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avfilter: add aexciter audio filter

tags/n4.4
Paul B Mahol 5 years ago
parent
129978af6b
commit
579e4e57a2
6 changed files with 377 additions and 1 deletions
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  1. +1
    -0
      Changelog
  2. +56
    -0
      doc/filters.texi
  3. +1
    -0
      libavfilter/Makefile
  4. +317
    -0
      libavfilter/af_aexciter.c
  5. +1
    -0
      libavfilter/allfilters.c
  6. +1
    -1
      libavfilter/version.h

+ 1
- 0
Changelog View File

@@ -69,6 +69,7 @@ version <next>:
- xbm_pipe demuxer
- colorize filter
- CRI parser
- aexciter audio filter


version 4.3:


+ 56
- 0
doc/filters.texi View File

@@ -1003,6 +1003,62 @@ aeval=val(0)|-val(1)
@end example
@end itemize

@section aexciter

An exciter is used to produce high sound that is not present in the
original signal. This is done by creating harmonic distortions of the
signal which are restricted in range and added to the original signal.
An Exciter raises the upper end of an audio signal without simply raising
the higher frequencies like an equalizer would do to create a more
"crisp" or "brilliant" sound.

The filter accepts the following options:

@table @option
@item level_in
Set input level prior processing of signal.
Allowed range is from 0 to 64.
Default value is 1.

@item level_out
Set output level after processing of signal.
Allowed range is from 0 to 64.
Default value is 1.

@item amount
Set the amount of harmonics added to original signal.
Allowed range is from 0 to 64.
Default value is 1.

@item drive
Set the amount of newly created harmonics.
Allowed range is from 0.1 to 10.
Default value is 8.5.

@item blend
Set the octave of newly created harmonics.
Allowed range is from -10 to 10.
Default value is 0.

@item freq
Set the lower frequency limit of producing harmonics in Hz.
Allowed range is from 2000 to 12000 Hz.
Default is 7500 Hz.

@item ceil
Set the upper frequency limit of producing harmonics.
Allowed range is from 9999 to 20000 Hz.
If value is lower than 10000 Hz no limit is applied.

@item listen
Mute the original signal and output only added harmonics.
By default is disabled.
@end table

@subsection Commands

This filter supports the all above options as @ref{commands}.

@anchor{afade}
@section afade



+ 1
- 0
libavfilter/Makefile View File

@@ -46,6 +46,7 @@ OBJS-$(CONFIG_ADERIVATIVE_FILTER) += af_aderivative.o
OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o
OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o
OBJS-$(CONFIG_AEXCITER_FILTER) += af_aexciter.o
OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
OBJS-$(CONFIG_AFFTDN_FILTER) += af_afftdn.o
OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o


+ 317
- 0
libavfilter/af_aexciter.c View File

@@ -0,0 +1,317 @@
/*
* Copyright (c) Markus Schmidt and Christian Holschuh
*
* 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/opt.h"
#include "avfilter.h"
#include "internal.h"
#include "audio.h"

typedef struct ChannelParams {
double blend_old, drive_old;
double rdrive, rbdr, kpa, kpb, kna, knb, ap,
an, imr, kc, srct, sq, pwrq;
double prev_med, prev_out;

double hp[5], lp[5];
double hw[4][2], lw[2][2];
} ChannelParams;

typedef struct AExciterContext {
const AVClass *class;

double level_in;
double level_out;
double amount;
double drive;
double blend;
double freq;
double ceil;
int listen;

ChannelParams *cp;
} AExciterContext;

#define OFFSET(x) offsetof(AExciterContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM

static const AVOption aexciter_options[] = {
{ "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A },
{ "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A },
{ "amount", "set amount", OFFSET(amount), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A },
{ "drive", "set harmonics", OFFSET(drive), AV_OPT_TYPE_DOUBLE, {.dbl=8.5}, 0.1, 10, A },
{ "blend", "set blend harmonics", OFFSET(blend), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -10, 10, A },
{ "freq", "set scope", OFFSET(freq), AV_OPT_TYPE_DOUBLE, {.dbl=7500}, 2000, 12000, A },
{ "ceil", "set ceiling", OFFSET(ceil), AV_OPT_TYPE_DOUBLE, {.dbl=9999}, 9999, 20000, A },
{ "listen", "enable listen mode", OFFSET(listen), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
{ NULL }
};

AVFILTER_DEFINE_CLASS(aexciter);

static inline double M(double x)
{
return (fabs(x) > 0.00000001) ? x : 0.0;
}

static inline double D(double x)
{
x = fabs(x);

return (x > 0.00000001) ? sqrt(x) : 0.0;
}

static void set_params(ChannelParams *p,
double blend, double drive,
double srate, double freq,
double ceil)
{
double a0, a1, a2, b0, b1, b2, w0, alpha;

p->rdrive = 12.0 / drive;
p->rbdr = p->rdrive / (10.5 - blend) * 780.0 / 33.0;
p->kpa = D(2.0 * (p->rdrive*p->rdrive) - 1.0) + 1.0;
p->kpb = (2.0 - p->kpa) / 2.0;
p->ap = ((p->rdrive*p->rdrive) - p->kpa + 1.0) / 2.0;
p->kc = p->kpa / D(2.0 * D(2.0 * (p->rdrive*p->rdrive) - 1.0) - 2.0 * p->rdrive*p->rdrive);

p->srct = (0.1 * srate) / (0.1 * srate + 1.0);
p->sq = p->kc*p->kc + 1.0;
p->knb = -1.0 * p->rbdr / D(p->sq);
p->kna = 2.0 * p->kc * p->rbdr / D(p->sq);
p->an = p->rbdr*p->rbdr / p->sq;
p->imr = 2.0 * p->knb + D(2.0 * p->kna + 4.0 * p->an - 1.0);
p->pwrq = 2.0 / (p->imr + 1.0);

w0 = 2 * M_PI * freq / srate;
alpha = sin(w0) / (2. * 0.707);
a0 = 1 + alpha;
a1 = -2 * cos(w0);
a2 = 1 - alpha;
b0 = (1 + cos(w0)) / 2;
b1 = -(1 + cos(w0));
b2 = (1 + cos(w0)) / 2;

p->hp[0] =-a1 / a0;
p->hp[1] =-a2 / a0;
p->hp[2] = b0 / a0;
p->hp[3] = b1 / a0;
p->hp[4] = b2 / a0;

w0 = 2 * M_PI * ceil / srate;
alpha = sin(w0) / (2. * 0.707);
a0 = 1 + alpha;
a1 = -2 * cos(w0);
a2 = 1 - alpha;
b0 = (1 - cos(w0)) / 2;
b1 = 1 - cos(w0);
b2 = (1 - cos(w0)) / 2;

p->lp[0] =-a1 / a0;
p->lp[1] =-a2 / a0;
p->lp[2] = b0 / a0;
p->lp[3] = b1 / a0;
p->lp[4] = b2 / a0;
}

static double bprocess(double in, const double *const c,
double *w1, double *w2)
{
double out = c[2] * in + *w1;

*w1 = c[3] * in + *w2 + c[0] * out;
*w2 = c[4] * in + c[1] * out;

return out;
}

static double distortion_process(AExciterContext *s, ChannelParams *p, double in)
{
double proc = in, med;

proc = bprocess(proc, p->hp, &p->hw[0][0], &p->hw[0][1]);
proc = bprocess(proc, p->hp, &p->hw[1][0], &p->hw[1][1]);

if (proc >= 0.0) {
med = (D(p->ap + proc * (p->kpa - proc)) + p->kpb) * p->pwrq;
} else {
med = (D(p->an - proc * (p->kna + proc)) + p->knb) * p->pwrq * -1.0;
}

proc = p->srct * (med - p->prev_med + p->prev_out);
p->prev_med = M(med);
p->prev_out = M(proc);

proc = bprocess(proc, p->hp, &p->hw[2][0], &p->hw[2][1]);
proc = bprocess(proc, p->hp, &p->hw[3][0], &p->hw[3][1]);

if (s->ceil >= 10000.) {
proc = bprocess(proc, p->lp, &p->lw[0][0], &p->lw[0][1]);
proc = bprocess(proc, p->lp, &p->lw[1][0], &p->lw[1][1]);
}

return proc;
}

static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AExciterContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *out;
const double *src = (const double *)in->data[0];
const double level_in = s->level_in;
const double level_out = s->level_out;
const double amount = s->amount;
const double listen = 1.0 - s->listen;
double *dst;

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 (int n = 0; n < in->nb_samples; n++) {
for (int c = 0; c < inlink->channels; c++) {
double sample = src[c] * level_in;

sample = distortion_process(s, &s->cp[c], sample);
sample = sample * amount + listen * src[c];

sample *= level_out;
if (ctx->is_disabled)
dst[c] = src[c];
else
dst[c] = sample;
}

src += inlink->channels;
dst += inlink->channels;
}

if (in != out)
av_frame_free(&in);

return ff_filter_frame(outlink, out);
}

static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats;
AVFilterChannelLayouts *layouts;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_DBL,
AV_SAMPLE_FMT_NONE
};
int ret;

layouts = ff_all_channel_counts();
if (!layouts)
return AVERROR(ENOMEM);
ret = ff_set_common_channel_layouts(ctx, layouts);
if (ret < 0)
return ret;

formats = ff_make_format_list(sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
ret = ff_set_common_formats(ctx, formats);
if (ret < 0)
return ret;

formats = ff_all_samplerates();
if (!formats)
return AVERROR(ENOMEM);
return ff_set_common_samplerates(ctx, formats);
}

static av_cold void uninit(AVFilterContext *ctx)
{
AExciterContext *s = ctx->priv;

av_freep(&s->cp);
}

static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AExciterContext *s = ctx->priv;

if (!s->cp)
s->cp = av_calloc(inlink->channels, sizeof(*s->cp));
if (!s->cp)
return AVERROR(ENOMEM);

for (int i = 0; i < inlink->channels; i++)
set_params(&s->cp[i], s->blend, s->drive, inlink->sample_rate,
s->freq, s->ceil);

return 0;
}

static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
char *res, int res_len, int flags)
{
AVFilterLink *inlink = ctx->inputs[0];
int ret;

ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
if (ret < 0)
return ret;

return config_input(inlink);
}

static const AVFilterPad avfilter_af_aexciter_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
.filter_frame = filter_frame,
},
{ NULL }
};

static const AVFilterPad avfilter_af_aexciter_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};

AVFilter ff_af_aexciter = {
.name = "aexciter",
.description = NULL_IF_CONFIG_SMALL("Enhance high frequency part of audio."),
.priv_size = sizeof(AExciterContext),
.priv_class = &aexciter_class,
.uninit = uninit,
.query_formats = query_formats,
.inputs = avfilter_af_aexciter_inputs,
.outputs = avfilter_af_aexciter_outputs,
.process_command = process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
};

+ 1
- 0
libavfilter/allfilters.c View File

@@ -39,6 +39,7 @@ extern AVFilter ff_af_aderivative;
extern AVFilter ff_af_aecho;
extern AVFilter ff_af_aemphasis;
extern AVFilter ff_af_aeval;
extern AVFilter ff_af_aexciter;
extern AVFilter ff_af_afade;
extern AVFilter ff_af_afftdn;
extern AVFilter ff_af_afftfilt;


+ 1
- 1
libavfilter/version.h View File

@@ -30,7 +30,7 @@
#include "libavutil/version.h"

#define LIBAVFILTER_VERSION_MAJOR 7
#define LIBAVFILTER_VERSION_MINOR 103
#define LIBAVFILTER_VERSION_MINOR 104
#define LIBAVFILTER_VERSION_MICRO 100




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