mirror of
https://github.com/FFmpeg/FFmpeg.git
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288 lines
8.7 KiB
C
288 lines
8.7 KiB
C
/*
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* Copyright (c) 2019 The FFmpeg Project
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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 "formats.h"
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enum ASoftClipTypes {
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ASC_TANH,
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ASC_ATAN,
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ASC_CUBIC,
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ASC_EXP,
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ASC_ALG,
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ASC_QUINTIC,
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ASC_SIN,
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NB_TYPES,
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};
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typedef struct ASoftClipContext {
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const AVClass *class;
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int type;
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double param;
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void (*filter)(struct ASoftClipContext *s, void **dst, const void **src,
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int nb_samples, int channels);
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} ASoftClipContext;
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#define OFFSET(x) offsetof(ASoftClipContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption asoftclip_options[] = {
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{ "type", "set softclip type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=0}, 0, NB_TYPES-1, A, "types" },
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{ "tanh", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_TANH}, 0, 0, A, "types" },
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{ "atan", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ATAN}, 0, 0, A, "types" },
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{ "cubic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_CUBIC}, 0, 0, A, "types" },
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{ "exp", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_EXP}, 0, 0, A, "types" },
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{ "alg", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ALG}, 0, 0, A, "types" },
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{ "quintic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_QUINTIC},0, 0, A, "types" },
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{ "sin", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_SIN}, 0, 0, A, "types" },
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{ "param", "set softclip parameter", OFFSET(param), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01, 3, A },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(asoftclip);
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats = NULL;
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AVFilterChannelLayouts *layouts = NULL;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_NONE
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};
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int ret;
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formats = ff_make_format_list(sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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ret = ff_set_common_formats(ctx, formats);
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if (ret < 0)
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return ret;
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layouts = ff_all_channel_counts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ret = ff_set_common_channel_layouts(ctx, layouts);
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if (ret < 0)
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return ret;
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formats = ff_all_samplerates();
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return ff_set_common_samplerates(ctx, formats);
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}
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#define SQR(x) ((x) * (x))
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static void filter_flt(ASoftClipContext *s,
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void **dptr, const void **sptr,
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int nb_samples, int channels)
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{
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float param = s->param;
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for (int c = 0; c < channels; c++) {
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const float *src = sptr[c];
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float *dst = dptr[c];
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switch (s->type) {
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case ASC_TANH:
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = tanhf(src[n] * param);
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}
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break;
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case ASC_ATAN:
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for (int n = 0; n < nb_samples; n++)
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dst[n] = 2.f / M_PI * atanf(src[n] * param);
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break;
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case ASC_CUBIC:
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for (int n = 0; n < nb_samples; n++) {
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if (FFABS(src[n]) >= 1.5f)
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dst[n] = FFSIGN(src[n]);
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else
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dst[n] = src[n] - 0.1481f * powf(src[n], 3.f);
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}
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break;
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case ASC_EXP:
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for (int n = 0; n < nb_samples; n++)
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dst[n] = 2.f / (1.f + expf(-2.f * src[n])) - 1.;
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break;
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case ASC_ALG:
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for (int n = 0; n < nb_samples; n++)
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dst[n] = src[n] / (sqrtf(param + src[n] * src[n]));
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break;
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case ASC_QUINTIC:
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for (int n = 0; n < nb_samples; n++) {
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if (FFABS(src[n]) >= 1.25)
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dst[n] = FFSIGN(src[n]);
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else
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dst[n] = src[n] - 0.08192f * powf(src[n], 5.f);
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}
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break;
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case ASC_SIN:
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for (int n = 0; n < nb_samples; n++) {
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if (FFABS(src[n]) >= M_PI_2)
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dst[n] = FFSIGN(src[n]);
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else
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dst[n] = sinf(src[n]);
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}
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break;
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}
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}
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}
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static void filter_dbl(ASoftClipContext *s,
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void **dptr, const void **sptr,
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int nb_samples, int channels)
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{
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double param = s->param;
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for (int c = 0; c < channels; c++) {
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const double *src = sptr[c];
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double *dst = dptr[c];
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switch (s->type) {
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case ASC_TANH:
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = tanh(src[n] * param);
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}
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break;
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case ASC_ATAN:
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for (int n = 0; n < nb_samples; n++)
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dst[n] = 2. / M_PI * atan(src[n] * param);
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break;
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case ASC_CUBIC:
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for (int n = 0; n < nb_samples; n++) {
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if (FFABS(src[n]) >= 1.5)
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dst[n] = FFSIGN(src[n]);
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else
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dst[n] = src[n] - 0.1481 * pow(src[n], 3.);
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}
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break;
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case ASC_EXP:
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for (int n = 0; n < nb_samples; n++)
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dst[n] = 2. / (1. + exp(-2. * src[n])) - 1.;
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break;
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case ASC_ALG:
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for (int n = 0; n < nb_samples; n++)
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dst[n] = src[n] / (sqrt(param + src[n] * src[n]));
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break;
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case ASC_QUINTIC:
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for (int n = 0; n < nb_samples; n++) {
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if (FFABS(src[n]) >= 1.25)
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dst[n] = FFSIGN(src[n]);
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else
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dst[n] = src[n] - 0.08192 * pow(src[n], 5.);
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}
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break;
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case ASC_SIN:
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for (int n = 0; n < nb_samples; n++) {
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if (FFABS(src[n]) >= M_PI_2)
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dst[n] = FFSIGN(src[n]);
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else
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dst[n] = sin(src[n]);
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}
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break;
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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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ASoftClipContext *s = ctx->priv;
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switch (inlink->format) {
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case AV_SAMPLE_FMT_FLT:
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case AV_SAMPLE_FMT_FLTP: s->filter = filter_flt; break;
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case AV_SAMPLE_FMT_DBL:
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case AV_SAMPLE_FMT_DBLP: s->filter = filter_dbl; 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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ASoftClipContext *s = ctx->priv;
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int nb_samples, channels;
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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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if (av_sample_fmt_is_planar(in->format)) {
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nb_samples = in->nb_samples;
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channels = in->channels;
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} else {
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nb_samples = in->channels * in->nb_samples;
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channels = 1;
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}
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s->filter(s, (void **)out->extended_data, (const void **)in->extended_data,
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nb_samples, channels);
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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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{ NULL }
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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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{ NULL }
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};
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AVFilter ff_af_asoftclip = {
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.name = "asoftclip",
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.description = NULL_IF_CONFIG_SMALL("Audio Soft Clipper."),
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.query_formats = query_formats,
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.priv_size = sizeof(ASoftClipContext),
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.priv_class = &asoftclip_class,
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.inputs = inputs,
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.outputs = outputs,
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};
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