FFmpeg/libavfilter/af_asoftclip.c

/*
 * Copyright (c) 2019 The FFmpeg Project
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
#include "formats.h"

enum ASoftClipTypes {
    ASC_TANH,
    ASC_ATAN,
    ASC_CUBIC,
    ASC_EXP,
    ASC_ALG,
    ASC_QUINTIC,
    ASC_SIN,
    NB_TYPES,
};

typedef struct ASoftClipContext {
    const AVClass *class;

    int type;
    double param;

    void (*filter)(struct ASoftClipContext *s, void **dst, const void **src,
                   int nb_samples, int channels);
} ASoftClipContext;

#define OFFSET(x) offsetof(ASoftClipContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

static const AVOption asoftclip_options[] = {
    { "type", "set softclip type", OFFSET(type), AV_OPT_TYPE_INT,    {.i64=0},          0, NB_TYPES-1, A, "types" },
    { "tanh",                NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_TANH},   0,          0, A, "types" },
    { "atan",                NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_ATAN},   0,          0, A, "types" },
    { "cubic",               NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_CUBIC},  0,          0, A, "types" },
    { "exp",                 NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_EXP},    0,          0, A, "types" },
    { "alg",                 NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_ALG},    0,          0, A, "types" },
    { "quintic",             NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_QUINTIC},0,          0, A, "types" },
    { "sin",                 NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_SIN},    0,          0, A, "types" },
    { "param", "set softclip parameter", OFFSET(param), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01,        3, A },
    { NULL }
};

AVFILTER_DEFINE_CLASS(asoftclip);

static int query_formats(AVFilterContext *ctx)
{
    AVFilterFormats *formats = NULL;
    AVFilterChannelLayouts *layouts = NULL;
    static const enum AVSampleFormat sample_fmts[] = {
        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
        AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
        AV_SAMPLE_FMT_NONE
    };
    int ret;

    formats = ff_make_format_list(sample_fmts);
    if (!formats)
        return AVERROR(ENOMEM);
    ret = ff_set_common_formats(ctx, formats);
    if (ret < 0)
        return ret;

    layouts = ff_all_channel_counts();
    if (!layouts)
        return AVERROR(ENOMEM);

    ret = ff_set_common_channel_layouts(ctx, layouts);
    if (ret < 0)
        return ret;

    formats = ff_all_samplerates();
    return ff_set_common_samplerates(ctx, formats);
}

#define SQR(x) ((x) * (x))

static void filter_flt(ASoftClipContext *s,
                       void **dptr, const void **sptr,
                       int nb_samples, int channels)
{
    float param = s->param;

    for (int c = 0; c < channels; c++) {
        const float *src = sptr[c];
        float *dst = dptr[c];

        switch (s->type) {
        case ASC_TANH:
            for (int n = 0; n < nb_samples; n++) {
                dst[n] = tanhf(src[n] * param);
            }
            break;
        case ASC_ATAN:
            for (int n = 0; n < nb_samples; n++)
                dst[n] = 2.f / M_PI * atanf(src[n] * param);
            break;
        case ASC_CUBIC:
            for (int n = 0; n < nb_samples; n++) {
                if (FFABS(src[n]) >= 1.5f)
                    dst[n] = FFSIGN(src[n]);
                else
                    dst[n] = src[n] - 0.1481f * powf(src[n], 3.f);
            }
            break;
        case ASC_EXP:
            for (int n = 0; n < nb_samples; n++)
                dst[n] = 2.f / (1.f + expf(-2.f * src[n])) - 1.;
            break;
        case ASC_ALG:
            for (int n = 0; n < nb_samples; n++)
                dst[n] = src[n] / (sqrtf(param + src[n] * src[n]));
            break;
        case ASC_QUINTIC:
            for (int n = 0; n < nb_samples; n++) {
                if (FFABS(src[n]) >= 1.25)
                    dst[n] = FFSIGN(src[n]);
                else
                    dst[n] = src[n] - 0.08192f * powf(src[n], 5.f);
            }
            break;
        case ASC_SIN:
            for (int n = 0; n < nb_samples; n++) {
                if (FFABS(src[n]) >= M_PI_2)
                    dst[n] = FFSIGN(src[n]);
                else
                    dst[n] = sinf(src[n]);
            }
            break;
        }
    }
}

static void filter_dbl(ASoftClipContext *s,
                       void **dptr, const void **sptr,
                       int nb_samples, int channels)
{
    double param = s->param;

    for (int c = 0; c < channels; c++) {
        const double *src = sptr[c];
        double *dst = dptr[c];

        switch (s->type) {
        case ASC_TANH:
            for (int n = 0; n < nb_samples; n++) {
                dst[n] = tanh(src[n] * param);
            }
            break;
        case ASC_ATAN:
            for (int n = 0; n < nb_samples; n++)
                dst[n] = 2. / M_PI * atan(src[n] * param);
            break;
        case ASC_CUBIC:
            for (int n = 0; n < nb_samples; n++) {
                if (FFABS(src[n]) >= 1.5)
                    dst[n] = FFSIGN(src[n]);
                else
                    dst[n] = src[n] - 0.1481 * pow(src[n], 3.);
            }
            break;
        case ASC_EXP:
            for (int n = 0; n < nb_samples; n++)
                dst[n] = 2. / (1. + exp(-2. * src[n])) - 1.;
            break;
        case ASC_ALG:
            for (int n = 0; n < nb_samples; n++)
                dst[n] = src[n] / (sqrt(param + src[n] * src[n]));
            break;
        case ASC_QUINTIC:
            for (int n = 0; n < nb_samples; n++) {
                if (FFABS(src[n]) >= 1.25)
                    dst[n] = FFSIGN(src[n]);
                else
                    dst[n] = src[n] - 0.08192 * pow(src[n], 5.);
            }
            break;
        case ASC_SIN:
            for (int n = 0; n < nb_samples; n++) {
                if (FFABS(src[n]) >= M_PI_2)
                    dst[n] = FFSIGN(src[n]);
                else
                    dst[n] = sin(src[n]);
            }
            break;
        }
    }
}

static int config_input(AVFilterLink *inlink)
{
    AVFilterContext *ctx = inlink->dst;
    ASoftClipContext *s = ctx->priv;

    switch (inlink->format) {
    case AV_SAMPLE_FMT_FLT:
    case AV_SAMPLE_FMT_FLTP: s->filter = filter_flt; break;
    case AV_SAMPLE_FMT_DBL:
    case AV_SAMPLE_FMT_DBLP: s->filter = filter_dbl; break;
    }

    return 0;
}

static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
    AVFilterContext *ctx = inlink->dst;
    AVFilterLink *outlink = ctx->outputs[0];
    ASoftClipContext *s = ctx->priv;
    int nb_samples, channels;
    AVFrame *out;

    if (av_frame_is_writable(in)) {
        out = in;
    } else {
        out = ff_get_audio_buffer(outlink, in->nb_samples);
        if (!out) {
            av_frame_free(&in);
            return AVERROR(ENOMEM);
        }
        av_frame_copy_props(out, in);
    }

    if (av_sample_fmt_is_planar(in->format)) {
        nb_samples = in->nb_samples;
        channels = in->channels;
    } else {
        nb_samples = in->channels * in->nb_samples;
        channels = 1;
    }

    s->filter(s, (void **)out->extended_data, (const void **)in->extended_data,
              nb_samples, channels);

    if (out != in)
        av_frame_free(&in);

    return ff_filter_frame(outlink, out);
}

static const AVFilterPad inputs[] = {
    {
        .name         = "default",
        .type         = AVMEDIA_TYPE_AUDIO,
        .filter_frame = filter_frame,
        .config_props = config_input,
    },
    { NULL }
};

static const AVFilterPad outputs[] = {
    {
        .name = "default",
        .type = AVMEDIA_TYPE_AUDIO,
    },
    { NULL }
};

AVFilter ff_af_asoftclip = {
    .name           = "asoftclip",
    .description    = NULL_IF_CONFIG_SMALL("Audio Soft Clipper."),
    .query_formats  = query_formats,
    .priv_size      = sizeof(ASoftClipContext),
    .priv_class     = &asoftclip_class,
    .inputs         = inputs,
    .outputs        = outputs,
    .flags          = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
};