avfilter: add adeclick and adeclip audio filters

Signed-off-by: Paul B Mahol <onemda@gmail.com>

1 files changed, 753 insertions, 0 deletions

diff --git a/libavfilter/af_adeclick.c b/libavfilter/af_adeclick.c
new file mode 100644
index 0000000000..bf0b7cb408
--- /dev/null
+++ b/libavfilter/af_adeclick.c
@@ -0,0 +1,753 @@
+/*
+ * Copyright (c) 2018 Paul B Mahol
+ *
+ * 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/audio_fifo.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "formats.h"
+
+typedef struct DeclickChannel {
+    double *auxiliary;
+    double *detection;
+    double *acoefficients;
+    double *acorrelation;
+    double *tmp;
+    double *interpolated;
+    double *matrix;
+    int matrix_size;
+    double *vector;
+    int vector_size;
+    double *y;
+    int y_size;
+    uint8_t *click;
+    int *index;
+    unsigned *histogram;
+    int histogram_size;
+} DeclickChannel;
+
+typedef struct AudioDeclickContext {
+    const AVClass *class;
+
+    double w;
+    double overlap;
+    double threshold;
+    double ar;
+    double burst;
+    int method;
+    int nb_hbins;
+
+    int is_declip;
+    int ar_order;
+    int nb_burst_samples;
+    int window_size;
+    int hop_size;
+    int overlap_skip;
+
+    AVFrame *in;
+    AVFrame *out;
+    AVFrame *buffer;
+    AVFrame *is;
+
+    DeclickChannel *chan;
+
+    int64_t pts;
+    int nb_channels;
+    uint64_t nb_samples;
+    uint64_t detected_errors;
+    int samples_left;
+
+    AVAudioFifo *fifo;
+    double *window_func_lut;
+
+    int (*detector)(struct AudioDeclickContext *s, DeclickChannel *c,
+                    double sigmae, double *detection,
+                    double *acoefficients, uint8_t *click, int *index,
+                    const double *src, double *dst);
+} AudioDeclickContext;
+
+#define OFFSET(x) offsetof(AudioDeclickContext, x)
+#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption adeclick_options[] = {
+    { "w", "set window size",          OFFSET(w),         AV_OPT_TYPE_DOUBLE, {.dbl=55}, 10,  100, AF },
+    { "o", "set window overlap",       OFFSET(overlap),   AV_OPT_TYPE_DOUBLE, {.dbl=75}, 50,   95, AF },
+    { "a", "set autoregression order", OFFSET(ar),        AV_OPT_TYPE_DOUBLE, {.dbl=2},   0,   25, AF },
+    { "t", "set threshold",            OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=2},   1,  100, AF },
+    { "b", "set burst fusion",         OFFSET(burst),     AV_OPT_TYPE_DOUBLE, {.dbl=2},   0,   10, AF },
+    { "m", "set overlap method",       OFFSET(method),    AV_OPT_TYPE_INT,    {.i64=0},   0,    1, AF, "m" },
+    { "a", "overlap-add",              0,                 AV_OPT_TYPE_CONST,  {.i64=0},   0,    0, AF, "m" },
+    { "s", "overlap-save",             0,                 AV_OPT_TYPE_CONST,  {.i64=1},   0,    0, AF, "m" },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(adeclick);
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats = NULL;
+    AVFilterChannelLayouts *layouts = NULL;
+    static const enum AVSampleFormat sample_fmts[] = {
+        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);
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AudioDeclickContext *s = ctx->priv;
+    int i;
+
+    s->pts = AV_NOPTS_VALUE;
+    s->window_size = inlink->sample_rate * s->w / 1000.;
+    if (s->window_size < 100)
+        return AVERROR(EINVAL);
+    s->ar_order = FFMAX(s->window_size * s->ar / 100., 1);
+    s->nb_burst_samples = s->window_size * s->burst / 1000.;
+    s->hop_size = s->window_size * (1. - (s->overlap / 100.));
+    if (s->hop_size < 1)
+        return AVERROR(EINVAL);
+
+    s->window_func_lut = av_calloc(s->window_size, sizeof(*s->window_func_lut));
+    if (!s->window_func_lut)
+        return AVERROR(ENOMEM);
+    for (i = 0; i < s->window_size; i++)
+        s->window_func_lut[i] = sin(M_PI * i / s->window_size) *
+                                (1. - (s->overlap / 100.)) * M_PI_2;
+
+    av_frame_free(&s->in);
+    av_frame_free(&s->out);
+    av_frame_free(&s->buffer);
+    av_frame_free(&s->is);
+    s->in = ff_get_audio_buffer(inlink, s->window_size);
+    s->out = ff_get_audio_buffer(inlink, s->window_size);
+    s->buffer = ff_get_audio_buffer(inlink, s->window_size * 2);
+    s->is = ff_get_audio_buffer(inlink, s->window_size);
+    if (!s->in || !s->out || !s->buffer || !s->is)
+        return AVERROR(ENOMEM);
+
+    s->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, s->window_size);
+    if (!s->fifo)
+        return AVERROR(ENOMEM);
+    s->overlap_skip = s->method ? (s->window_size - s->hop_size) / 2 : 0;
+    if (s->overlap_skip > 0) {
+        av_audio_fifo_write(s->fifo, (void **)s->in->extended_data,
+                            s->overlap_skip);
+    }
+
+    s->nb_channels = inlink->channels;
+    s->chan = av_calloc(inlink->channels, sizeof(*s->chan));
+    if (!s->chan)
+        return AVERROR(ENOMEM);
+
+    for (i = 0; i < inlink->channels; i++) {
+        DeclickChannel *c = &s->chan[i];
+
+        c->detection = av_calloc(s->window_size, sizeof(*c->detection));
+        c->auxiliary = av_calloc(s->ar_order + 1, sizeof(*c->auxiliary));
+        c->acoefficients = av_calloc(s->ar_order + 1, sizeof(*c->acoefficients));
+        c->acorrelation = av_calloc(s->ar_order + 1, sizeof(*c->acorrelation));
+        c->tmp = av_calloc(s->ar_order, sizeof(*c->tmp));
+        c->click = av_calloc(s->window_size, sizeof(*c->click));
+        c->index = av_calloc(s->window_size, sizeof(*c->index));
+        c->interpolated = av_calloc(s->window_size, sizeof(*c->interpolated));
+        if (!c->auxiliary || !c->acoefficients || !c->detection || !c->click ||
+            !c->index || !c->interpolated || !c->acorrelation || !c->tmp)
+            return AVERROR(ENOMEM);
+    }
+
+    return 0;
+}
+
+static void autocorrelation(const double *input, int order, int size,
+                            double *output, double scale)
+{
+    int i, j;
+
+    for (i = 0; i <= order; i++) {
+        double value = 0.;
+
+        for (j = i; j < size; j++)
+            value += input[j] * input[j - i];
+
+        output[i] = value * scale;
+    }
+}
+
+static double autoregression(const double *samples, int ar_order,
+                             int nb_samples, double *k, double *r, double *a)
+{
+    double alpha;
+    int i, j;
+
+    memset(a, 0, ar_order * sizeof(*a));
+
+    autocorrelation(samples, ar_order, nb_samples, r, 1. / nb_samples);
+
+    /* Levinson-Durbin algorithm */
+    k[0] = a[0] = -r[1] / r[0];
+    alpha = r[0] * (1. - k[0] * k[0]);
+    for (i = 1; i < ar_order; i++) {
+        double epsilon = 0.;
+
+        for (j = 0; j < i; j++)
+            epsilon += a[j] * r[i - j];
+        epsilon += r[i + 1];
+
+        k[i] = -epsilon / alpha;
+        alpha *= (1. - k[i] * k[i]);
+        for (j = i - 1; j >= 0; j--)
+            k[j] = a[j] + k[i] * a[i - j - 1];
+        for (j = 0; j <= i; j++)
+            a[j] = k[j];
+    }
+
+    k[0] = 1.;
+    for (i = 1; i <= ar_order; i++)
+        k[i] = a[i - 1];
+
+    return sqrt(alpha);
+}
+
+static int isfinite_array(double *samples, int nb_samples)
+{
+    int i;
+
+    for (i = 0; i < nb_samples; i++)
+        if (!isfinite(samples[i]))
+            return 0;
+
+    return 1;
+}
+
+static int find_index(int *index, int value, int size)
+{
+    int i, start, end;
+
+    if ((value < index[0]) || (value > index[size - 1]))
+        return 1;
+
+    i = start = 0;
+    end = size - 1;
+
+    while (start <= end) {
+        i = (end + start) / 2;
+        if (index[i] == value)
+            return 0;
+        if (value < index[i])
+            end = i - 1;
+        if (value > index[i])
+            start = i + 1;
+    }
+
+    return 1;
+}
+
+static int factorization(double *matrix, int n)
+{
+    int i, j, k;
+
+    for (i = 0; i < n; i++) {
+        const int in = i * n;
+        double value;
+
+        value = matrix[in + i];
+        for (j = 0; j < i; j++)
+            value -= matrix[j * n + j] * matrix[in + j] * matrix[in + j];
+
+        if (value == 0.) {
+            return -1;
+        }
+
+        matrix[in + i] = value;
+        for (j = i + 1; j < n; j++) {
+            const int jn = j * n;
+            double x;
+
+            x = matrix[jn + i];
+            for (k = 0; k < i; k++)
+                x -= matrix[k * n + k] * matrix[in + k] * matrix[jn + k];
+            matrix[jn + i] = x / matrix[in + i];
+        }
+    }
+
+    return 0;
+}
+
+static int do_interpolation(DeclickChannel *c, double *matrix,
+                            double *vector, int n, double *out)
+{
+    int i, j, ret;
+    double *y;
+
+    ret = factorization(matrix, n);
+    if (ret < 0)
+        return ret;
+
+    av_fast_malloc(&c->y, &c->y_size, n * sizeof(*c->y));
+    y = c->y;
+    if (!y)
+        return AVERROR(ENOMEM);
+
+    for (i = 0; i < n; i++) {
+        const int in = i * n;
+        double value;
+
+        value = vector[i];
+        for (j = 0; j < i; j++)
+            value -= matrix[in + j] * y[j];
+        y[i] = value;
+    }
+
+    for (i = n - 1; i >= 0; i--) {
+        out[i] = y[i] / matrix[i * n + i];
+        for (j = i + 1; j < n; j++)
+            out[i] -= matrix[j * n + i] * out[j];
+    }
+
+    return 0;
+}
+
+static int interpolation(DeclickChannel *c, const double *src, int ar_order,
+                         double *acoefficients, int *index, int nb_errors,
+                         double *auxiliary, double *interpolated)
+{
+    double *vector, *matrix;
+    int i, j;
+
+    av_fast_malloc(&c->matrix, &c->matrix_size, nb_errors * nb_errors * sizeof(*c->matrix));
+    matrix = c->matrix;
+    if (!matrix)
+        return AVERROR(ENOMEM);
+
+    av_fast_malloc(&c->vector, &c->vector_size, nb_errors * sizeof(*c->vector));
+    vector = c->vector;
+    if (!vector)
+        return AVERROR(ENOMEM);
+
+    autocorrelation(acoefficients, ar_order, ar_order + 1, auxiliary, 1.);
+
+    for (i = 0; i < nb_errors; i++) {
+        const int im = i * nb_errors;
+
+        for (j = i; j < nb_errors; j++) {
+            if (abs(index[j] - index[i]) <= ar_order) {
+                matrix[j * nb_errors + i] = matrix[im + j] = auxiliary[abs(index[j] - index[i])];
+            } else {
+                matrix[j * nb_errors + i] = matrix[im + j] = 0;
+            }
+        }
+    }
+
+    for (i = 0; i < nb_errors; i++) {
+        double value = 0.;
+
+        for (j = -ar_order; j <= ar_order; j++)
+            if (find_index(index, index[i] - j, nb_errors))
+                value -= src[index[i] - j] * auxiliary[abs(j)];
+
+        vector[i] = value;
+    }
+
+    return do_interpolation(c, matrix, vector, nb_errors, interpolated);
+}
+
+static int detect_clips(AudioDeclickContext *s, DeclickChannel *c,
+                        double unused0,
+                        double *unused1, double *unused2,
+                        uint8_t *clip, int *index,
+                        const double *src, double *dst)
+{
+    const double threshold = s->threshold;
+    double max_amplitude = 0;
+    unsigned *histogram;
+    int i, nb_clips = 0;
+
+    av_fast_malloc(&c->histogram, &c->histogram_size, s->nb_hbins * sizeof(*c->histogram));
+    if (!c->histogram)
+        return AVERROR(ENOMEM);
+    histogram = c->histogram;
+    memset(histogram, 0, sizeof(*histogram) * s->nb_hbins);
+
+    for (i = 0; i < s->window_size; i++) {
+        const unsigned index = fmin(fabs(src[i]), 1) * (s->nb_hbins - 1);
+
+        histogram[index]++;
+        dst[i] = src[i];
+        clip[i] = 0;
+    }
+
+    for (i = s->nb_hbins - 1; i > 1; i--) {
+        if (histogram[i]) {
+            if (histogram[i] / (double)FFMAX(histogram[i - 1], 1) > threshold) {
+                max_amplitude = i / (double)s->nb_hbins;
+            }
+            break;
+        }
+    }
+
+    if (max_amplitude > 0.) {
+        for (i = 0; i < s->window_size; i++) {
+            clip[i] = fabs(src[i]) >= max_amplitude;
+        }
+    }
+
+    memset(clip, 0, s->ar_order * sizeof(*clip));
+    memset(clip + (s->window_size - s->ar_order), 0, s->ar_order * sizeof(*clip));
+
+    for (i = s->ar_order; i < s->window_size - s->ar_order; i++)
+        if (clip[i])
+            index[nb_clips++] = i;
+
+    return nb_clips;
+}
+
+static int detect_clicks(AudioDeclickContext *s, DeclickChannel *c,
+                         double sigmae,
+                         double *detection, double *acoefficients,
+                         uint8_t *click, int *index,
+                         const double *src, double *dst)
+{
+    const double threshold = s->threshold;
+    int i, j, nb_clicks = 0, prev = -1;
+
+    memset(detection, 0, s->window_size * sizeof(*detection));
+
+    for (i = s->ar_order; i < s->window_size; i++) {
+        for (j = 0; j <= s->ar_order; j++) {
+            detection[i] += acoefficients[j] * src[i - j];
+        }
+    }
+
+    for (i = 0; i < s->window_size; i++) {
+        click[i] = fabs(detection[i]) > sigmae * threshold;
+        dst[i] = src[i];
+    }
+
+    for (i = 0; i < s->window_size; i++) {
+        if (!click[i])
+            continue;
+
+        if (prev >= 0 && (i > prev + 1) && (i <= s->nb_burst_samples + prev))
+            for (j = prev + 1; j < i; j++)
+                click[j] = 1;
+        prev = i;
+    }
+
+    memset(click, 0, s->ar_order * sizeof(*click));
+    memset(click + (s->window_size - s->ar_order), 0, s->ar_order * sizeof(*click));
+
+    for (i = s->ar_order; i < s->window_size - s->ar_order; i++)
+        if (click[i])
+            index[nb_clicks++] = i;
+
+    return nb_clicks;
+}
+
+typedef struct ThreadData {
+    AVFrame *out;
+} ThreadData;
+
+static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
+{
+    AudioDeclickContext *s = ctx->priv;
+    ThreadData *td = arg;
+    AVFrame *out = td->out;
+    const double *src = (const double *)s->in->extended_data[ch];
+    double *is = (double *)s->is->extended_data[ch];
+    double *dst = (double *)s->out->extended_data[ch];
+    double *ptr = (double *)out->extended_data[ch];
+    double *buf = (double *)s->buffer->extended_data[ch];
+    const double *w = s->window_func_lut;
+    DeclickChannel *c = &s->chan[ch];
+    double sigmae;
+    int j, ret;
+
+    sigmae = autoregression(src, s->ar_order, s->window_size, c->acoefficients, c->acorrelation, c->tmp);
+
+    if (isfinite_array(c->acoefficients, s->ar_order + 1)) {
+        double *interpolated = c->interpolated;
+        int *index = c->index;
+        int nb_errors;
+
+        nb_errors = s->detector(s, c, sigmae, c->detection, c->acoefficients,
+                                c->click, index, src, dst);
+        if (nb_errors > 0) {
+            ret = interpolation(c, src, s->ar_order, c->acoefficients, index,
+                                nb_errors, c->auxiliary, interpolated);
+            if (ret < 0)
+                return ret;
+
+            for (j = 0; j < nb_errors; j++) {
+                dst[index[j]] = interpolated[j];
+                is[index[j]] = 1;
+            }
+        }
+    } else {
+        memcpy(dst, src, s->window_size * sizeof(*dst));
+    }
+
+    if (s->method == 0) {
+        for (j = 0; j < s->window_size; j++)
+            buf[j] += dst[j] * w[j];
+    } else {
+        const int skip = s->overlap_skip;
+
+        for (j = 0; j < s->hop_size; j++)
+            buf[j] = dst[skip + j];
+    }
+    for (j = 0; j < s->hop_size; j++)
+        ptr[j] = buf[j];
+
+    memmove(buf, buf + s->hop_size, (s->window_size * 2 - s->hop_size) * sizeof(*buf));
+    memmove(is, is + s->hop_size, (s->window_size - s->hop_size) * sizeof(*is));
+    memset(buf + s->window_size * 2 - s->hop_size, 0, s->hop_size * sizeof(*buf));
+    memset(is + s->window_size - s->hop_size, 0, s->hop_size * sizeof(*is));
+
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AVFilterLink *outlink = ctx->outputs[0];
+    AudioDeclickContext *s = ctx->priv;
+    AVFrame *out = NULL;
+    int ret = 0;
+
+    if (s->pts == AV_NOPTS_VALUE)
+        s->pts = in->pts;
+
+    ret = av_audio_fifo_write(s->fifo, (void **)in->extended_data,
+                              in->nb_samples);
+    av_frame_free(&in);
+
+    while (av_audio_fifo_size(s->fifo) >= s->window_size) {
+        int j, ch, detected_errors = 0;
+        ThreadData td;
+
+        out = ff_get_audio_buffer(outlink, s->hop_size);
+        if (!out)
+            return AVERROR(ENOMEM);
+
+        ret = av_audio_fifo_peek(s->fifo, (void **)s->in->extended_data,
+                                 s->window_size);
+        if (ret < 0)
+            break;
+
+        td.out = out;
+        ret = ctx->internal->execute(ctx, filter_channel, &td, NULL, inlink->channels);
+        if (ret < 0)
+            goto fail;
+
+        for (ch = 0; ch < s->in->channels; ch++) {
+            double *is = (double *)s->is->extended_data[ch];
+
+            for (j = 0; j < s->hop_size; j++) {
+                if (is[j])
+                    detected_errors++;
+            }
+        }
+
+        av_audio_fifo_drain(s->fifo, s->hop_size);
+
+        if (s->samples_left > 0)
+            out->nb_samples = FFMIN(s->hop_size, s->samples_left);
+
+        out->pts = s->pts;
+        s->pts += s->hop_size;
+
+        s->detected_errors += detected_errors;
+        s->nb_samples += out->nb_samples * inlink->channels;
+
+        ret = ff_filter_frame(outlink, out);
+        if (ret < 0)
+            break;
+
+        if (s->samples_left > 0) {
+            s->samples_left -= s->hop_size;
+            if (s->samples_left <= 0)
+                av_audio_fifo_drain(s->fifo, av_audio_fifo_size(s->fifo));
+        }
+    }
+
+fail:
+    if (ret < 0)
+        av_frame_free(&out);
+    return ret;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    AudioDeclickContext *s = ctx->priv;
+    int ret = 0;
+
+    ret = ff_request_frame(ctx->inputs[0]);
+
+    if (ret == AVERROR_EOF && av_audio_fifo_size(s->fifo) > 0) {
+        if (!s->samples_left)
+            s->samples_left = av_audio_fifo_size(s->fifo) - s->overlap_skip;
+
+        if (s->samples_left > 0) {
+            AVFrame *in = ff_get_audio_buffer(outlink, s->window_size - s->samples_left);
+            if (!in)
+                return AVERROR(ENOMEM);
+            ret = filter_frame(ctx->inputs[0], in);
+        }
+    }
+
+    return ret;
+}
+
+static av_cold int init(AVFilterContext *ctx)
+{
+    AudioDeclickContext *s = ctx->priv;
+
+    s->is_declip = !strcmp(ctx->filter->name, "adeclip");
+    if (s->is_declip) {
+        s->detector = detect_clips;
+    } else {
+        s->detector = detect_clicks;
+    }
+
+    return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    AudioDeclickContext *s = ctx->priv;
+    int i;
+
+    av_log(ctx, AV_LOG_INFO, "Detected %s in %"PRId64" of %"PRId64" samples (%g%%).\n",
+           s->is_declip ? "clips" : "clicks", s->detected_errors,
+           s->nb_samples, 100. * s->detected_errors / s->nb_samples);
+
+    av_audio_fifo_free(s->fifo);
+    av_freep(&s->window_func_lut);
+    av_frame_free(&s->in);
+    av_frame_free(&s->out);
+    av_frame_free(&s->buffer);
+    av_frame_free(&s->is);
+
+    if (s->chan) {
+        for (i = 0; i < s->nb_channels; i++) {
+            DeclickChannel *c = &s->chan[i];
+
+            av_freep(&c->detection);
+            av_freep(&c->auxiliary);
+            av_freep(&c->acoefficients);
+            av_freep(&c->acorrelation);
+            av_freep(&c->tmp);
+            av_freep(&c->click);
+            av_freep(&c->index);
+            av_freep(&c->interpolated);
+            av_freep(&c->matrix);
+            c->matrix_size = 0;
+            av_freep(&c->histogram);
+            c->histogram_size = 0;
+            av_freep(&c->vector);
+            c->vector_size = 0;
+            av_freep(&c->y);
+            c->y_size = 0;
+        }
+    }
+    av_freep(&s->chan);
+    s->nb_channels = 0;
+}
+
+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,
+        .request_frame = request_frame,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_adeclick = {
+    .name          = "adeclick",
+    .description   = NULL_IF_CONFIG_SMALL("Remove impulsive noise from input audio."),
+    .query_formats = query_formats,
+    .priv_size     = sizeof(AudioDeclickContext),
+    .priv_class    = &adeclick_class,
+    .init          = init,
+    .uninit        = uninit,
+    .inputs        = inputs,
+    .outputs       = outputs,
+    .flags         = AVFILTER_FLAG_SLICE_THREADS,
+};
+
+static const AVOption adeclip_options[] = {
+    { "w", "set window size",          OFFSET(w),              AV_OPT_TYPE_DOUBLE, {.dbl=55},     10,  100, AF },
+    { "o", "set window overlap",       OFFSET(overlap),        AV_OPT_TYPE_DOUBLE, {.dbl=75},     50,   95, AF },
+    { "a", "set autoregression order", OFFSET(ar),             AV_OPT_TYPE_DOUBLE, {.dbl=8},       0,   25, AF },
+    { "t", "set threshold",            OFFSET(threshold),      AV_OPT_TYPE_DOUBLE, {.dbl=10},      1,  100, AF },
+    { "n", "set histogram size",       OFFSET(nb_hbins),       AV_OPT_TYPE_INT,    {.i64=1000},  100, 9999, AF },
+    { "m", "set overlap method",       OFFSET(method),         AV_OPT_TYPE_INT,    {.i64=0},       0,    1, AF, "m" },
+    { "a", "overlap-add",              0,                      AV_OPT_TYPE_CONST,  {.i64=0},       0,    0, AF, "m" },
+    { "s", "overlap-save",             0,                      AV_OPT_TYPE_CONST,  {.i64=1},       0,    0, AF, "m" },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(adeclip);
+
+AVFilter ff_af_adeclip = {
+    .name          = "adeclip",
+    .description   = NULL_IF_CONFIG_SMALL("Remove clipping from input audio."),
+    .query_formats = query_formats,
+    .priv_size     = sizeof(AudioDeclickContext),
+    .priv_class    = &adeclip_class,
+    .init          = init,
+    .uninit        = uninit,
+    .inputs        = inputs,
+    .outputs       = outputs,
+    .flags         = AVFILTER_FLAG_SLICE_THREADS,
+};