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https://github.com/FFmpeg/FFmpeg.git
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1afe42852b
internal.h currently mixes interfaces intended to be used by filters with those that should be limited to generic filter- or graph-level code.
182 lines
4.9 KiB
C
182 lines
4.9 KiB
C
/*
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* Copyright (c) 2008 Rob Sykes
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* Copyright (c) 2017 Paul B Mahol
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*
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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/channel_layout.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 "filters.h"
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typedef struct AudioContrastContext {
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const AVClass *class;
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float contrast;
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void (*filter)(void **dst, const void **src,
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int nb_samples, int channels, float contrast);
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} AudioContrastContext;
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#define OFFSET(x) offsetof(AudioContrastContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption acontrast_options[] = {
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{ "contrast", "set contrast", OFFSET(contrast), AV_OPT_TYPE_FLOAT, {.dbl=33}, 0, 100, A },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(acontrast);
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static void filter_flt(void **d, const void **s,
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int nb_samples, int channels,
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float contrast)
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{
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const float *src = s[0];
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float *dst = d[0];
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int n, c;
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for (n = 0; n < nb_samples; n++) {
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for (c = 0; c < channels; c++) {
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float d = src[c] * M_PI_2;
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dst[c] = sinf(d + contrast * sinf(d * 4));
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}
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dst += c;
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src += c;
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}
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}
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static void filter_dbl(void **d, const void **s,
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int nb_samples, int channels,
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float contrast)
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{
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const double *src = s[0];
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double *dst = d[0];
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int n, c;
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for (n = 0; n < nb_samples; n++) {
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for (c = 0; c < channels; c++) {
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double d = src[c] * M_PI_2;
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dst[c] = sin(d + contrast * sin(d * 4));
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}
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dst += c;
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src += c;
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}
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}
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static void filter_fltp(void **d, const void **s,
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int nb_samples, int channels,
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float contrast)
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{
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int n, c;
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for (c = 0; c < channels; c++) {
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const float *src = s[c];
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float *dst = d[c];
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for (n = 0; n < nb_samples; n++) {
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float d = src[n] * M_PI_2;
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dst[n] = sinf(d + contrast * sinf(d * 4));
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}
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}
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}
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static void filter_dblp(void **d, const void **s,
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int nb_samples, int channels,
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float contrast)
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{
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int n, c;
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for (c = 0; c < channels; c++) {
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const double *src = s[c];
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double *dst = d[c];
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for (n = 0; n < nb_samples; n++) {
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double d = src[n] * M_PI_2;
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dst[n] = sin(d + contrast * sin(d * 4));
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}
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}
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}
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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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AudioContrastContext *s = ctx->priv;
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switch (inlink->format) {
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case AV_SAMPLE_FMT_FLT: s->filter = filter_flt; break;
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case AV_SAMPLE_FMT_DBL: s->filter = filter_dbl; break;
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case AV_SAMPLE_FMT_FLTP: s->filter = filter_fltp; break;
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case AV_SAMPLE_FMT_DBLP: s->filter = filter_dblp; break;
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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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AudioContrastContext *s = ctx->priv;
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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->filter((void **)out->extended_data, (const void **)in->extended_data,
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in->nb_samples, in->ch_layout.nb_channels, s->contrast / 750);
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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 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_acontrast = {
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.name = "acontrast",
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.description = NULL_IF_CONFIG_SMALL("Simple audio dynamic range compression/expansion filter."),
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.priv_size = sizeof(AudioContrastContext),
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.priv_class = &acontrast_class,
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FILTER_INPUTS(inputs),
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FILTER_OUTPUTS(ff_audio_default_filterpad),
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FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP),
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};
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