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avfilter: add flanger filter

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Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol 2014-06-27 08:42:35 +00:00
parent 7e8c1f0c38
commit b52c26c66f
6 changed files with 281 additions and 1 deletions
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1
Changelog
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@ -30,6 +30,7 @@ version <next>:
- zoompan filter
- signalstats filter
- hqx filter (hq2x, hq3x, hq4x)
- flanger filter
version 2.2:

36
doc/filters.texi
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@ -1439,6 +1439,42 @@ equalizer=f=1000:width_type=q:width=1:g=2,equalizer=f=100:width_type=q:width=2:g
@end example
@end itemize
@section flanger
Apply a flanging effect to the audio.
The filter accepts the following options:
@table @option
@item delay
Set base delay in milliseconds. Range from 0 to 30. Default value is 0.
@item depth
Set added swep delay in milliseconds. Range from 0 to 10. Default value is 2.
@item regen
Set percentage regeneneration (delayed signal feedback). Range from -95 to 95.
Default value is 0.
@item width
Set percentage of delayed signal mixed with original. Range from 0 to 100.
Default valu is 71.
@item speed
Set sweeps per second (Hz). Range from 0.1 to 10. Default value is 0.5.
@item shape
Set swept wave shape, can be @var{triangular} or @var{sinusoidal}.
Default value is @var{sinusoidal}.
@item phase
Set swept wave percentage-shift for multi channel. Range from 0 to 100.
Default value is 25.
@item interp
Set delay-line interpolation, @var{linear} or @var{quadratic}.
Default is @var{linear}.
@end table
@section highpass
Apply a high-pass filter with 3dB point frequency.

1
libavfilter/Makefile
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@ -69,6 +69,7 @@ OBJS-$(CONFIG_COMPAND_FILTER) += af_compand.o
OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
OBJS-$(CONFIG_EBUR128_FILTER) += f_ebur128.o
OBJS-$(CONFIG_EQUALIZER_FILTER) += af_biquads.o
OBJS-$(CONFIG_FLANGER_FILTER) += af_flanger.o generate_wave_table.o
OBJS-$(CONFIG_HIGHPASS_FILTER) += af_biquads.o
OBJS-$(CONFIG_JOIN_FILTER) += af_join.o
OBJS-$(CONFIG_LADSPA_FILTER) += af_ladspa.o

241
libavfilter/af_flanger.c Normal file
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@ -0,0 +1,241 @@
/*
* 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/avstring.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, "type" },
{ "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, "type" },
{ "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, "type" },
{ "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, "type" },
{ "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, "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, "itype" },
{ "linear", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_LINEAR}, 0, 0, A, "itype" },
{ "quadratic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_QUADRATIC}, 0, 0, A, "itype" },
{ NULL }
};
AVFILTER_DEFINE_CLASS(flanger);
static 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 query_formats(AVFilterContext *ctx)
{
AVFilterChannelLayouts *layouts;
AVFilterFormats *formats;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE
};
layouts = ff_all_channel_layouts();
if (!layouts)
return AVERROR(ENOMEM);
ff_set_common_channel_layouts(ctx, layouts);
formats = ff_make_format_list(sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
ff_set_common_formats(ctx, formats);
formats = ff_all_samplerates();
if (!formats)
return AVERROR(ENOMEM);
ff_set_common_samplerates(ctx, formats);
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->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,
floor(s->delay_min * inlink->sample_rate + 0.5),
s->max_samples - 2., 3 * M_PI_2);
return av_samples_alloc_array_and_samples(&s->delay_buffer, NULL,
inlink->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(inlink, frame->nb_samples);
if (!out_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->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,
},
{ NULL }
};
static const AVFilterPad flanger_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};
AVFilter ff_af_flanger = {
.name = "flanger",
.description = NULL_IF_CONFIG_SMALL("Apply a flanging effect to the audio."),
.query_formats = query_formats,
.priv_size = sizeof(FlangerContext),
.priv_class = &flanger_class,
.init = init,
.uninit = uninit,
.inputs = flanger_inputs,
.outputs = flanger_outputs,
};

1
libavfilter/allfilters.c
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@ -87,6 +87,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(EARWAX, earwax, af);
REGISTER_FILTER(EBUR128, ebur128, af);
REGISTER_FILTER(EQUALIZER, equalizer, af);
REGISTER_FILTER(FLANGER, flanger, af);
REGISTER_FILTER(HIGHPASS, highpass, af);
REGISTER_FILTER(JOIN, join, af);
REGISTER_FILTER(LADSPA, ladspa, af);

2
libavfilter/version.h
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@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 4
#define LIBAVFILTER_VERSION_MINOR 9
#define LIBAVFILTER_VERSION_MINOR 10
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \