mirror of
https://github.com/FFmpeg/FFmpeg.git
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143 lines
4.3 KiB
C
143 lines
4.3 KiB
C
/*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/ffmath.h"
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#include "libavutil/opt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "formats.h"
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typedef struct AudioDynamicSmoothContext {
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const AVClass *class;
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double sensitivity;
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double basefreq;
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AVFrame *coeffs;
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} AudioDynamicSmoothContext;
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioDynamicSmoothContext *s = ctx->priv;
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s->coeffs = ff_get_audio_buffer(inlink, 3);
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if (!s->coeffs)
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return AVERROR(ENOMEM);
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return 0;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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AVFilterContext *ctx = inlink->dst;
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AVFilterLink *outlink = ctx->outputs[0];
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AudioDynamicSmoothContext *s = ctx->priv;
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const double sensitivity = s->sensitivity;
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const double wc = s->basefreq / in->sample_rate;
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AVFrame *out;
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if (av_frame_is_writable(in)) {
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out = in;
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} else {
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out = ff_get_audio_buffer(outlink, in->nb_samples);
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if (!out) {
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av_frame_free(&in);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(out, in);
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}
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for (int ch = 0; ch < out->ch_layout.nb_channels; ch++) {
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const double *src = (const double *)in->extended_data[ch];
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double *dst = (double *)out->extended_data[ch];
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double *coeffs = (double *)s->coeffs->extended_data[ch];
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double low1 = coeffs[0];
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double low2 = coeffs[1];
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double inz = coeffs[2];
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for (int n = 0; n < out->nb_samples; n++) {
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double low1z = low1;
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double low2z = low2;
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double bandz = low2z - low1z;
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double wd = wc + sensitivity * fabs(bandz);
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double g = fmin(1., wd * (5.9948827 + wd * (-11.969296 + wd * 15.959062)));
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low1 = low1z + g * (0.5 * (src[n] + inz) - low1z);
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low2 = low2z + g * (0.5 * (low1 + low1z) - low2z);
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inz = src[n];
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dst[n] = ctx->is_disabled ? src[n] : low2;
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}
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coeffs[0] = low1;
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coeffs[1] = low2;
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coeffs[2] = inz;
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}
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if (out != in)
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av_frame_free(&in);
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return ff_filter_frame(outlink, out);
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AudioDynamicSmoothContext *s = ctx->priv;
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av_frame_free(&s->coeffs);
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}
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#define OFFSET(x) offsetof(AudioDynamicSmoothContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
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static const AVOption adynamicsmooth_options[] = {
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{ "sensitivity", "set smooth sensitivity", OFFSET(sensitivity), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 1000000, FLAGS },
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{ "basefreq", "set base frequency", OFFSET(basefreq), AV_OPT_TYPE_DOUBLE, {.dbl=22050}, 2, 1000000, FLAGS },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(adynamicsmooth);
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static const AVFilterPad inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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.config_props = config_input,
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},
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};
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static const AVFilterPad outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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};
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const AVFilter ff_af_adynamicsmooth = {
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.name = "adynamicsmooth",
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.description = NULL_IF_CONFIG_SMALL("Apply Dynamic Smoothing of input audio."),
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.priv_size = sizeof(AudioDynamicSmoothContext),
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.priv_class = &adynamicsmooth_class,
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.uninit = uninit,
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FILTER_INPUTS(inputs),
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FILTER_OUTPUTS(outputs),
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FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
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.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
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.process_command = ff_filter_process_command,
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};
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