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/*
* Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
*
* 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/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "filters.h"
#include "formats.h"
#include "audio.h"
enum PulsatorModes { SINE, TRIANGLE, SQUARE, SAWUP, SAWDOWN, NB_MODES };
enum PulsatorTimings { UNIT_BPM, UNIT_MS, UNIT_HZ, NB_TIMINGS };
typedef struct SimpleLFO {
double phase;
double freq;
double offset;
double amount;
double pwidth;
int mode;
int srate;
} SimpleLFO;
typedef struct AudioPulsatorContext {
const AVClass *class;
int mode;
double level_in;
double level_out;
double amount;
double offset_l;
double offset_r;
double pwidth;
double bpm;
double hertz;
int ms;
int timing;
SimpleLFO lfoL, lfoR;
} AudioPulsatorContext;
#define OFFSET(x) offsetof(AudioPulsatorContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption apulsator_options[] = {
{ "level_in", "set input gain", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, },
{ "level_out", "set output gain", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, },
{ "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=SINE}, SINE, NB_MODES-1, FLAGS, .unit = "mode" },
{ "sine", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SINE}, 0, 0, FLAGS, .unit = "mode" },
{ "triangle", NULL, 0, AV_OPT_TYPE_CONST, {.i64=TRIANGLE},0, 0, FLAGS, .unit = "mode" },
{ "square", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SQUARE}, 0, 0, FLAGS, .unit = "mode" },
{ "sawup", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SAWUP}, 0, 0, FLAGS, .unit = "mode" },
{ "sawdown", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SAWDOWN}, 0, 0, FLAGS, .unit = "mode" },
{ "amount", "set modulation", OFFSET(amount), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, FLAGS },
{ "offset_l", "set offset L", OFFSET(offset_l), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, FLAGS },
{ "offset_r", "set offset R", OFFSET(offset_r), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, 0, 1, FLAGS },
{ "width", "set pulse width", OFFSET(pwidth), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 2, FLAGS },
{ "timing", "set timing", OFFSET(timing), AV_OPT_TYPE_INT, {.i64=2}, 0, NB_TIMINGS-1, FLAGS, .unit = "timing" },
{ "bpm", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_BPM}, 0, 0, FLAGS, .unit = "timing" },
{ "ms", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_MS}, 0, 0, FLAGS, .unit = "timing" },
{ "hz", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_HZ}, 0, 0, FLAGS, .unit = "timing" },
{ "bpm", "set BPM", OFFSET(bpm), AV_OPT_TYPE_DOUBLE, {.dbl=120}, 30, 300, FLAGS },
{ "ms", "set ms", OFFSET(ms), AV_OPT_TYPE_INT, {.i64=500}, 10, 2000, FLAGS },
{ "hz", "set frequency", OFFSET(hertz), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0.01, 100, FLAGS },
{ NULL }
};
AVFILTER_DEFINE_CLASS(apulsator);
static void lfo_advance(SimpleLFO *lfo, unsigned count)
{
lfo->phase = fabs(lfo->phase + count * lfo->freq / lfo->srate);
if (lfo->phase >= 1)
lfo->phase = fmod(lfo->phase, 1);
}
static double lfo_get_value(SimpleLFO *lfo)
{
double phs = FFMIN(100, lfo->phase / FFMIN(1.99, FFMAX(0.01, lfo->pwidth)) + lfo->offset);
double val;
if (phs > 1)
phs = fmod(phs, 1.);
switch (lfo->mode) {
case SINE:
val = sin(phs * 2 * M_PI);
break;
case TRIANGLE:
if (phs > 0.75)
val = (phs - 0.75) * 4 - 1;
else if (phs > 0.25)
val = -4 * phs + 2;
else
val = phs * 4;
break;
case SQUARE:
val = phs < 0.5 ? -1 : +1;
break;
case SAWUP:
val = phs * 2 - 1;
break;
case SAWDOWN:
val = 1 - phs * 2;
break;
default: av_assert0(0);
}
return val * lfo->amount;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioPulsatorContext *s = ctx->priv;
const double *src = (const double *)in->data[0];
const int nb_samples = in->nb_samples;
const double level_out = s->level_out;
const double level_in = s->level_in;
const double amount = s->amount;
AVFrame *out;
double *dst;
int n;
if (av_frame_is_writable(in)) {
out = in;
} else {
out = ff_get_audio_buffer(inlink, in->nb_samples);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
}
dst = (double *)out->data[0];
for (n = 0; n < nb_samples; n++) {
double outL;
double outR;
double inL = src[0] * level_in;
double inR = src[1] * level_in;
double procL = inL;
double procR = inR;
procL *= lfo_get_value(&s->lfoL) * 0.5 + amount / 2;
procR *= lfo_get_value(&s->lfoR) * 0.5 + amount / 2;
outL = procL + inL * (1 - amount);
outR = procR + inR * (1 - amount);
outL *= level_out;
outR *= level_out;
dst[0] = outL;
dst[1] = outR;
lfo_advance(&s->lfoL, 1);
lfo_advance(&s->lfoR, 1);
dst += 2;
src += 2;
}
if (in != out)
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static int query_formats(const AVFilterContext *ctx,
AVFilterFormatsConfig **cfg_in,
AVFilterFormatsConfig **cfg_out)
{
static const enum AVSampleFormat formats[] = {
AV_SAMPLE_FMT_DBL,
AV_SAMPLE_FMT_NONE,
};
static const AVChannelLayout layouts[] = {
AV_CHANNEL_LAYOUT_STEREO,
{ .nb_channels = 0 },
};
int ret;
ret = ff_set_common_formats_from_list2(ctx, cfg_in, cfg_out, formats);
if (ret < 0)
return ret;
ret = ff_set_common_channel_layouts_from_list2(ctx, cfg_in, cfg_out, layouts);
if (ret < 0)
return ret;
return 0;
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioPulsatorContext *s = ctx->priv;
double freq;
switch (s->timing) {
case UNIT_BPM: freq = s->bpm / 60; break;
case UNIT_MS: freq = 1 / (s->ms / 1000.); break;
case UNIT_HZ: freq = s->hertz; break;
default: av_assert0(0);
}
s->lfoL.freq = freq;
s->lfoR.freq = freq;
s->lfoL.mode = s->mode;
s->lfoR.mode = s->mode;
s->lfoL.offset = s->offset_l;
s->lfoR.offset = s->offset_r;
s->lfoL.srate = inlink->sample_rate;
s->lfoR.srate = inlink->sample_rate;
s->lfoL.amount = s->amount;
s->lfoR.amount = s->amount;
s->lfoL.pwidth = s->pwidth;
s->lfoR.pwidth = s->pwidth;
return 0;
}
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
.filter_frame = filter_frame,
},
};
const AVFilter ff_af_apulsator = {
.name = "apulsator",
.description = NULL_IF_CONFIG_SMALL("Audio pulsator."),
.priv_size = sizeof(AudioPulsatorContext),
.priv_class = &apulsator_class,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
FILTER_QUERY_FUNC2(query_formats),
};
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