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[FFmpeg-devel] avfilter: add frequency and phase shift filters

Message ID 20201018165341.21825-1-onemda@gmail.com
State New
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Series [FFmpeg-devel] avfilter: add frequency and phase shift filters
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Commit Message

Paul B Mahol Oct. 18, 2020, 4:53 p.m. UTC
Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
Now with better output quality.
---
 doc/filters.texi            |  30 +++
 libavfilter/Makefile        |   2 +
 libavfilter/af_afreqshift.c | 379 ++++++++++++++++++++++++++++++++++++
 libavfilter/allfilters.c    |   2 +
 4 files changed, 413 insertions(+)
 create mode 100644 libavfilter/af_afreqshift.c

Comments

Paul B Mahol Oct. 20, 2020, 3 p.m. UTC | #1
Will apply very soon!

On Sun, Oct 18, 2020 at 6:53 PM Paul B Mahol <onemda@gmail.com> wrote:

> Signed-off-by: Paul B Mahol <onemda@gmail.com>
> ---
> Now with better output quality.
> ---
>  doc/filters.texi            |  30 +++
>  libavfilter/Makefile        |   2 +
>  libavfilter/af_afreqshift.c | 379 ++++++++++++++++++++++++++++++++++++
>  libavfilter/allfilters.c    |   2 +
>  4 files changed, 413 insertions(+)
>  create mode 100644 libavfilter/af_afreqshift.c
>
> diff --git a/doc/filters.texi b/doc/filters.texi
> index 037a37be23..34207ed0b6 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -1314,6 +1314,21 @@ Force the output to either unsigned 8-bit or signed
> 16-bit stereo
>  aformat=sample_fmts=u8|s16:channel_layouts=stereo
>  @end example
>
> +@section afreqshift
> +Apply frequency shift to input audio samples.
> +
> +The filter accepts the following options:
> +
> +@table @option
> +@item shift
> +Specify frequency shift. Allowed range is -INT_MAX to INT_MAX.
> +Default value is 0.0.
> +@end table
> +
> +@subsection Commands
> +
> +This filter supports the above option as @ref{commands}.
> +
>  @section agate
>
>  A gate is mainly used to reduce lower parts of a signal. This kind of
> signal
> @@ -2064,6 +2079,21 @@ It accepts the following values:
>  @end table
>  @end table
>
> +@section aphaseshift
> +Apply phase shift to input audio samples.
> +
> +The filter accepts the following options:
> +
> +@table @option
> +@item shift
> +Specify phase shift. Allowed range is from -1.0 to 1.0.
> +Default value is 0.0.
> +@end table
> +
> +@subsection Commands
> +
> +This filter supports the above option as @ref{commands}.
> +
>  @section apulsator
>
>  Audio pulsator is something between an autopanner and a tremolo.
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 2691612179..480e191987 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -50,6 +50,7 @@ OBJS-$(CONFIG_AFFTDN_FILTER)                 +=
> af_afftdn.o
>  OBJS-$(CONFIG_AFFTFILT_FILTER)               += af_afftfilt.o
>  OBJS-$(CONFIG_AFIR_FILTER)                   += af_afir.o
>  OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o
> +OBJS-$(CONFIG_AFREQSHIFT_FILTER)             += af_afreqshift.o
>  OBJS-$(CONFIG_AGATE_FILTER)                  += af_agate.o
>  OBJS-$(CONFIG_AIIR_FILTER)                   += af_aiir.o
>  OBJS-$(CONFIG_AINTEGRAL_FILTER)              += af_aderivative.o
> @@ -69,6 +70,7 @@ OBJS-$(CONFIG_ANULL_FILTER)                  +=
> af_anull.o
>  OBJS-$(CONFIG_APAD_FILTER)                   += af_apad.o
>  OBJS-$(CONFIG_APERMS_FILTER)                 += f_perms.o
>  OBJS-$(CONFIG_APHASER_FILTER)                += af_aphaser.o
> generate_wave_table.o
> +OBJS-$(CONFIG_APHASESHIFT_FILTER)            += af_afreqshift.o
>  OBJS-$(CONFIG_APULSATOR_FILTER)              += af_apulsator.o
>  OBJS-$(CONFIG_AREALTIME_FILTER)              += f_realtime.o
>  OBJS-$(CONFIG_ARESAMPLE_FILTER)              += af_aresample.o
> diff --git a/libavfilter/af_afreqshift.c b/libavfilter/af_afreqshift.c
> new file mode 100644
> index 0000000000..e83575813d
> --- /dev/null
> +++ b/libavfilter/af_afreqshift.c
> @@ -0,0 +1,379 @@
> +/*
> + * 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 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;
> +    double *c = s->c;
> +    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;
> +    }
> +}
> +
> +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;
> +    double *c = s->c;
> +    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);
> +    }
> +}
> +
> +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;
> +}
> +
> +static int filter_frame(AVFilterLink *inlink, AVFrame *in)
> +{
> +    AVFilterContext *ctx = inlink->dst;
> +    AVFilterLink *outlink = ctx->outputs[0];
> +    AFreqShift *s = ctx->priv;
> +    AVFrame *out;
> +
> +    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);
> +    }
> +
> +    for (int ch = 0; ch < in->channels; 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]);
> +    }
> +
> +    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 },
> +    { 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,
> +};
> +
> +static const AVOption aphaseshift_options[] = {
> +    { "shift", "set phase shift", OFFSET(shift), AV_OPT_TYPE_DOUBLE,
> {.dbl=0}, -1.0, 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,
> +};
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index 26a8e87b0b..a5ec6bd4ca 100644
> --- a/libavfilter/allfilters.c
> +++ b/libavfilter/allfilters.c
> @@ -43,6 +43,7 @@ extern AVFilter ff_af_afftdn;
>  extern AVFilter ff_af_afftfilt;
>  extern AVFilter ff_af_afir;
>  extern AVFilter ff_af_aformat;
> +extern AVFilter ff_af_afreqshift;
>  extern AVFilter ff_af_agate;
>  extern AVFilter ff_af_aiir;
>  extern AVFilter ff_af_aintegral;
> @@ -62,6 +63,7 @@ extern AVFilter ff_af_anull;
>  extern AVFilter ff_af_apad;
>  extern AVFilter ff_af_aperms;
>  extern AVFilter ff_af_aphaser;
> +extern AVFilter ff_af_aphaseshift;
>  extern AVFilter ff_af_apulsator;
>  extern AVFilter ff_af_arealtime;
>  extern AVFilter ff_af_aresample;
> --
> 2.17.1
>
>
diff mbox series

Patch

diff --git a/doc/filters.texi b/doc/filters.texi
index 037a37be23..34207ed0b6 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1314,6 +1314,21 @@  Force the output to either unsigned 8-bit or signed 16-bit stereo
 aformat=sample_fmts=u8|s16:channel_layouts=stereo
 @end example
 
+@section afreqshift
+Apply frequency shift to input audio samples.
+
+The filter accepts the following options:
+
+@table @option
+@item shift
+Specify frequency shift. Allowed range is -INT_MAX to INT_MAX.
+Default value is 0.0.
+@end table
+
+@subsection Commands
+
+This filter supports the above option as @ref{commands}.
+
 @section agate
 
 A gate is mainly used to reduce lower parts of a signal. This kind of signal
@@ -2064,6 +2079,21 @@  It accepts the following values:
 @end table
 @end table
 
+@section aphaseshift
+Apply phase shift to input audio samples.
+
+The filter accepts the following options:
+
+@table @option
+@item shift
+Specify phase shift. Allowed range is from -1.0 to 1.0.
+Default value is 0.0.
+@end table
+
+@subsection Commands
+
+This filter supports the above option as @ref{commands}.
+
 @section apulsator
 
 Audio pulsator is something between an autopanner and a tremolo.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 2691612179..480e191987 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -50,6 +50,7 @@  OBJS-$(CONFIG_AFFTDN_FILTER)                 += af_afftdn.o
 OBJS-$(CONFIG_AFFTFILT_FILTER)               += af_afftfilt.o
 OBJS-$(CONFIG_AFIR_FILTER)                   += af_afir.o
 OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o
+OBJS-$(CONFIG_AFREQSHIFT_FILTER)             += af_afreqshift.o
 OBJS-$(CONFIG_AGATE_FILTER)                  += af_agate.o
 OBJS-$(CONFIG_AIIR_FILTER)                   += af_aiir.o
 OBJS-$(CONFIG_AINTEGRAL_FILTER)              += af_aderivative.o
@@ -69,6 +70,7 @@  OBJS-$(CONFIG_ANULL_FILTER)                  += af_anull.o
 OBJS-$(CONFIG_APAD_FILTER)                   += af_apad.o
 OBJS-$(CONFIG_APERMS_FILTER)                 += f_perms.o
 OBJS-$(CONFIG_APHASER_FILTER)                += af_aphaser.o generate_wave_table.o
+OBJS-$(CONFIG_APHASESHIFT_FILTER)            += af_afreqshift.o
 OBJS-$(CONFIG_APULSATOR_FILTER)              += af_apulsator.o
 OBJS-$(CONFIG_AREALTIME_FILTER)              += f_realtime.o
 OBJS-$(CONFIG_ARESAMPLE_FILTER)              += af_aresample.o
diff --git a/libavfilter/af_afreqshift.c b/libavfilter/af_afreqshift.c
new file mode 100644
index 0000000000..e83575813d
--- /dev/null
+++ b/libavfilter/af_afreqshift.c
@@ -0,0 +1,379 @@ 
+/*
+ * 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 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;
+    double *c = s->c;
+    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;
+    }
+}
+
+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;
+    double *c = s->c;
+    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);
+    }
+}
+
+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;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AVFilterLink *outlink = ctx->outputs[0];
+    AFreqShift *s = ctx->priv;
+    AVFrame *out;
+
+    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);
+    }
+
+    for (int ch = 0; ch < in->channels; 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]);
+    }
+
+    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 },
+    { 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,
+};
+
+static const AVOption aphaseshift_options[] = {
+    { "shift", "set phase shift", OFFSET(shift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1.0, 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,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 26a8e87b0b..a5ec6bd4ca 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -43,6 +43,7 @@  extern AVFilter ff_af_afftdn;
 extern AVFilter ff_af_afftfilt;
 extern AVFilter ff_af_afir;
 extern AVFilter ff_af_aformat;
+extern AVFilter ff_af_afreqshift;
 extern AVFilter ff_af_agate;
 extern AVFilter ff_af_aiir;
 extern AVFilter ff_af_aintegral;
@@ -62,6 +63,7 @@  extern AVFilter ff_af_anull;
 extern AVFilter ff_af_apad;
 extern AVFilter ff_af_aperms;
 extern AVFilter ff_af_aphaser;
+extern AVFilter ff_af_aphaseshift;
 extern AVFilter ff_af_apulsator;
 extern AVFilter ff_af_arealtime;
 extern AVFilter ff_af_aresample;