/* * 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/ffmath.h"#include "libavutil/opt.h"#include "avfilter.h"#include "audio.h"#include "filters.h" typedef struct AudioDynamicSmoothContext {    const AVClass *class;     double sensitivity;    double basefreq;     AVFrame *coeffs;} AudioDynamicSmoothContext; static int config_input(AVFilterLink *inlink){    AVFilterContext *ctx = inlink->dst;    AudioDynamicSmoothContext *s = ctx->priv;     s->coeffs = ff_get_audio_buffer(inlink, 3);    if (!s->coeffs)        return AVERROR(ENOMEM);     return 0;} static int filter_frame(AVFilterLink *inlink, AVFrame *in){    AVFilterContext *ctx = inlink->dst;    AVFilterLink *outlink = ctx->outputs[0];    AudioDynamicSmoothContext *s = ctx->priv;    const double sensitivity = s->sensitivity;    const double wc = s->basefreq / in->sample_rate;    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 < out->ch_layout.nb_channels; ch++) {        const double *src = (const double *)in->extended_data[ch];        double *dst = (double *)out->extended_data[ch];        double *coeffs = (double *)s->coeffs->extended_data[ch];        double low1 = coeffs[0];        double low2 = coeffs[1];        double inz  = coeffs[2];         for (int n = 0; n < out->nb_samples; n++) {            double low1z = low1;            double low2z = low2;            double bandz = low2z - low1z;            double wd = wc + sensitivity * fabs(bandz);            double g = fmin(1., wd * (5.9948827 + wd * (-11.969296 + wd * 15.959062)));             low1 = low1z + g * (0.5 * (src[n] + inz)   - low1z);            low2 = low2z + g * (0.5 * (low1   + low1z) - low2z);            inz = src[n];            dst[n] = ctx->is_disabled ? src[n] : low2;        }         coeffs[0] = low1;        coeffs[1] = low2;        coeffs[2] = inz;    }     if (out != in)        av_frame_free(&in);    return ff_filter_frame(outlink, out);} static av_cold void uninit(AVFilterContext *ctx){    AudioDynamicSmoothContext *s = ctx->priv;     av_frame_free(&s->coeffs);} #define OFFSET(x) offsetof(AudioDynamicSmoothContext, x)#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM static const AVOption adynamicsmooth_options[] = {    { "sensitivity",  "set smooth sensitivity",  OFFSET(sensitivity),  AV_OPT_TYPE_DOUBLE, {.dbl=2},     0, 1000000, FLAGS },    { "basefreq",     "set base frequency",      OFFSET(basefreq),     AV_OPT_TYPE_DOUBLE, {.dbl=22050}, 2, 1000000, FLAGS },    { NULL }}; AVFILTER_DEFINE_CLASS(adynamicsmooth); static const AVFilterPad inputs[] = {    {        .name         = "default",        .type         = AVMEDIA_TYPE_AUDIO,        .filter_frame = filter_frame,        .config_props = config_input,    },}; const FFFilter ff_af_adynamicsmooth = {    .p.name          = "adynamicsmooth",    .p.description   = NULL_IF_CONFIG_SMALL("Apply Dynamic Smoothing of input audio."),    .p.priv_class    = &adynamicsmooth_class,    .p.flags         = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,    .priv_size       = sizeof(AudioDynamicSmoothContext),    .uninit          = uninit,    FILTER_INPUTS(inputs),    FILTER_OUTPUTS(ff_audio_default_filterpad),    FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),    .process_command = ff_filter_process_command,};