FFmpeg/libavfilter/af_adynamicequalizer.c

/*
 * 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 <float.h>

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

typedef struct AudioDynamicEqualizerContext {
    const AVClass *class;

    double threshold;
    double dfrequency;
    double dqfactor;
    double tfrequency;
    double tqfactor;
    double ratio;
    double range;
    double makeup;
    double attack;
    double release;
    double attack_coef;
    double release_coef;
    int mode;
    int direction;
    int detection;
    int type;

    AVFrame *state;
} AudioDynamicEqualizerContext;

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

    s->state = ff_get_audio_buffer(inlink, 8);
    if (!s->state)
        return AVERROR(ENOMEM);

    for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
        double *state = (double *)s->state->extended_data[ch];

        state[4] = 1.;
    }

    return 0;
}

static double get_svf(double in, double *m, double *a, double *b)
{
    const double v0 = in;
    const double v3 = v0 - b[1];
    const double v1 = a[0] * b[0] + a[1] * v3;
    const double v2 = b[1] + a[1] * b[0] + a[2] * v3;

    b[0] = 2. * v1 - b[0];
    b[1] = 2. * v2 - b[1];

    return m[0] * v0 + m[1] * v1 + m[2] * v2;
}

typedef struct ThreadData {
    AVFrame *in, *out;
} ThreadData;

static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
    AudioDynamicEqualizerContext *s = ctx->priv;
    ThreadData *td = arg;
    AVFrame *in = td->in;
    AVFrame *out = td->out;
    const double sample_rate = in->sample_rate;
    const double makeup = s->makeup;
    const double ratio = s->ratio;
    const double range = s->range;
    const double dfrequency = fmin(s->dfrequency, sample_rate * 0.5);
    const double tfrequency = fmin(s->tfrequency, sample_rate * 0.5);
    const double release = s->release_coef;
    const double irelease = 1. - release;
    const double attack = s->attack_coef;
    const double iattack = 1. - attack;
    const double dqfactor = s->dqfactor;
    const double tqfactor = s->tqfactor;
    const double fg = tan(M_PI * tfrequency / sample_rate);
    const double dg = tan(M_PI * dfrequency / sample_rate);
    const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
    const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
    const int detection = s->detection;
    const int direction = s->direction;
    const int mode = s->mode;
    const int type = s->type;
    double da[3], dm[3];

    {
        double k = 1. / dqfactor;

        da[0] = 1. / (1. + dg * (dg + k));
        da[1] = dg * da[0];
        da[2] = dg * da[1];

        dm[0] = 0.;
        dm[1] = k;
        dm[2] = 0.;
    }

    for (int ch = start; ch < end; ch++) {
        const double *src = (const double *)in->extended_data[ch];
        double *dst = (double *)out->extended_data[ch];
        double *state = (double *)s->state->extended_data[ch];
        const double threshold = detection == 0 ? state[5] : s->threshold;

        if (detection < 0)
            state[5] = threshold;

        for (int n = 0; n < out->nb_samples; n++) {
            double detect, gain, v, listen;
            double fa[3], fm[3];
            double k, g;

            detect = listen = get_svf(src[n], dm, da, state);
            detect = fabs(detect);

            if (detection > 0)
                state[5] = fmax(state[5], detect);

            if (direction == 0 && mode == 0 && detect < threshold)
                detect = 1. / av_clipd(1. + makeup + (threshold - detect) * ratio, 1., range);
            else if (direction == 0 && mode == 1 && detect < threshold)
                detect = av_clipd(1. + makeup + (threshold - detect) * ratio, 1., range);
            else if (direction == 1 && mode == 0 && detect > threshold)
                detect = 1. / av_clipd(1. + makeup + (detect - threshold) * ratio, 1., range);
            else if (direction == 1 && mode == 1 && detect > threshold)
                detect = av_clipd(1. + makeup + (detect - threshold) * ratio, 1., range);
            else
                detect = 1.;

            if (direction == 0) {
                if (detect > state[4]) {
                    detect = iattack * detect + attack * state[4];
                } else {
                    detect = irelease * detect + release * state[4];
                }
            } else {
                if (detect < state[4]) {
                    detect = iattack * detect + attack * state[4];
                } else {
                    detect = irelease * detect + release * state[4];
                }
            }

            if (state[4] != detect || n == 0) {
                state[4] = gain = detect;

                switch (type) {
                case 0:
                    k = 1. / (tqfactor * gain);

                    fa[0] = 1. / (1. + fg * (fg + k));
                    fa[1] = fg * fa[0];
                    fa[2] = fg * fa[1];

                    fm[0] = 1.;
                    fm[1] = k * (gain * gain - 1.);
                    fm[2] = 0.;
                    break;
                case 1:
                    k = 1. / tqfactor;
                    g = fg / sqrt(gain);

                    fa[0] = 1. / (1. + g * (g + k));
                    fa[1] = g * fa[0];
                    fa[2] = g * fa[1];

                    fm[0] = 1.;
                    fm[1] = k * (gain - 1.);
                    fm[2] = gain * gain - 1.;
                    break;
                case 2:
                    k = 1. / tqfactor;
                    g = fg / sqrt(gain);

                    fa[0] = 1. / (1. + g * (g + k));
                    fa[1] = g * fa[0];
                    fa[2] = g * fa[1];

                    fm[0] = gain * gain;
                    fm[1] = k * (1. - gain) * gain;
                    fm[2] = 1. - gain * gain;
                    break;
                }
            }

            v = get_svf(src[n], fm, fa, &state[2]);
            v = mode == -1 ? listen : v;
            dst[n] = ctx->is_disabled ? src[n] : v;
        }
    }

    return 0;
}

static double get_coef(double x, double sr)
{
    return exp(-1000. / (x * sr));
}

static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
    AVFilterContext *ctx = inlink->dst;
    AVFilterLink *outlink = ctx->outputs[0];
    AudioDynamicEqualizerContext *s = ctx->priv;
    ThreadData td;
    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);
    }

    s->attack_coef = get_coef(s->attack, in->sample_rate);
    s->release_coef = get_coef(s->release, in->sample_rate);

    td.in = in;
    td.out = out;
    ff_filter_execute(ctx, filter_channels, &td, NULL,
                     FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));

    if (out != in)
        av_frame_free(&in);
    return ff_filter_frame(outlink, out);
}

static av_cold void uninit(AVFilterContext *ctx)
{
    AudioDynamicEqualizerContext *s = ctx->priv;

    av_frame_free(&s->state);
}

#define OFFSET(x) offsetof(AudioDynamicEqualizerContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM

static const AVOption adynamicequalizer_options[] = {
    { "threshold",  "set detection threshold", OFFSET(threshold),  AV_OPT_TYPE_DOUBLE, {.dbl=0},        0, 100,     FLAGS },
    { "dfrequency", "set detection frequency", OFFSET(dfrequency), AV_OPT_TYPE_DOUBLE, {.dbl=1000},     2, 1000000, FLAGS },
    { "dqfactor",   "set detection Q factor",  OFFSET(dqfactor),   AV_OPT_TYPE_DOUBLE, {.dbl=1},    0.001, 1000,    FLAGS },
    { "tfrequency", "set target frequency",    OFFSET(tfrequency), AV_OPT_TYPE_DOUBLE, {.dbl=1000},     2, 1000000, FLAGS },
    { "tqfactor",   "set target Q factor",     OFFSET(tqfactor),   AV_OPT_TYPE_DOUBLE, {.dbl=1},    0.001, 1000,    FLAGS },
    { "attack",     "set attack duration",     OFFSET(attack),     AV_OPT_TYPE_DOUBLE, {.dbl=20},       1, 2000,    FLAGS },
    { "release",    "set release duration",    OFFSET(release),    AV_OPT_TYPE_DOUBLE, {.dbl=200},      1, 2000,    FLAGS },
    { "ratio",      "set ratio factor",        OFFSET(ratio),      AV_OPT_TYPE_DOUBLE, {.dbl=1},        0, 30,      FLAGS },
    { "makeup",     "set makeup gain",         OFFSET(makeup),     AV_OPT_TYPE_DOUBLE, {.dbl=0},        0, 100,     FLAGS },
    { "range",      "set max gain",            OFFSET(range),      AV_OPT_TYPE_DOUBLE, {.dbl=50},       1, 200,     FLAGS },
    { "mode",       "set mode",                OFFSET(mode),       AV_OPT_TYPE_INT,    {.i64=0},       -1, 1,       FLAGS, "mode" },
    {   "listen",   0,                         0,                  AV_OPT_TYPE_CONST,  {.i64=-1},       0, 0,       FLAGS, "mode" },
    {   "cut",      0,                         0,                  AV_OPT_TYPE_CONST,  {.i64=0},        0, 0,       FLAGS, "mode" },
    {   "boost",    0,                         0,                  AV_OPT_TYPE_CONST,  {.i64=1},        0, 0,       FLAGS, "mode" },
    { "tftype",     "set target filter type",  OFFSET(type),       AV_OPT_TYPE_INT,    {.i64=0},        0, 2,       FLAGS, "type" },
    {   "bell",     0,                         0,                  AV_OPT_TYPE_CONST,  {.i64=0},        0, 0,       FLAGS, "type" },
    {   "lowshelf", 0,                         0,                  AV_OPT_TYPE_CONST,  {.i64=1},        0, 0,       FLAGS, "type" },
    {   "highshelf",0,                         0,                  AV_OPT_TYPE_CONST,  {.i64=2},        0, 0,       FLAGS, "type" },
    { "direction",  "set direction",           OFFSET(direction),  AV_OPT_TYPE_INT,    {.i64=0},        0, 1,       FLAGS, "direction" },
    {   "downward", 0,                         0,                  AV_OPT_TYPE_CONST,  {.i64=0},        0, 0,       FLAGS, "direction" },
    {   "upward",   0,                         0,                  AV_OPT_TYPE_CONST,  {.i64=1},        0, 0,       FLAGS, "direction" },
    { "auto",       "set auto threshold",      OFFSET(detection),  AV_OPT_TYPE_INT,    {.i64=-1},      -1, 1,       FLAGS, "auto" },
    {   "disabled", 0,                         0,                  AV_OPT_TYPE_CONST,  {.i64=-1},       0, 0,       FLAGS, "auto" },
    {   "off",      0,                         0,                  AV_OPT_TYPE_CONST,  {.i64=0},        0, 0,       FLAGS, "auto" },
    {   "on",       0,                         0,                  AV_OPT_TYPE_CONST,  {.i64=1},        0, 0,       FLAGS, "auto" },
    { NULL }
};

AVFILTER_DEFINE_CLASS(adynamicequalizer);

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

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

const AVFilter ff_af_adynamicequalizer = {
    .name            = "adynamicequalizer",
    .description     = NULL_IF_CONFIG_SMALL("Apply Dynamic Equalization of input audio."),
    .priv_size       = sizeof(AudioDynamicEqualizerContext),
    .priv_class      = &adynamicequalizer_class,
    .uninit          = uninit,
    FILTER_INPUTS(inputs),
    FILTER_OUTPUTS(outputs),
    FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
    .flags           = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
                       AVFILTER_FLAG_SLICE_THREADS,
    .process_command = ff_filter_process_command,
};