2 * Copyright (c) 2019 The FFmpeg Project
4 * This file is part of FFmpeg.
6 * FFmpeg is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
9 * version 2.1 of the License, or (at your option) any later version.
11 * FFmpeg is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with FFmpeg; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 #include "libavutil/channel_layout.h"
22 #include "libavutil/opt.h"
38 typedef struct ASoftClipContext {
44 void (*filter)(struct ASoftClipContext *s, void **dst, const void **src,
45 int nb_samples, int channels);
48 #define OFFSET(x) offsetof(ASoftClipContext, x)
49 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
51 static const AVOption asoftclip_options[] = {
52 { "type", "set softclip type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=0}, 0, NB_TYPES-1, A, "types" },
53 { "tanh", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_TANH}, 0, 0, A, "types" },
54 { "atan", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ATAN}, 0, 0, A, "types" },
55 { "cubic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_CUBIC}, 0, 0, A, "types" },
56 { "exp", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_EXP}, 0, 0, A, "types" },
57 { "alg", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ALG}, 0, 0, A, "types" },
58 { "quintic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_QUINTIC},0, 0, A, "types" },
59 { "sin", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_SIN}, 0, 0, A, "types" },
60 { "param", "set softclip parameter", OFFSET(param), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01, 3, A },
64 AVFILTER_DEFINE_CLASS(asoftclip);
66 static int query_formats(AVFilterContext *ctx)
68 AVFilterFormats *formats = NULL;
69 AVFilterChannelLayouts *layouts = NULL;
70 static const enum AVSampleFormat sample_fmts[] = {
71 AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
72 AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
77 formats = ff_make_format_list(sample_fmts);
79 return AVERROR(ENOMEM);
80 ret = ff_set_common_formats(ctx, formats);
84 layouts = ff_all_channel_counts();
86 return AVERROR(ENOMEM);
88 ret = ff_set_common_channel_layouts(ctx, layouts);
92 formats = ff_all_samplerates();
93 return ff_set_common_samplerates(ctx, formats);
96 #define SQR(x) ((x) * (x))
98 static void filter_flt(ASoftClipContext *s,
99 void **dptr, const void **sptr,
100 int nb_samples, int channels)
102 float param = s->param;
104 for (int c = 0; c < channels; c++) {
105 const float *src = sptr[c];
106 float *dst = dptr[c];
110 for (int n = 0; n < nb_samples; n++) {
111 dst[n] = tanhf(src[n] * param);
115 for (int n = 0; n < nb_samples; n++)
116 dst[n] = 2.f / M_PI * atanf(src[n] * param);
119 for (int n = 0; n < nb_samples; n++) {
120 if (FFABS(src[n]) >= 1.5f)
121 dst[n] = FFSIGN(src[n]);
123 dst[n] = src[n] - 0.1481f * powf(src[n], 3.f);
127 for (int n = 0; n < nb_samples; n++)
128 dst[n] = 2.f / (1.f + expf(-2.f * src[n])) - 1.;
131 for (int n = 0; n < nb_samples; n++)
132 dst[n] = src[n] / (sqrtf(param + src[n] * src[n]));
135 for (int n = 0; n < nb_samples; n++) {
136 if (FFABS(src[n]) >= 1.25)
137 dst[n] = FFSIGN(src[n]);
139 dst[n] = src[n] - 0.08192f * powf(src[n], 5.f);
143 for (int n = 0; n < nb_samples; n++) {
144 if (FFABS(src[n]) >= M_PI_2)
145 dst[n] = FFSIGN(src[n]);
147 dst[n] = sinf(src[n]);
154 static void filter_dbl(ASoftClipContext *s,
155 void **dptr, const void **sptr,
156 int nb_samples, int channels)
158 double param = s->param;
160 for (int c = 0; c < channels; c++) {
161 const double *src = sptr[c];
162 double *dst = dptr[c];
166 for (int n = 0; n < nb_samples; n++) {
167 dst[n] = tanh(src[n] * param);
171 for (int n = 0; n < nb_samples; n++)
172 dst[n] = 2. / M_PI * atan(src[n] * param);
175 for (int n = 0; n < nb_samples; n++) {
176 if (FFABS(src[n]) >= 1.5)
177 dst[n] = FFSIGN(src[n]);
179 dst[n] = src[n] - 0.1481 * pow(src[n], 3.);
183 for (int n = 0; n < nb_samples; n++)
184 dst[n] = 2. / (1. + exp(-2. * src[n])) - 1.;
187 for (int n = 0; n < nb_samples; n++)
188 dst[n] = src[n] / (sqrt(param + src[n] * src[n]));
191 for (int n = 0; n < nb_samples; n++) {
192 if (FFABS(src[n]) >= 1.25)
193 dst[n] = FFSIGN(src[n]);
195 dst[n] = src[n] - 0.08192 * pow(src[n], 5.);
199 for (int n = 0; n < nb_samples; n++) {
200 if (FFABS(src[n]) >= M_PI_2)
201 dst[n] = FFSIGN(src[n]);
203 dst[n] = sin(src[n]);
210 static int config_input(AVFilterLink *inlink)
212 AVFilterContext *ctx = inlink->dst;
213 ASoftClipContext *s = ctx->priv;
215 switch (inlink->format) {
216 case AV_SAMPLE_FMT_FLT:
217 case AV_SAMPLE_FMT_FLTP: s->filter = filter_flt; break;
218 case AV_SAMPLE_FMT_DBL:
219 case AV_SAMPLE_FMT_DBLP: s->filter = filter_dbl; break;
225 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
227 AVFilterContext *ctx = inlink->dst;
228 AVFilterLink *outlink = ctx->outputs[0];
229 ASoftClipContext *s = ctx->priv;
230 int nb_samples, channels;
233 if (av_frame_is_writable(in)) {
236 out = ff_get_audio_buffer(outlink, in->nb_samples);
239 return AVERROR(ENOMEM);
241 av_frame_copy_props(out, in);
244 if (av_sample_fmt_is_planar(in->format)) {
245 nb_samples = in->nb_samples;
246 channels = in->channels;
248 nb_samples = in->channels * in->nb_samples;
252 s->filter(s, (void **)out->extended_data, (const void **)in->extended_data,
253 nb_samples, channels);
258 return ff_filter_frame(outlink, out);
261 static const AVFilterPad inputs[] = {
264 .type = AVMEDIA_TYPE_AUDIO,
265 .filter_frame = filter_frame,
266 .config_props = config_input,
271 static const AVFilterPad outputs[] = {
274 .type = AVMEDIA_TYPE_AUDIO,
279 AVFilter ff_af_asoftclip = {
281 .description = NULL_IF_CONFIG_SMALL("Audio Soft Clipper."),
282 .query_formats = query_formats,
283 .priv_size = sizeof(ASoftClipContext),
284 .priv_class = &asoftclip_class,