307 lines
11 KiB
C
307 lines
11 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 <float.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 AudioDynamicEqualizerContext {
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const AVClass *class;
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double threshold;
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double dfrequency;
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double dqfactor;
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double tfrequency;
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double tqfactor;
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double ratio;
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double range;
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double makeup;
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double attack;
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double release;
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double attack_coef;
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double release_coef;
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int mode;
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int direction;
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int type;
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AVFrame *state;
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} AudioDynamicEqualizerContext;
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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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AudioDynamicEqualizerContext *s = ctx->priv;
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s->state = ff_get_audio_buffer(inlink, 8);
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if (!s->state)
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return AVERROR(ENOMEM);
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for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
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double *state = (double *)s->state->extended_data[ch];
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state[4] = 1.;
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}
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return 0;
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}
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static double get_svf(double in, double *m, double *a, double *b)
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{
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const double v0 = in;
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const double v3 = v0 - b[1];
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const double v1 = a[0] * b[0] + a[1] * v3;
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const double v2 = b[1] + a[1] * b[0] + a[2] * v3;
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b[0] = 2. * v1 - b[0];
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b[1] = 2. * v2 - b[1];
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return m[0] * v0 + m[1] * v1 + m[2] * v2;
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}
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typedef struct ThreadData {
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AVFrame *in, *out;
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} ThreadData;
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static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
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{
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AudioDynamicEqualizerContext *s = ctx->priv;
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ThreadData *td = arg;
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AVFrame *in = td->in;
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AVFrame *out = td->out;
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const double sample_rate = in->sample_rate;
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const double makeup = s->makeup;
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const double ratio = s->ratio;
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const double range = s->range;
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const double dfrequency = fmin(s->dfrequency, sample_rate * 0.5);
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const double tfrequency = fmin(s->tfrequency, sample_rate * 0.5);
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const double threshold = s->threshold;
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const double release = s->release_coef;
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const double irelease = 1. - release;
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const double attack = s->attack_coef;
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const double iattack = 1. - attack;
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const double dqfactor = s->dqfactor;
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const double tqfactor = s->tqfactor;
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const double fg = tan(M_PI * tfrequency / sample_rate);
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const double dg = tan(M_PI * dfrequency / sample_rate);
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const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
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const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
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const int direction = s->direction;
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const int mode = s->mode;
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const int type = s->type;
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double da[3], dm[3];
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{
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double k = 1. / dqfactor;
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da[0] = 1. / (1. + dg * (dg + k));
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da[1] = dg * da[0];
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da[2] = dg * da[1];
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dm[0] = 0.;
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dm[1] = k;
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dm[2] = 0.;
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}
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for (int ch = start; ch < end; 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 *state = (double *)s->state->extended_data[ch];
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for (int n = 0; n < out->nb_samples; n++) {
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double detect, gain, v, listen;
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double fa[3], fm[3];
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double k, g;
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detect = listen = get_svf(src[n], dm, da, state);
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detect = fabs(detect);
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if (direction == 0 && mode == 0 && detect < threshold)
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detect = 1. / av_clipd(1. + makeup + (threshold - detect) * ratio, 1., range);
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else if (direction == 0 && mode == 1 && detect < threshold)
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detect = av_clipd(1. + makeup + (threshold - detect) * ratio, 1., range);
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else if (direction == 1 && mode == 0 && detect > threshold)
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detect = 1. / av_clipd(1. + makeup + (detect - threshold) * ratio, 1., range);
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else if (direction == 1 && mode == 1 && detect > threshold)
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detect = av_clipd(1. + makeup + (detect - threshold) * ratio, 1., range);
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else
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detect = 1.;
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if (detect < state[4]) {
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detect = iattack * detect + attack * state[4];
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} else {
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detect = irelease * detect + release * state[4];
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}
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if (state[4] != detect || n == 0) {
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state[4] = gain = detect;
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switch (type) {
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case 0:
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k = 1. / (tqfactor * gain);
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fa[0] = 1. / (1. + fg * (fg + k));
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fa[1] = fg * fa[0];
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fa[2] = fg * fa[1];
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fm[0] = 1.;
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fm[1] = k * (gain * gain - 1.);
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fm[2] = 0.;
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break;
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case 1:
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k = 1. / tqfactor;
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g = fg / sqrt(gain);
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fa[0] = 1. / (1. + g * (g + k));
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fa[1] = g * fa[0];
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fa[2] = g * fa[1];
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fm[0] = 1.;
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fm[1] = k * (gain - 1.);
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fm[2] = gain * gain - 1.;
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break;
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case 2:
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k = 1. / tqfactor;
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g = fg / sqrt(gain);
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fa[0] = 1. / (1. + g * (g + k));
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fa[1] = g * fa[0];
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fa[2] = g * fa[1];
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fm[0] = gain * gain;
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fm[1] = k * (1. - gain) * gain;
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fm[2] = 1. - gain * gain;
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break;
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}
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}
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v = get_svf(src[n], fm, fa, &state[2]);
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v = mode == -1 ? listen : v;
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dst[n] = ctx->is_disabled ? src[n] : v;
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}
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}
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return 0;
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}
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static double get_coef(double x, double sr)
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{
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return exp(-1000. / (x * sr));
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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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AudioDynamicEqualizerContext *s = ctx->priv;
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ThreadData td;
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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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s->attack_coef = get_coef(s->attack, in->sample_rate);
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s->release_coef = get_coef(s->release, in->sample_rate);
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td.in = in;
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td.out = out;
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ff_filter_execute(ctx, filter_channels, &td, NULL,
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FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
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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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AudioDynamicEqualizerContext *s = ctx->priv;
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av_frame_free(&s->state);
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}
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#define OFFSET(x) offsetof(AudioDynamicEqualizerContext, 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 adynamicequalizer_options[] = {
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{ "threshold", "set detection threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 100, FLAGS },
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{ "dfrequency", "set detection frequency", OFFSET(dfrequency), AV_OPT_TYPE_DOUBLE, {.dbl=1000}, 2, 1000000, FLAGS },
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{ "dqfactor", "set detection Q factor", OFFSET(dqfactor), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.001, 1000, FLAGS },
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{ "tfrequency", "set target frequency", OFFSET(tfrequency), AV_OPT_TYPE_DOUBLE, {.dbl=1000}, 2, 1000000, FLAGS },
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{ "tqfactor", "set target Q factor", OFFSET(tqfactor), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.001, 1000, FLAGS },
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{ "attack", "set attack duration", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 1, 2000, FLAGS },
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{ "release", "set release duration", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=200}, 1, 2000, FLAGS },
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{ "ratio", "set ratio factor", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 30, FLAGS },
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{ "makeup", "set makeup gain", OFFSET(makeup), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 100, FLAGS },
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{ "range", "set max gain", OFFSET(range), AV_OPT_TYPE_DOUBLE, {.dbl=50}, 1, 200, FLAGS },
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{ "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, -1, 1, FLAGS, "mode" },
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{ "listen", 0, 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, FLAGS, "mode" },
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{ "cut", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "mode" },
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{ "boost", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "mode" },
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{ "tftype", "set target filter type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, FLAGS, "type" },
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{ "bell", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "type" },
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{ "lowshelf", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "type" },
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{ "highshelf",0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, "type" },
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{ "direction", "set direction", OFFSET(direction), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS, "direction" },
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{ "downward", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "direction" },
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{ "upward", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "direction" },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(adynamicequalizer);
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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_adynamicequalizer = {
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.name = "adynamicequalizer",
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.description = NULL_IF_CONFIG_SMALL("Apply Dynamic Equalization of input audio."),
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.priv_size = sizeof(AudioDynamicEqualizerContext),
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.priv_class = &adynamicequalizer_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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AVFILTER_FLAG_SLICE_THREADS,
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.process_command = ff_filter_process_command,
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};
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