;

     double shift;
     double level;

     double c[NB_COEFS];

     int64_t in_samples;

     AVFrame *i1, *o1;
     AVFrame *i2, *o2;

     void (*filter_channel)(AVFilterContext *ctx,
                            int nb_samples,
                            int sample_rate,
                            const double *src, double *dst,
                            double *i1, double *o1,
                            double *i2, double *o2);
 } AFreqShift;

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

     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_all_samplerates();
     return ff_set_common_samplerates(ctx, formats);
 }

 static void pfilter_channel(AVFilterContext *ctx,
                             int nb_samples,
                             int sample_rate,
                             const double *src, double *dst,
                             double *i1, double *o1,
                             double *i2, double *o2)
 {
     AFreqShift *s = ctx->priv;
     const double *c = s->c;
     const double level = s->level;
     double shift = s->shift * M_PI;
     double cos_theta = cos(shift);
     double sin_theta = sin(shift);

     for (int n = 0; n < nb_samples; n++) {
         double xn1 = src[n], xn2 = src[n];
         double I, Q;

         for (int j = 0; j < NB_COEFS / 2; j++) {
             I = c[j] * (xn1 + o2[j]) - i2[j];
             i2[j] = i1[j];
             i1[j] = xn1;
             o2[j] = o1[j];
             o1[j] = I;
             xn1 = I;
         }

         for (int j = NB_COEFS / 2; j < NB_COEFS; j++) {
             Q = c[j] * (xn2 + o2[j]) - i2[j];
             i2[j] = i1[j];
             i1[j] = xn2;
             o2[j] = o1[j];
             o1[j] = Q;
             xn2 = Q;
         }
         Q = o2[NB_COEFS - 1];

         dst[n] = (I * cos_theta - Q * sin_theta) * level;
     }
 }

 static void ffilter_channel(AVFilterContext *ctx,
                             int nb_samples,
                             int sample_rate,
                             const double *src, double *dst,
                             double *i1, double *o1,
                             double *i2, double *o2)
 {
     AFreqShift *s = ctx->priv;
     const double *c = s->c;
     const double level = s->level;
     double ts = 1. / sample_rate;
     double shift = s->shift;
     int64_t N = s->in_samples;

     for (int n = 0; n < nb_samples; n++) {
         double xn1 = src[n], xn2 = src[n];
         double I, Q, theta;

         for (int j = 0; j < NB_COEFS / 2; j++) {
             I = c[j] * (xn1 + o2[j]) - i2[j];
             i2[j] = i1[j];
             i1[j] = xn1;
             o2[j] = o1[j];
             o1[j] = I;
             xn1 = I;
         }

         for (int j = NB_COEFS / 2; j < NB_COEFS; j++) {
             Q = c[j] * (xn2 + o2[j]) - i2[j];
             i2[j] = i1[j];
             i1[j] = xn2;
             o2[j] = o1[j];
             o1[j] = Q;
             xn2 = Q;
         }
         Q = o2[NB_COEFS - 1];

         theta = 2. * M_PI * fmod(shift * (N + n) * ts, 1.);
         dst[n] = (I * cos(theta) - Q * sin(theta)) * level;
     }
 }

 static void compute_transition_param(double *K, double *Q, double transition)
 {
     double kksqrt, e, e2, e4, k, q;

     k  = tan((1. - transition * 2.) * M_PI / 4.);
     k *= k;
     kksqrt = pow(1 - k * k, 0.25);
     e = 0.5 * (1. - kksqrt) / (1. + kksqrt);
     e2 = e * e;
     e4 = e2 * e2;
     q = e * (1. + e4 * (2. + e4 * (15. + 150. * e4)));

     *Q = q;
     *K = k;
 }

 static double ipowp(double x, int64_t n)
 {
     double z = 1.;

     while (n != 0) {
         if (n & 1)
             z *= x;
         n >>= 1;
         x *= x;
     }

     return z;
 }

 static double compute_acc_num(double q, int order, int c)
 {
     int64_t i = 0;
     int j = 1;
     double acc = 0.;
     double q_ii1;

     do {
         q_ii1  = ipowp(q, i * (i + 1));
         q_ii1 *= sin((i * 2 + 1) * c * M_PI / order) * j;
         acc   += q_ii1;

         j = -j;
         i++;
     } while (fabs(q_ii1) > 1e-100);

     return acc;
 }

 static double compute_acc_den(double q, int order, int c)
 {
     int64_t i = 1;
     int j = -1;
     double acc = 0.;
     double q_i2;

     do {
         q_i2  = ipowp(q, i * i);
         q_i2 *= cos(i * 2 * c * M_PI / order) * j;
         acc  += q_i2;

         j = -j;
         i++;
     } while (fabs(q_i2) > 1e-100);

     return acc;
 }

 static double compute_coef(int index, double k, double q, int order)
 {
     const int    c    = index + 1;
     const double num  = compute_acc_num(q, order, c) * pow(q, 0.25);
     const double den  = compute_acc_den(q, order, c) + 0.5;
     const double ww   = num / den;
     const double wwsq = ww * ww;

     const double x    = sqrt((1 - wwsq * k) * (1 - wwsq / k)) / (1 + wwsq);
     const double coef = (1 - x) / (1 + x);

     return coef;
 }

 static void compute_coefs(double *coef_arr, int nbr_coefs, double transition)
 {
     const int order = nbr_coefs * 2 + 1;
     double k, q;

     compute_transition_param(&k, &q, transition);

     for (int n = 0; n < nbr_coefs; n++)
         coef_arr[(n / 2) + (n & 1) * nbr_coefs / 2] = compute_coef(n, k, q, order);
 }

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

     compute_coefs(s->c, NB_COEFS, 2. * 20. / inlink->sample_rate);

     s->i1 = ff_get_audio_buffer(inlink, NB_COEFS);
     s->o1 = ff_get_audio_buffer(inlink, NB_COEFS);
     s->i2 = ff_get_audio_buffer(inlink, NB_COEFS);
     s->o2 = ff_get_audio_buffer(inlink, NB_COEFS);
     if (!s->i1 || !s->o1 || !s->i2 || !s->o2)
         return AVERROR(ENOMEM);

     if (!strcmp(ctx->filter->name, "afreqshift"))
         s->filter_channel = ffilter_channel;
     else
         s->filter_channel = pfilter_channel;

     return 0;
 }

 typedef struct ThreadData {
     AVFrame *in, *out;
 } ThreadData;

 static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
 {
     AFreqShift *s = ctx->priv;
     ThreadData *td = arg;
     AVFrame *out = td->out;
     AVFrame *in = td->in;
     const int start = (in->channels * jobnr) / nb_jobs;
     const int end = (in->channels * (jobnr+1)) / nb_jobs;

     for (int ch = start; ch < end; ch++) {
         s->filter_channel(ctx, in->nb_samples,
                           in->sample_rate,
                           (const double *)in->extended_data[ch],
                           (double *)out->extended_data[ch],
                           (double *)s->i1->extended_data[ch],
                           (double *)s->o1->extended_data[ch],
                           (double *)s->i2->extended_data[ch],
                           (double *)s->o2->extended_data[ch]);
     }

     return 0;
 }

 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 {
     AVFilterContext *ctx = inlink->dst;
     AVFilterLink *outlink = ctx->outputs[0];
     AFreqShift *s = ctx->priv;
     AVFrame *out;
     ThreadData td;

     if (av_frame_is_writable(in)) {
         out = in;
     } else {
         out = ff_get_audio_buffer(outlink, in->nb_samples);
         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);
         }
         av_frame_copy_props(out, in);
     }

     td.in = in; td.out = out;
     ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(inlink->channels,
                                                             ff_filter_get_nb_threads(ctx)));

     s->in_samples += in->nb_samples;

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

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

     av_frame_free(&s->i1);
     av_frame_free(&s->o1);
     av_frame_free(&s->i2);
     av_frame_free(&s->o2);
 }

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

 static const AVOption afreqshift_options[] = {
     { "shift", "set frequency shift", OFFSET(shift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -INT_MAX, INT_MAX, FLAGS },
     { "level", "set output level",    OFFSET(level), AV_OPT_TYPE_DOUBLE, {.dbl=1},      0.0,     1.0, FLAGS },
     { NULL }
 };

 AVFILTER_DEFINE_CLASS(afreqshift);

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

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

 AVFilter ff_af_afreqshift = {
     .name            = "afreqshift",
     .description     = NULL_IF_CONFIG_SMALL("Apply frequency shifting to input audio."),
     .query_formats   = query_formats,
     .priv_size       = sizeof(AFreqShift),
     .priv_class      = &afreqshift_class,
     .uninit          = uninit,
     .inputs          = inputs,
     .outputs         = outputs,
     .process_command = ff_filter_process_command,
     .flags           = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
                        AVFILTER_FLAG_SLICE_THREADS,
 };

 static const AVOption aphaseshift_options[] = {
     { "shift", "set phase shift", OFFSET(shift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1.0, 1.0, FLAGS },
     { "level", "set output level",OFFSET(level), AV_OPT_TYPE_DOUBLE, {.dbl=1},  0.0, 1.0, FLAGS },
     { NULL }
 };

 AVFILTER_DEFINE_CLASS(aphaseshift);

 AVFilter ff_af_aphaseshift = {
     .name            = "aphaseshift",
     .description     = NULL_IF_CONFIG_SMALL("Apply phase shifting to input audio."),
     .query_formats   = query_formats,
     .priv_size       = sizeof(AFreqShift),
     .priv_class      = &aphaseshift_class,
     .uninit          = uninit,
     .inputs          = inputs,
     .outputs         = outputs,
     .process_command = ff_filter_process_command,
     .flags           = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
                        AVFILTER_FLAG_SLICE_THREADS,
 };
	/*
	* Copyright (c) Paul B Mahol
	* Copyright (c) Laurent de Soras, 2005
	*
	* 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/channel_layout.h"
	#include "libavutil/ffmath.h"
	#include "libavutil/opt.h"
	#include "avfilter.h"
	#include "audio.h"
	#include "formats.h"

	#define NB_COEFS 16

	typedef struct AFreqShift {
	const AVClass *class;

	double shift;
	double level;

	double c[NB_COEFS];

	int64_t in_samples;

	AVFrame i1, o1;
	AVFrame i2, o2;

	void (filter_channel)(AVFilterContext ctx,
	int nb_samples,
	int sample_rate,
	const double src, double dst,
	double i1, double o1,
	double i2, double o2);
	} AFreqShift;

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

	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_all_samplerates();
	return ff_set_common_samplerates(ctx, formats);
	}

	static void pfilter_channel(AVFilterContext *ctx,
	int nb_samples,
	int sample_rate,
	const double src, double dst,
	double i1, double o1,
	double i2, double o2)
	{
	AFreqShift *s = ctx->priv;
	const double *c = s->c;
	const double level = s->level;
	double shift = s->shift * M_PI;
	double cos_theta = cos(shift);
	double sin_theta = sin(shift);

	for (int n = 0; n < nb_samples; n++) {
	double xn1 = src[n], xn2 = src[n];
	double I, Q;

	for (int j = 0; j < NB_COEFS / 2; j++) {
	I = c[j] * (xn1 + o2[j]) - i2[j];
	i2[j] = i1[j];
	i1[j] = xn1;
	o2[j] = o1[j];
	o1[j] = I;
	xn1 = I;
	}

	for (int j = NB_COEFS / 2; j < NB_COEFS; j++) {
	Q = c[j] * (xn2 + o2[j]) - i2[j];
	i2[j] = i1[j];
	i1[j] = xn2;
	o2[j] = o1[j];
	o1[j] = Q;
	xn2 = Q;
	}
	Q = o2[NB_COEFS - 1];

	dst[n] = (I * cos_theta - Q * sin_theta) * level;
	}
	}

	static void ffilter_channel(AVFilterContext *ctx,
	int nb_samples,
	int sample_rate,
	const double src, double dst,
	double i1, double o1,
	double i2, double o2)
	{
	AFreqShift *s = ctx->priv;
	const double *c = s->c;
	const double level = s->level;
	double ts = 1. / sample_rate;
	double shift = s->shift;
	int64_t N = s->in_samples;

	for (int n = 0; n < nb_samples; n++) {
	double xn1 = src[n], xn2 = src[n];
	double I, Q, theta;

	for (int j = 0; j < NB_COEFS / 2; j++) {
	I = c[j] * (xn1 + o2[j]) - i2[j];
	i2[j] = i1[j];
	i1[j] = xn1;
	o2[j] = o1[j];
	o1[j] = I;
	xn1 = I;
	}

	for (int j = NB_COEFS / 2; j < NB_COEFS; j++) {
	Q = c[j] * (xn2 + o2[j]) - i2[j];
	i2[j] = i1[j];
	i1[j] = xn2;
	o2[j] = o1[j];
	o1[j] = Q;
	xn2 = Q;
	}
	Q = o2[NB_COEFS - 1];

	theta = 2. * M_PI * fmod(shift * (N + n) * ts, 1.);
	dst[n] = (I * cos(theta) - Q * sin(theta)) * level;
	}
	}

	static void compute_transition_param(double K, double Q, double transition)
	{
	double kksqrt, e, e2, e4, k, q;

	k = tan((1. - transition * 2.) * M_PI / 4.);
	k *= k;
	kksqrt = pow(1 - k * k, 0.25);
	e = 0.5 * (1. - kksqrt) / (1. + kksqrt);
	e2 = e * e;
	e4 = e2 * e2;
	q = e * (1. + e4 * (2. + e4 * (15. + 150. * e4)));

	*Q = q;
	*K = k;
	}

	static double ipowp(double x, int64_t n)
	{
	double z = 1.;

	while (n != 0) {
	if (n & 1)
	z *= x;
	n >>= 1;
	x *= x;
	}

	return z;
	}

	static double compute_acc_num(double q, int order, int c)
	{
	int64_t i = 0;
	int j = 1;
	double acc = 0.;
	double q_ii1;

	do {
	q_ii1 = ipowp(q, i * (i + 1));
	q_ii1 = sin((i 2 + 1) * c * M_PI / order) * j;
	acc += q_ii1;

	j = -j;
	i++;
	} while (fabs(q_ii1) > 1e-100);

	return acc;
	}

	static double compute_acc_den(double q, int order, int c)
	{
	int64_t i = 1;
	int j = -1;
	double acc = 0.;
	double q_i2;

	do {
	q_i2 = ipowp(q, i * i);
	q_i2 = cos(i 2 * c * M_PI / order) * j;
	acc += q_i2;

	j = -j;
	i++;
	} while (fabs(q_i2) > 1e-100);

	return acc;
	}

	static double compute_coef(int index, double k, double q, int order)
	{
	const int c = index + 1;
	const double num = compute_acc_num(q, order, c) * pow(q, 0.25);
	const double den = compute_acc_den(q, order, c) + 0.5;
	const double ww = num / den;
	const double wwsq = ww * ww;

	const double x = sqrt((1 - wwsq * k) * (1 - wwsq / k)) / (1 + wwsq);
	const double coef = (1 - x) / (1 + x);

	return coef;
	}

	static void compute_coefs(double *coef_arr, int nbr_coefs, double transition)
	{
	const int order = nbr_coefs * 2 + 1;
	double k, q;

	compute_transition_param(&k, &q, transition);

	for (int n = 0; n < nbr_coefs; n++)
	coef_arr[(n / 2) + (n & 1) * nbr_coefs / 2] = compute_coef(n, k, q, order);
	}

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

	compute_coefs(s->c, NB_COEFS, 2. * 20. / inlink->sample_rate);

	s->i1 = ff_get_audio_buffer(inlink, NB_COEFS);
	s->o1 = ff_get_audio_buffer(inlink, NB_COEFS);
	s->i2 = ff_get_audio_buffer(inlink, NB_COEFS);
	s->o2 = ff_get_audio_buffer(inlink, NB_COEFS);
	if (!s->i1 \|\| !s->o1 \|\| !s->i2 \|\| !s->o2)
	return AVERROR(ENOMEM);

	if (!strcmp(ctx->filter->name, "afreqshift"))
	s->filter_channel = ffilter_channel;
	else
	s->filter_channel = pfilter_channel;

	return 0;
	}

	typedef struct ThreadData {
	AVFrame in, out;
	} ThreadData;

	static int filter_channels(AVFilterContext ctx, void arg, int jobnr, int nb_jobs)
	{
	AFreqShift *s = ctx->priv;
	ThreadData *td = arg;
	AVFrame *out = td->out;
	AVFrame *in = td->in;
	const int start = (in->channels * jobnr) / nb_jobs;
	const int end = (in->channels * (jobnr+1)) / nb_jobs;

	for (int ch = start; ch < end; ch++) {
	s->filter_channel(ctx, in->nb_samples,
	in->sample_rate,
	(const double *)in->extended_data[ch],
	(double *)out->extended_data[ch],
	(double *)s->i1->extended_data[ch],
	(double *)s->o1->extended_data[ch],
	(double *)s->i2->extended_data[ch],
	(double *)s->o2->extended_data[ch]);
	}

	return 0;
	}

	static int filter_frame(AVFilterLink inlink, AVFrame in)
	{
	AVFilterContext *ctx = inlink->dst;
	AVFilterLink *outlink = ctx->outputs[0];
	AFreqShift *s = ctx->priv;
	AVFrame *out;
	ThreadData td;

	if (av_frame_is_writable(in)) {
	out = in;
	} else {
	out = ff_get_audio_buffer(outlink, in->nb_samples);
	if (!out) {
	av_frame_free(&in);
	return AVERROR(ENOMEM);
	}
	av_frame_copy_props(out, in);
	}

	td.in = in; td.out = out;
	ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(inlink->channels,
	ff_filter_get_nb_threads(ctx)));

	s->in_samples += in->nb_samples;

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

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

	av_frame_free(&s->i1);
	av_frame_free(&s->o1);
	av_frame_free(&s->i2);
	av_frame_free(&s->o2);
	}

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

	static const AVOption afreqshift_options[] = {
	{ "shift", "set frequency shift", OFFSET(shift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -INT_MAX, INT_MAX, FLAGS },
	{ "level", "set output level", OFFSET(level), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0, 1.0, FLAGS },
	{ NULL }
	};

	AVFILTER_DEFINE_CLASS(afreqshift);

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

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

	AVFilter ff_af_afreqshift = {
	.name = "afreqshift",
	.description = NULL_IF_CONFIG_SMALL("Apply frequency shifting to input audio."),
	.query_formats = query_formats,
	.priv_size = sizeof(AFreqShift),
	.priv_class = &afreqshift_class,
	.uninit = uninit,
	.inputs = inputs,
	.outputs = outputs,
	.process_command = ff_filter_process_command,
	.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC \|
	AVFILTER_FLAG_SLICE_THREADS,
	};

	static const AVOption aphaseshift_options[] = {
	{ "shift", "set phase shift", OFFSET(shift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1.0, 1.0, FLAGS },
	{ "level", "set output level",OFFSET(level), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0, 1.0, FLAGS },
	{ NULL }
	};

	AVFILTER_DEFINE_CLASS(aphaseshift);

	AVFilter ff_af_aphaseshift = {
	.name = "aphaseshift",
	.description = NULL_IF_CONFIG_SMALL("Apply phase shifting to input audio."),
	.query_formats = query_formats,
	.priv_size = sizeof(AFreqShift),
	.priv_class = &aphaseshift_class,
	.uninit = uninit,
	.inputs = inputs,
	.outputs = outputs,
	.process_command = ff_filter_process_command,
	.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC \|
	AVFILTER_FLAG_SLICE_THREADS,
	};