288 lines
11 KiB
C
288 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/ffmath.h"
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#include "libavutil/mem.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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enum DetectionModes {
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DET_UNSET = 0,
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DET_DISABLED,
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DET_OFF,
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DET_ON,
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DET_ADAPTIVE,
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NB_DMODES,
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};
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enum FilterModes {
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LISTEN = -1,
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CUT_BELOW,
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CUT_ABOVE,
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BOOST_BELOW,
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BOOST_ABOVE,
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NB_FMODES,
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};
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typedef struct ChannelContext {
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double fa_double[3], fm_double[3];
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double dstate_double[2];
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double fstate_double[2];
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double tstate_double[2];
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double lin_gain_double;
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double detect_double;
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double threshold_log_double;
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double new_threshold_log_double;
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double log_sum_double;
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double sum_double;
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float fa_float[3], fm_float[3];
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float dstate_float[2];
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float fstate_float[2];
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float tstate_float[2];
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float lin_gain_float;
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float detect_float;
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float threshold_log_float;
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float new_threshold_log_float;
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float log_sum_float;
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float sum_float;
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void *dqueue;
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void *queue;
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int position;
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int size;
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int front;
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int back;
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int detection;
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int init;
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} ChannelContext;
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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 threshold_log;
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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 dattack;
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double drelease;
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double dattack_coef;
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double drelease_coef;
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double gattack_coef;
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double grelease_coef;
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int mode;
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int detection;
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int tftype;
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int dftype;
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int precision;
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int format;
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int nb_channels;
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int (*filter_prepare)(AVFilterContext *ctx);
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int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
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double da_double[3], dm_double[3];
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float da_float[3], dm_float[3];
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ChannelContext *cc;
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} AudioDynamicEqualizerContext;
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static int query_formats(AVFilterContext *ctx)
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{
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AudioDynamicEqualizerContext *s = ctx->priv;
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static const enum AVSampleFormat sample_fmts[3][3] = {
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{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
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{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE },
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{ AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
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};
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int ret;
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if ((ret = ff_set_common_all_channel_counts(ctx)) < 0)
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return ret;
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if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts[s->precision])) < 0)
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return ret;
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return ff_set_common_all_samplerates(ctx);
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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 1.0 - exp(-1.0 / (0.001 * x * sr));
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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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#define DEPTH 32
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#include "adynamicequalizer_template.c"
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#undef DEPTH
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#define DEPTH 64
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#include "adynamicequalizer_template.c"
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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->format = inlink->format;
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s->cc = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->cc));
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if (!s->cc)
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return AVERROR(ENOMEM);
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s->nb_channels = inlink->ch_layout.nb_channels;
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switch (s->format) {
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case AV_SAMPLE_FMT_DBLP:
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s->filter_prepare = filter_prepare_double;
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s->filter_channels = filter_channels_double;
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break;
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case AV_SAMPLE_FMT_FLTP:
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s->filter_prepare = filter_prepare_float;
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s->filter_channels = filter_channels_float;
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break;
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}
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for (int ch = 0; ch < s->nb_channels; ch++) {
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ChannelContext *cc = &s->cc[ch];
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cc->queue = av_calloc(inlink->sample_rate, sizeof(double));
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cc->dqueue = av_calloc(inlink->sample_rate, sizeof(double));
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if (!cc->queue || !cc->dqueue)
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return AVERROR(ENOMEM);
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}
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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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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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td.in = in;
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td.out = out;
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s->filter_prepare(ctx);
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ff_filter_execute(ctx, s->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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for (int ch = 0; ch < s->nb_channels; ch++) {
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ChannelContext *cc = &s->cc[ch];
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av_freep(&cc->queue);
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av_freep(&cc->dqueue);
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}
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av_freep(&s->cc);
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}
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#define OFFSET(x) offsetof(AudioDynamicEqualizerContext, x)
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#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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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 detection attack duration", OFFSET(dattack), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 0.01, 2000, FLAGS },
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{ "release","set detection release duration",OFFSET(drelease), AV_OPT_TYPE_DOUBLE, {.dbl=200}, 0.01, 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, 1000, FLAGS },
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{ "range", "set max gain", OFFSET(range), AV_OPT_TYPE_DOUBLE, {.dbl=50}, 1, 2000, FLAGS },
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{ "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, LISTEN,NB_FMODES-1,FLAGS, .unit = "mode" },
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{ "listen", 0, 0, AV_OPT_TYPE_CONST, {.i64=LISTEN}, 0, 0, FLAGS, .unit = "mode" },
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{ "cutbelow", 0, 0, AV_OPT_TYPE_CONST, {.i64=CUT_BELOW},0, 0, FLAGS, .unit = "mode" },
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{ "cutabove", 0, 0, AV_OPT_TYPE_CONST, {.i64=CUT_ABOVE},0, 0, FLAGS, .unit = "mode" },
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{ "boostbelow", 0, 0, AV_OPT_TYPE_CONST, {.i64=BOOST_BELOW},0, 0, FLAGS, .unit = "mode" },
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{ "boostabove", 0, 0, AV_OPT_TYPE_CONST, {.i64=BOOST_ABOVE},0, 0, FLAGS, .unit = "mode" },
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{ "dftype", "set detection filter type",OFFSET(dftype), AV_OPT_TYPE_INT, {.i64=0}, 0, 3, FLAGS, .unit = "dftype" },
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{ "bandpass", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, .unit = "dftype" },
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{ "lowpass", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, .unit = "dftype" },
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{ "highpass", 0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, .unit = "dftype" },
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{ "peak", 0, 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, FLAGS, .unit = "dftype" },
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{ "tftype", "set target filter type", OFFSET(tftype), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, FLAGS, .unit = "tftype" },
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{ "bell", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, .unit = "tftype" },
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{ "lowshelf", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, .unit = "tftype" },
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{ "highshelf",0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, .unit = "tftype" },
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{ "auto", "set auto threshold", OFFSET(detection), AV_OPT_TYPE_INT, {.i64=DET_OFF},DET_DISABLED,NB_DMODES-1,FLAGS, .unit = "auto" },
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{ "disabled", 0, 0, AV_OPT_TYPE_CONST, {.i64=DET_DISABLED}, 0, 0, FLAGS, .unit = "auto" },
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{ "off", 0, 0, AV_OPT_TYPE_CONST, {.i64=DET_OFF}, 0, 0, FLAGS, .unit = "auto" },
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{ "on", 0, 0, AV_OPT_TYPE_CONST, {.i64=DET_ON}, 0, 0, FLAGS, .unit = "auto" },
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{ "adaptive", 0, 0, AV_OPT_TYPE_CONST, {.i64=DET_ADAPTIVE}, 0, 0, FLAGS, .unit = "auto" },
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{ "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, AF, .unit = "precision" },
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{ "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "precision" },
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{ "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "precision" },
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{ "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "precision" },
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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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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(ff_audio_default_filterpad),
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FILTER_QUERY_FUNC(query_formats),
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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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