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/*
* 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 *w;
AVFrame *buffer;
} ASubBoostContext;
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->w = ff_get_audio_buffer(inlink, 2);
s->write_pos = av_calloc(inlink->channels, sizeof(*s->write_pos));
if (!s->buffer || !s->w || !s->write_pos)
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)
{
ASubBoostContext *s = ctx->priv;
ThreadData *td = arg;
AVFrame *out = td->out;
AVFrame *in = td->in;
const double mix = ctx->is_disabled ? 0. : 1.;
const double wet = ctx->is_disabled ? 1. : s->wet_gain;
const double dry = ctx->is_disabled ? 1. : s->dry_gain;
const double feedback = s->feedback, decay = s->decay;
const double b0 = s->b0;
const double b1 = s->b1;
const double b2 = s->b2;
const double a1 = -s->a1;
const double a2 = -s->a2;
const int start = (in->channels * jobnr) / nb_jobs;
const int end = (in->channels * (jobnr+1)) / nb_jobs;
const int buffer_samples = s->buffer_samples;
for (int ch = start; ch < end; 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 *w = (double *)s->w->extended_data[ch];
int write_pos = s->write_pos[ch];
for (int n = 0; n < in->nb_samples; n++) {
double out_sample;
out_sample = src[n] * b0 + w[0];
w[0] = b1 * src[n] + w[1] + a1 * out_sample;
w[1] = b2 * src[n] + a2 * out_sample;
buffer[write_pos] = buffer[write_pos] * decay + out_sample * feedback;
dst[n] = (src[n] * dry + buffer[write_pos] * mix) * wet;
if (++write_pos >= buffer_samples)
write_pos = 0;
}
s->write_pos[ch] = write_pos;
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
ThreadData td;
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);
}
td.in = in; td.out = out;
ff_filter_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 av_cold void uninit(AVFilterContext *ctx)
{
ASubBoostContext *s = ctx->priv;
av_frame_free(&s->buffer);
av_frame_free(&s->w);
av_freep(&s->write_pos);
}
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=0.7}, 0, 1, FLAGS },
{ "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.7}, 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=0.9}, 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,
},
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
};
const AVFilter ff_af_asubboost = {
.name = "asubboost",
.description = NULL_IF_CONFIG_SMALL("Boost subwoofer frequencies."),
.priv_size = sizeof(ASubBoostContext),
.priv_class = &asubboost_class,
.uninit = uninit,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
.process_command = process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
};
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