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/*
* Copyright (c) 2006 Rob Sykes <robs@users.sourceforge.net>
*
* 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/mem.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "avfilter.h"
#include "audio.h"
#include "internal.h"
#include "generate_wave_table.h"
#define INTERPOLATION_LINEAR 0
#define INTERPOLATION_QUADRATIC 1
typedef struct FlangerContext {
const AVClass *class;
double delay_min;
double delay_depth;
double feedback_gain;
double delay_gain;
double speed;
int wave_shape;
double channel_phase;
int interpolation;
double in_gain;
int max_samples;
uint8_t **delay_buffer;
int delay_buf_pos;
double *delay_last;
float *lfo;
int lfo_length;
int lfo_pos;
} FlangerContext;
#define OFFSET(x) offsetof(FlangerContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption flanger_options[] = {
{ "delay", "base delay in milliseconds", OFFSET(delay_min), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 30, A },
{ "depth", "added swept delay in milliseconds", OFFSET(delay_depth), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 10, A },
{ "regen", "percentage regeneration (delayed signal feedback)", OFFSET(feedback_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -95, 95, A },
{ "width", "percentage of delayed signal mixed with original", OFFSET(delay_gain), AV_OPT_TYPE_DOUBLE, {.dbl=71}, 0, 100, A },
{ "speed", "sweeps per second (Hz)", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.1, 10, A },
{ "shape", "swept wave shape", OFFSET(wave_shape), AV_OPT_TYPE_INT, {.i64=WAVE_SIN}, WAVE_SIN, WAVE_NB-1, A, .unit = "type" },
{ "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, .unit = "type" },
{ "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, .unit = "type" },
{ "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, .unit = "type" },
{ "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, .unit = "type" },
{ "phase", "swept wave percentage phase-shift for multi-channel", OFFSET(channel_phase), AV_OPT_TYPE_DOUBLE, {.dbl=25}, 0, 100, A },
{ "interp", "delay-line interpolation", OFFSET(interpolation), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, .unit = "itype" },
{ "linear", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_LINEAR}, 0, 0, A, .unit = "itype" },
{ "quadratic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_QUADRATIC}, 0, 0, A, .unit = "itype" },
{ NULL }
};
AVFILTER_DEFINE_CLASS(flanger);
static av_cold int init(AVFilterContext *ctx)
{
FlangerContext *s = ctx->priv;
s->feedback_gain /= 100;
s->delay_gain /= 100;
s->channel_phase /= 100;
s->delay_min /= 1000;
s->delay_depth /= 1000;
s->in_gain = 1 / (1 + s->delay_gain);
s->delay_gain /= 1 + s->delay_gain;
s->delay_gain *= 1 - fabs(s->feedback_gain);
return 0;
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
FlangerContext *s = ctx->priv;
s->max_samples = (s->delay_min + s->delay_depth) * inlink->sample_rate + 2.5;
s->lfo_length = inlink->sample_rate / s->speed;
s->delay_last = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->delay_last));
s->lfo = av_calloc(s->lfo_length, sizeof(*s->lfo));
if (!s->lfo || !s->delay_last)
return AVERROR(ENOMEM);
ff_generate_wave_table(s->wave_shape, AV_SAMPLE_FMT_FLT, s->lfo, s->lfo_length,
rint(s->delay_min * inlink->sample_rate),
s->max_samples - 2., 3 * M_PI_2);
return av_samples_alloc_array_and_samples(&s->delay_buffer, NULL,
inlink->ch_layout.nb_channels, s->max_samples,
inlink->format, 0);
}
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
{
AVFilterContext *ctx = inlink->dst;
FlangerContext *s = ctx->priv;
AVFrame *out_frame;
int chan, i;
if (av_frame_is_writable(frame)) {
out_frame = frame;
} else {
out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
if (!out_frame) {
av_frame_free(&frame);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out_frame, frame);
}
for (i = 0; i < frame->nb_samples; i++) {
s->delay_buf_pos = (s->delay_buf_pos + s->max_samples - 1) % s->max_samples;
for (chan = 0; chan < inlink->ch_layout.nb_channels; chan++) {
double *src = (double *)frame->extended_data[chan];
double *dst = (double *)out_frame->extended_data[chan];
double delayed_0, delayed_1;
double delayed;
double in, out;
int channel_phase = chan * s->lfo_length * s->channel_phase + .5;
double delay = s->lfo[(s->lfo_pos + channel_phase) % s->lfo_length];
int int_delay = (int)delay;
double frac_delay = modf(delay, &delay);
double *delay_buffer = (double *)s->delay_buffer[chan];
in = src[i];
delay_buffer[s->delay_buf_pos] = in + s->delay_last[chan] *
s->feedback_gain;
delayed_0 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
delayed_1 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
if (s->interpolation == INTERPOLATION_LINEAR) {
delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay;
} else {
double a, b;
double delayed_2 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
delayed_2 -= delayed_0;
delayed_1 -= delayed_0;
a = delayed_2 * .5 - delayed_1;
b = delayed_1 * 2 - delayed_2 *.5;
delayed = delayed_0 + (a * frac_delay + b) * frac_delay;
}
s->delay_last[chan] = delayed;
out = in * s->in_gain + delayed * s->delay_gain;
dst[i] = out;
}
s->lfo_pos = (s->lfo_pos + 1) % s->lfo_length;
}
if (frame != out_frame)
av_frame_free(&frame);
return ff_filter_frame(ctx->outputs[0], out_frame);
}
static av_cold void uninit(AVFilterContext *ctx)
{
FlangerContext *s = ctx->priv;
av_freep(&s->lfo);
av_freep(&s->delay_last);
if (s->delay_buffer)
av_freep(&s->delay_buffer[0]);
av_freep(&s->delay_buffer);
}
static const AVFilterPad flanger_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
.filter_frame = filter_frame,
},
};
const AVFilter ff_af_flanger = {
.name = "flanger",
.description = NULL_IF_CONFIG_SMALL("Apply a flanging effect to the audio."),
.priv_size = sizeof(FlangerContext),
.priv_class = &flanger_class,
.init = init,
.uninit = uninit,
FILTER_INPUTS(flanger_inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
};
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