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FFmpeg/libavfilter/af_afreqshift.c

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/*
* 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 MAX_NB_COEFFS 16
typedef struct AFreqShift {
const AVClass *class;
double shift;
double level;
int nb_coeffs;
int old_nb_coeffs;
double cd[MAX_NB_COEFFS * 2];
float cf[MAX_NB_COEFFS * 2];
int64_t in_samples;
AVFrame *i1, *o1;
AVFrame *i2, *o2;
void (*filter_channel)(AVFilterContext *ctx,
int channel,
AVFrame *in, AVFrame *out);
} AFreqShift;
static int query_formats(AVFilterContext *ctx)
{
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
int ret = ff_set_common_formats_from_list(ctx, sample_fmts);
if (ret < 0)
return ret;
ret = ff_set_common_all_channel_counts(ctx);
if (ret < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
}
#define PFILTER(name, type, sin, cos, cc) \
static void pfilter_channel_## name(AVFilterContext *ctx, \
int ch, \
AVFrame *in, AVFrame *out) \
{ \
AFreqShift *s = ctx->priv; \
const int nb_samples = in->nb_samples; \
const type *src = (const type *)in->extended_data[ch]; \
type *dst = (type *)out->extended_data[ch]; \
type *i1 = (type *)s->i1->extended_data[ch]; \
type *o1 = (type *)s->o1->extended_data[ch]; \
type *i2 = (type *)s->i2->extended_data[ch]; \
type *o2 = (type *)s->o2->extended_data[ch]; \
const type *c = s->cc; \
const type level = s->level; \
type shift = s->shift * M_PI; \
type cos_theta = cos(shift); \
type sin_theta = sin(shift); \
\
for (int n = 0; n < nb_samples; n++) { \
type xn1 = src[n], xn2 = src[n]; \
type I, Q; \
\
for (int j = 0; j < s->nb_coeffs; 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 = s->nb_coeffs; j < s->nb_coeffs*2; 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[s->nb_coeffs * 2 - 1]; \
\
dst[n] = (I * cos_theta - Q * sin_theta) * level; \
} \
}
PFILTER(flt, float, sin, cos, cf)
PFILTER(dbl, double, sin, cos, cd)
#define FFILTER(name, type, sin, cos, fmod, cc) \
static void ffilter_channel_## name(AVFilterContext *ctx, \
int ch, \
AVFrame *in, AVFrame *out) \
{ \
AFreqShift *s = ctx->priv; \
const int nb_samples = in->nb_samples; \
const type *src = (const type *)in->extended_data[ch]; \
type *dst = (type *)out->extended_data[ch]; \
type *i1 = (type *)s->i1->extended_data[ch]; \
type *o1 = (type *)s->o1->extended_data[ch]; \
type *i2 = (type *)s->i2->extended_data[ch]; \
type *o2 = (type *)s->o2->extended_data[ch]; \
const type *c = s->cc; \
const type level = s->level; \
type ts = 1. / in->sample_rate; \
type shift = s->shift; \
int64_t N = s->in_samples; \
\
for (int n = 0; n < nb_samples; n++) { \
type xn1 = src[n], xn2 = src[n]; \
type I, Q, theta; \
\
for (int j = 0; j < s->nb_coeffs; 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 = s->nb_coeffs; j < s->nb_coeffs*2; 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[s->nb_coeffs * 2 - 1]; \
\
theta = 2. * M_PI * fmod(shift * (N + n) * ts, 1.); \
dst[n] = (I * cos(theta) - Q * sin(theta)) * level; \
} \
}
FFILTER(flt, float, sinf, cosf, fmodf, cf)
FFILTER(dbl, double, sin, cos, fmod, cd)
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_arrd, float *coef_arrf, 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++) {
const int idx = (n / 2) + (n & 1) * nbr_coefs / 2;
coef_arrd[idx] = compute_coef(n, k, q, order);
coef_arrf[idx] = coef_arrd[idx];
}
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AFreqShift *s = ctx->priv;
if (s->old_nb_coeffs != s->nb_coeffs)
compute_coefs(s->cd, s->cf, s->nb_coeffs * 2, 2. * 20. / inlink->sample_rate);
s->old_nb_coeffs = s->nb_coeffs;
s->i1 = ff_get_audio_buffer(inlink, MAX_NB_COEFFS * 2);
s->o1 = ff_get_audio_buffer(inlink, MAX_NB_COEFFS * 2);
s->i2 = ff_get_audio_buffer(inlink, MAX_NB_COEFFS * 2);
s->o2 = ff_get_audio_buffer(inlink, MAX_NB_COEFFS * 2);
if (!s->i1 || !s->o1 || !s->i2 || !s->o2)
return AVERROR(ENOMEM);
if (inlink->format == AV_SAMPLE_FMT_DBLP) {
if (!strcmp(ctx->filter->name, "afreqshift"))
s->filter_channel = ffilter_channel_dbl;
else
s->filter_channel = pfilter_channel_dbl;
} else {
if (!strcmp(ctx->filter->name, "afreqshift"))
s->filter_channel = ffilter_channel_flt;
else
s->filter_channel = pfilter_channel_flt;
}
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, ch, in, out);
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 (s->old_nb_coeffs != s->nb_coeffs)
compute_coefs(s->cd, s->cf, s->nb_coeffs * 2, 2. * 20. / inlink->sample_rate);
s->old_nb_coeffs = s->nb_coeffs;
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;
ff_filter_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 },
{ "order", "set filter order", OFFSET(nb_coeffs),AV_OPT_TYPE_INT, {.i64=8}, 1, MAX_NB_COEFFS, FLAGS },
{ NULL }
};
AVFILTER_DEFINE_CLASS(afreqshift);
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
},
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
};
const AVFilter ff_af_afreqshift = {
.name = "afreqshift",
.description = NULL_IF_CONFIG_SMALL("Apply frequency shifting to input audio."),
.priv_size = sizeof(AFreqShift),
.priv_class = &afreqshift_class,
.uninit = uninit,
2021-08-12 13:05:31 +02:00
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
avfilter: Replace query_formats callback with union of list and callback If one looks at the many query_formats callbacks in existence, one will immediately recognize that there is one type of default callback for video and a slightly different default callback for audio: It is "return ff_set_common_formats_from_list(ctx, pix_fmts);" for video with a filter-specific pix_fmts list. For audio, it is the same with a filter-specific sample_fmts list together with ff_set_common_all_samplerates() and ff_set_common_all_channel_counts(). This commit allows to remove the boilerplate query_formats callbacks by replacing said callback with a union consisting the old callback and pointers for pixel and sample format arrays. For the not uncommon case in which these lists only contain a single entry (besides the sentinel) enum AVPixelFormat and enum AVSampleFormat fields are also added to the union to store them directly in the AVFilter, thereby avoiding a relocation. The state of said union will be contained in a new, dedicated AVFilter field (the nb_inputs and nb_outputs fields have been shrunk to uint8_t in order to create a hole for this new field; this is no problem, as the maximum of all the nb_inputs is four; for nb_outputs it is only two). The state's default value coincides with the earlier default of query_formats being unset, namely that the filter accepts all formats (and also sample rates and channel counts/layouts for audio) provided that these properties agree coincide for all inputs and outputs. By using different union members for audio and video filters the type-unsafety of using the same functions for audio and video lists will furthermore be more confined to formats.c than before. When the new fields are used, they will also avoid allocations: Currently something nearly equivalent to ff_default_query_formats() is called after every successful call to a query_formats callback; yet in the common case that the newly allocated AVFilterFormats are not used at all (namely if there are no free links) these newly allocated AVFilterFormats are freed again without ever being used. Filters no longer using the callback will not exhibit this any more. Reviewed-by: Paul B Mahol <onemda@gmail.com> Reviewed-by: Nicolas George <george@nsup.org> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-09-27 12:07:35 +02:00
FILTER_QUERY_FUNC(query_formats),
.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 },
{ "order", "set filter order",OFFSET(nb_coeffs), AV_OPT_TYPE_INT,{.i64=8}, 1, MAX_NB_COEFFS, FLAGS },
{ NULL }
};
AVFILTER_DEFINE_CLASS(aphaseshift);
const AVFilter ff_af_aphaseshift = {
.name = "aphaseshift",
.description = NULL_IF_CONFIG_SMALL("Apply phase shifting to input audio."),
.priv_size = sizeof(AFreqShift),
.priv_class = &aphaseshift_class,
.uninit = uninit,
2021-08-12 13:05:31 +02:00
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
avfilter: Replace query_formats callback with union of list and callback If one looks at the many query_formats callbacks in existence, one will immediately recognize that there is one type of default callback for video and a slightly different default callback for audio: It is "return ff_set_common_formats_from_list(ctx, pix_fmts);" for video with a filter-specific pix_fmts list. For audio, it is the same with a filter-specific sample_fmts list together with ff_set_common_all_samplerates() and ff_set_common_all_channel_counts(). This commit allows to remove the boilerplate query_formats callbacks by replacing said callback with a union consisting the old callback and pointers for pixel and sample format arrays. For the not uncommon case in which these lists only contain a single entry (besides the sentinel) enum AVPixelFormat and enum AVSampleFormat fields are also added to the union to store them directly in the AVFilter, thereby avoiding a relocation. The state of said union will be contained in a new, dedicated AVFilter field (the nb_inputs and nb_outputs fields have been shrunk to uint8_t in order to create a hole for this new field; this is no problem, as the maximum of all the nb_inputs is four; for nb_outputs it is only two). The state's default value coincides with the earlier default of query_formats being unset, namely that the filter accepts all formats (and also sample rates and channel counts/layouts for audio) provided that these properties agree coincide for all inputs and outputs. By using different union members for audio and video filters the type-unsafety of using the same functions for audio and video lists will furthermore be more confined to formats.c than before. When the new fields are used, they will also avoid allocations: Currently something nearly equivalent to ff_default_query_formats() is called after every successful call to a query_formats callback; yet in the common case that the newly allocated AVFilterFormats are not used at all (namely if there are no free links) these newly allocated AVFilterFormats are freed again without ever being used. Filters no longer using the callback will not exhibit this any more. Reviewed-by: Paul B Mahol <onemda@gmail.com> Reviewed-by: Nicolas George <george@nsup.org> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-09-27 12:07:35 +02:00
FILTER_QUERY_FUNC(query_formats),
.process_command = ff_filter_process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
AVFILTER_FLAG_SLICE_THREADS,
};