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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"
#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 const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE
};
#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 int nb_coeffs = s->nb_coeffs; \
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 < 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 = nb_coeffs; j < 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[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 int nb_coeffs = s->nb_coeffs; \
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 < 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 = nb_coeffs; j < 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[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->ch_layout.nb_channels * jobnr) / nb_jobs;
const int end = (in->ch_layout.nb_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->ch_layout.nb_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,
},
};
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,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
FILTER_SAMPLEFMTS_ARRAY(sample_fmts),
.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,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
FILTER_SAMPLEFMTS_ARRAY(sample_fmts),
.process_command = ff_filter_process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
AVFILTER_FLAG_SLICE_THREADS,
};
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