avfilter: add asupercut filter

pull/358/head
Paul B Mahol 4 years ago
parent 68e452c367
commit 3c922681c3
  1. 1
      Changelog
  2. 21
      doc/filters.texi
  3. 1
      libavfilter/Makefile
  4. 247
      libavfilter/af_asupercut.c
  5. 1
      libavfilter/allfilters.c
  6. 2
      libavfilter/version.h

@ -47,6 +47,7 @@ version <next>:
- DXVA2/D3D11VA hardware accelerated AV1 decoding - DXVA2/D3D11VA hardware accelerated AV1 decoding
- speechnorm filter - speechnorm filter
- SpeedHQ encoder - SpeedHQ encoder
- asupercut filter
version 4.3: version 4.3:

@ -1838,7 +1838,7 @@ Set central frequency for band.
If input doesn't have that frequency the entry is ignored. If input doesn't have that frequency the entry is ignored.
@item w @item w
Set band width in hertz. Set band width in Hertz.
@item g @item g
Set band gain in dB. Set band gain in dB.
@ -1903,7 +1903,7 @@ Syntax for the commands is : "@var{fN}|f=@var{freq}|w=@var{width}|g=@var{gain}"
@var{fN} is existing filter number, starting from 0, if no such filter is available @var{fN} is existing filter number, starting from 0, if no such filter is available
error is returned. error is returned.
@var{freq} set new frequency parameter. @var{freq} set new frequency parameter.
@var{width} set new width parameter in herz. @var{width} set new width parameter in Hertz.
@var{gain} set new gain parameter in dB. @var{gain} set new gain parameter in dB.
Full filter invocation with asendcmd may look like this: Full filter invocation with asendcmd may look like this:
@ -2584,7 +2584,7 @@ Set delay line feedback gain value. Allowed range is from 0 to 1.
Default value is 0.5. Default value is 0.5.
@item cutoff @item cutoff
Set cutoff frequency in herz. Allowed range is 50 to 900. Set cutoff frequency in Hertz. Allowed range is 50 to 900.
Default value is 100. Default value is 100.
@item slope @item slope
@ -2600,6 +2600,21 @@ Default value is 20.
This filter supports the all above options as @ref{commands}. This filter supports the all above options as @ref{commands}.
@section asupercut
Cut super frequencies.
The filter accepts the following options:
@table @option
@item cutoff
Set cutoff frequency in Hertz. Allowed range is 20000 to 192000.
Default value is 20000.
@end table
@subsection Commands
This filter supports the all above options as @ref{commands}.
@section atempo @section atempo
Adjust audio tempo. Adjust audio tempo.

@ -90,6 +90,7 @@ OBJS-$(CONFIG_ASR_FILTER) += af_asr.o
OBJS-$(CONFIG_ASTATS_FILTER) += af_astats.o OBJS-$(CONFIG_ASTATS_FILTER) += af_astats.o
OBJS-$(CONFIG_ASTREAMSELECT_FILTER) += f_streamselect.o framesync.o OBJS-$(CONFIG_ASTREAMSELECT_FILTER) += f_streamselect.o framesync.o
OBJS-$(CONFIG_ASUBBOOST_FILTER) += af_asubboost.o OBJS-$(CONFIG_ASUBBOOST_FILTER) += af_asubboost.o
OBJS-$(CONFIG_ASUPERCUT_FILTER) += af_asupercut.o
OBJS-$(CONFIG_ATEMPO_FILTER) += af_atempo.o OBJS-$(CONFIG_ATEMPO_FILTER) += af_atempo.o
OBJS-$(CONFIG_ATRIM_FILTER) += trim.o OBJS-$(CONFIG_ATRIM_FILTER) += trim.o
OBJS-$(CONFIG_AXCORRELATE_FILTER) += af_axcorrelate.o OBJS-$(CONFIG_AXCORRELATE_FILTER) += af_axcorrelate.o

@ -0,0 +1,247 @@
/*
* 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"
#include "formats.h"
typedef struct BiquadCoeffs {
double a1, a2;
double b0, b1, b2;
} BiquadCoeffs;
typedef struct ASuperCutContext {
const AVClass *class;
double cutoff;
int bypass;
BiquadCoeffs coeffs[5];
AVFrame *w;
} ASuperCutContext;
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layouts = NULL;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
int 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;
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_all_samplerates();
return ff_set_common_samplerates(ctx, formats);
}
static int get_coeffs(AVFilterContext *ctx)
{
ASuperCutContext *s = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
double w0 = s->cutoff / inlink->sample_rate;
double K = tan(M_PI * w0);
double q[5];
s->bypass = w0 >= 0.5;
if (s->bypass)
return 0;
q[0] = 0.50623256;
q[1] = 0.56116312;
q[2] = 0.70710678;
q[3] = 1.10134463;
q[4] = 3.19622661;
for (int b = 0; b < 5; b++) {
BiquadCoeffs *coeffs = &s->coeffs[b];
double norm = 1.0 / (1.0 + K / q[b] + K * K);
coeffs->b0 = K * K * norm;
coeffs->b1 = 2.0 * coeffs->b0;
coeffs->b2 = coeffs->b0;
coeffs->a1 = -2.0 * (K * K - 1.0) * norm;
coeffs->a2 = -(1.0 - K / q[b] + K * K) * norm;
}
return 0;
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
ASuperCutContext *s = ctx->priv;
s->w = ff_get_audio_buffer(inlink, 2 * 5);
if (!s->w)
return AVERROR(ENOMEM);
return get_coeffs(ctx);
}
typedef struct ThreadData {
AVFrame *in, *out;
} ThreadData;
static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
ASuperCutContext *s = ctx->priv;
ThreadData *td = arg;
AVFrame *out = td->out;
AVFrame *in = td->in;
const int start = (in->channels * jobnr) / nb_jobs;
const int end = (in->channels * (jobnr+1)) / nb_jobs;
for (int ch = start; ch < end; ch++) {
const double *src = (const double *)in->extended_data[ch];
double *dst = (double *)out->extended_data[ch];
for (int b = 0; b < 5; b++) {
BiquadCoeffs *coeffs = &s->coeffs[b];
const double a1 = coeffs->a1;
const double a2 = coeffs->a2;
const double b0 = coeffs->b0;
const double b1 = coeffs->b1;
const double b2 = coeffs->b2;
double *w = ((double *)s->w->extended_data[ch]) + b * 2;
for (int n = 0; n < in->nb_samples; n++) {
double sin = b ? dst[n] : src[n];
double sout = sin * b0 + w[0];
w[0] = b1 * sin + w[1] + a1 * sout;
w[1] = b2 * sin + a2 * sout;
dst[n] = sout;
}
}
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
ASuperCutContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
ThreadData td;
AVFrame *out;
if (s->bypass)
return ff_filter_frame(outlink, in);
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;
ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(inlink->channels,
ff_filter_get_nb_threads(ctx)));
if (out != in)
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
char *res, int res_len, int flags)
{
int ret;
ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
if (ret < 0)
return ret;
return get_coeffs(ctx);
}
static av_cold void uninit(AVFilterContext *ctx)
{
ASuperCutContext *s = ctx->priv;
av_frame_free(&s->w);
}
#define OFFSET(x) offsetof(ASuperCutContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption asupercut_options[] = {
{ "cutoff", "set cutoff frequency", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, {.dbl=20000}, 20000, 192000, FLAGS },
{ NULL }
};
AVFILTER_DEFINE_CLASS(asupercut);
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
},
{ NULL }
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};
AVFilter ff_af_asupercut = {
.name = "asupercut",
.description = NULL_IF_CONFIG_SMALL("Cut super frequencies."),
.query_formats = query_formats,
.priv_size = sizeof(ASuperCutContext),
.priv_class = &asupercut_class,
.uninit = uninit,
.inputs = inputs,
.outputs = outputs,
.process_command = process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
AVFILTER_FLAG_SLICE_THREADS,
};

@ -83,6 +83,7 @@ extern AVFilter ff_af_asr;
extern AVFilter ff_af_astats; extern AVFilter ff_af_astats;
extern AVFilter ff_af_astreamselect; extern AVFilter ff_af_astreamselect;
extern AVFilter ff_af_asubboost; extern AVFilter ff_af_asubboost;
extern AVFilter ff_af_asupercut;
extern AVFilter ff_af_atempo; extern AVFilter ff_af_atempo;
extern AVFilter ff_af_atrim; extern AVFilter ff_af_atrim;
extern AVFilter ff_af_axcorrelate; extern AVFilter ff_af_axcorrelate;

@ -30,7 +30,7 @@
#include "libavutil/version.h" #include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 7 #define LIBAVFILTER_VERSION_MAJOR 7
#define LIBAVFILTER_VERSION_MINOR 90 #define LIBAVFILTER_VERSION_MINOR 91
#define LIBAVFILTER_VERSION_MICRO 100 #define LIBAVFILTER_VERSION_MICRO 100

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