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https://github.com/FFmpeg/FFmpeg.git
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790f793844
There are lots of files that don't need it: The number of object files that actually need it went down from 2011 to 884 here. Keep it for external users in order to not cause breakages. Also improve the other headers a bit while just at it. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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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