Commit c7d80823 authored by Paul B Mahol's avatar Paul B Mahol

avfilter: add asubboost filter

parent d817b57d
......@@ -63,6 +63,7 @@ version <next>:
- maskedthreshold filter
- Support for muxing pcm and pgs in m2ts
- Cunning Developments ADPCM decoder
- asubboost filter
version 4.2:
......
......@@ -2454,6 +2454,45 @@ Number of points where the waveform crosses the zero level axis.
Rate of Zero crossings and number of audio samples.
@end table
@section asubboost
Boost subwoofer frequencies.
The filter accepts the following options:
@table @option
@item dry
Set dry gain, how much of original signal is kept. Allowed range is from 0 to 1.
Default value is 0.5.
@item wet
Set wet gain, how much of filtered signal is kept. Allowed range is from 0 to 1.
Default value is 0.8.
@item decay
Set delay line decay gain value. Allowed range is from 0 to 1.
Default value is 0.7.
@item feedback
Set delay line feedback gain value. Allowed range is from 0 to 1.
Default value is 0.5.
@item cutoff
Set cutoff frequency in herz. Allowed range is 50 to 900.
Default value is 100.
@item slope
Set slope amount for cutoff frequency. Allowed range is 0.0001 to 1.
Default value is 0.5.
@item delay
Set delay. Allowed range is from 1 to 100.
Default value is 20.
@end table
@subsection Commands
This filter supports the all above options as @ref{commands}.
@section atempo
Adjust audio tempo.
......
......@@ -86,6 +86,7 @@ OBJS-$(CONFIG_ASPLIT_FILTER) += split.o
OBJS-$(CONFIG_ASR_FILTER) += af_asr.o
OBJS-$(CONFIG_ASTATS_FILTER) += af_astats.o
OBJS-$(CONFIG_ASTREAMSELECT_FILTER) += f_streamselect.o framesync.o
OBJS-$(CONFIG_ASUBBOOST_FILTER) += af_asubboost.o
OBJS-$(CONFIG_ATEMPO_FILTER) += af_atempo.o
OBJS-$(CONFIG_ATRIM_FILTER) += trim.o
OBJS-$(CONFIG_AXCORRELATE_FILTER) += af_axcorrelate.o
......
/*
* 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 ASubBoostContext {
const AVClass *class;
double dry_gain;
double wet_gain;
double feedback;
double decay;
double delay;
double cutoff;
double slope;
double a0, a1, a2;
double b0, b1, b2;
int write_pos;
int buffer_samples;
AVFrame *i, *o;
AVFrame *buffer;
} ASubBoostContext;
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)
{
ASubBoostContext *s = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
double w0 = 2 * M_PI * s->cutoff / inlink->sample_rate;
double alpha = sin(w0) / 2 * sqrt(2. * (1. / s->slope - 1.) + 2.);
s->a0 = 1 + alpha;
s->a1 = -2 * cos(w0);
s->a2 = 1 - alpha;
s->b0 = (1 - cos(w0)) / 2;
s->b1 = 1 - cos(w0);
s->b2 = (1 - cos(w0)) / 2;
s->a1 /= s->a0;
s->a2 /= s->a0;
s->b0 /= s->a0;
s->b1 /= s->a0;
s->b2 /= s->a0;
s->buffer_samples = inlink->sample_rate * s->delay / 1000;
return 0;
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
ASubBoostContext *s = ctx->priv;
s->buffer = ff_get_audio_buffer(inlink, inlink->sample_rate / 10);
s->i = ff_get_audio_buffer(inlink, 2);
s->o = ff_get_audio_buffer(inlink, 2);
if (!s->buffer || !s->i || !s->o)
return AVERROR(ENOMEM);
return get_coeffs(ctx);
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
ASubBoostContext *s = ctx->priv;
const float wet = s->wet_gain, dry = s->dry_gain, feedback = s->feedback, decay = s->decay;
int write_pos;
AVFrame *out;
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);
}
for (int ch = 0; ch < in->channels; ch++) {
const double *src = (const double *)in->extended_data[ch];
double *dst = (double *)out->extended_data[ch];
double *buffer = (double *)s->buffer->extended_data[ch];
double *ix = (double *)s->i->extended_data[ch];
double *ox = (double *)s->o->extended_data[ch];
write_pos = s->write_pos;
for (int n = 0; n < in->nb_samples; n++) {
double out_sample;
out_sample = src[n] * s->b0 + ix[0] * s->b1 + ix[1] * s->b2 - ox[0] * s->a1 - ox[1] * s->a2;
ix[1] = ix[0];
ix[0] = src[n];
ox[1] = ox[0];
ox[0] = out_sample;
buffer[write_pos] = buffer[write_pos] * decay + out_sample * feedback;
dst[n] = src[n] * dry + buffer[write_pos] * wet;
if (++write_pos >= s->buffer_samples)
write_pos = 0;
}
}
s->write_pos = write_pos;
if (out != in)
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static av_cold void uninit(AVFilterContext *ctx)
{
ASubBoostContext *s = ctx->priv;
av_frame_free(&s->buffer);
av_frame_free(&s->i);
av_frame_free(&s->o);
}
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);
}
#define OFFSET(x) offsetof(ASubBoostContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption asubboost_options[] = {
{ "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, FLAGS },
{ "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.8}, 0, 1, FLAGS },
{ "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=0.7}, 0, 1, FLAGS },
{ "feedback", "set feedback", OFFSET(feedback), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, FLAGS },
{ "cutoff", "set cutoff", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, {.dbl=100}, 50, 900, FLAGS },
{ "slope", "set slope", OFFSET(slope), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.0001, 1, FLAGS },
{ "delay", "set delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 1, 100, FLAGS },
{ NULL }
};
AVFILTER_DEFINE_CLASS(asubboost);
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_asubboost = {
.name = "asubboost",
.description = NULL_IF_CONFIG_SMALL("Boost subwoofer frequencies."),
.query_formats = query_formats,
.priv_size = sizeof(ASubBoostContext),
.priv_class = &asubboost_class,
.uninit = uninit,
.inputs = inputs,
.outputs = outputs,
.process_command = process_command,
};
......@@ -79,6 +79,7 @@ extern AVFilter ff_af_asplit;
extern AVFilter ff_af_asr;
extern AVFilter ff_af_astats;
extern AVFilter ff_af_astreamselect;
extern AVFilter ff_af_asubboost;
extern AVFilter ff_af_atempo;
extern AVFilter ff_af_atrim;
extern AVFilter ff_af_axcorrelate;
......
......@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 7
#define LIBAVFILTER_VERSION_MINOR 79
#define LIBAVFILTER_VERSION_MINOR 80
#define LIBAVFILTER_VERSION_MICRO 100
......
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