[FFmpeg-devel] [PATCH] avfilter: add frequency and phase shift filters
Paul B Mahol
onemda at gmail.com
Tue Oct 20 18:00:30 EEST 2020
Will apply very soon!
On Sun, Oct 18, 2020 at 6:53 PM Paul B Mahol <onemda at gmail.com> wrote:
> Signed-off-by: Paul B Mahol <onemda at 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
>
> + at section afreqshift
> +Apply frequency shift to input audio samples.
> +
> +The filter accepts the following options:
> +
> + at table @option
> + at item shift
> +Specify frequency shift. Allowed range is -INT_MAX to INT_MAX.
> +Default value is 0.0.
> + at end table
> +
> + at 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
>
> + at section aphaseshift
> +Apply phase shift to input audio samples.
> +
> +The filter accepts the following options:
> +
> + at table @option
> + at item shift
> +Specify phase shift. Allowed range is from -1.0 to 1.0.
> +Default value is 0.0.
> + at end table
> +
> + at 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
>
>
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