[FFmpeg-devel] [PATCH] avfilter: add declick audio filter

Paul B Mahol onemda at gmail.com
Sat Mar 24 15:41:24 EET 2018


Signed-off-by: Paul B Mahol <onemda at gmail.com>
---
 doc/filters.texi         |  35 +++
 libavfilter/Makefile     |   1 +
 libavfilter/af_declick.c | 586 +++++++++++++++++++++++++++++++++++++++++++++++
 libavfilter/allfilters.c |   1 +
 4 files changed, 623 insertions(+)
 create mode 100644 libavfilter/af_declick.c

diff --git a/doc/filters.texi b/doc/filters.texi
index 1620ae1cfa..9a067ba9ea 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2562,6 +2562,41 @@ Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is
 used to prevent clipping.
 @end table
 
+ at section declick
+Remove impulsive noise from input audio.
+
+Samples detected as impulsive noise are replaced by interpolated samples using
+autoregressive modeling.
+
+ at table @option
+ at item w
+Set window size, in milliseconds. Allowed range is from @code{10} to @code{100}.
+Default value is @code{54} milliseconds.
+This sets size of window which will be processed at once.
+
+ at item o
+Set window overlap, in percentage of window size. Allowed range is from @code{50}
+to @code{95}. Default value is @code{75} percent.
+Setting this to very high value increases impulsive noise removal but makes whole
+processs much slower.
+
+ at item a
+Set autoregression order, in percentage of window size. Allowed range is from
+ at code{1} to @code{50}. Default value is @code{2} percent. This option also controls
+quality of interpolated samples using neighbour good samples.
+
+ at item t
+Set threshold value. Allowed range is from @code{1} to @code{10}.
+Default value is @code{2}.
+This controls the strength of impulse noise which is going to be removed.
+
+ at item b
+Set burst fusion, in percentage of window size. Allowed range is @code{0} to
+ at code{40}. Default value is @code{10} percent.
+This controls between how much samples, which are detected as impulsive noise,
+any sample between 2 detected noise samples is considered also as noise sample.
+ at end table
+
 @section drmeter
 Measure audio dynamic range.
 
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 1043b41d80..978751d2a0 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -87,6 +87,7 @@ OBJS-$(CONFIG_COMPENSATIONDELAY_FILTER)      += af_compensationdelay.o
 OBJS-$(CONFIG_CROSSFEED_FILTER)              += af_crossfeed.o
 OBJS-$(CONFIG_CRYSTALIZER_FILTER)            += af_crystalizer.o
 OBJS-$(CONFIG_DCSHIFT_FILTER)                += af_dcshift.o
+OBJS-$(CONFIG_DECLICK_FILTER)                += af_declick.o
 OBJS-$(CONFIG_DRMETER_FILTER)                += af_drmeter.o
 OBJS-$(CONFIG_DYNAUDNORM_FILTER)             += af_dynaudnorm.o
 OBJS-$(CONFIG_EARWAX_FILTER)                 += af_earwax.o
diff --git a/libavfilter/af_declick.c b/libavfilter/af_declick.c
new file mode 100644
index 0000000000..0de4c35c95
--- /dev/null
+++ b/libavfilter/af_declick.c
@@ -0,0 +1,586 @@
+/*
+ * 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 DeclickContext {
+    const AVClass *class;
+    double w;
+    double overlap;
+    double threshold;
+    double ar;
+    double burst;
+
+    int ar_order;
+    int nb_burst_samples;
+    int window_size;
+    int hop_size;
+
+    AVFrame *in;
+    AVFrame *out;
+    AVFrame *buffer;
+    AVFrame *is;
+    double *auxiliary;
+    double *detection;
+    double *acoefficients;
+    double *acorrelation;
+    double *tmp;
+    double *interpolated;
+    double *matrix;
+    int matrix_size;
+    double *vector;
+    int vector_size;
+    uint8_t *click;
+    int *index;
+
+    double *ltriangular;
+    int ltriangular_size;
+    double *diagonal;
+    int d_size;
+    double *y;
+    int y_size;
+
+    int64_t pts;
+    uint64_t nb_samples;
+    uint64_t detected_clicks;
+    int samples_left;
+
+    AVAudioFifo *fifo;
+    double *window_func_lut;
+} DeclickContext;
+
+#define OFFSET(x) offsetof(DeclickContext, x)
+#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption declick_options[] = {
+    { "w", "set window size",          OFFSET(w),         AV_OPT_TYPE_DOUBLE, {.dbl=54}, 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},   1,   50, AF },
+    { "t", "set threshold",            OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=2},   1,   10, AF },
+    { "b", "set burst fusion",         OFFSET(burst),     AV_OPT_TYPE_DOUBLE, {.dbl=10},  0,   40, AF },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(declick);
+
+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;
+    DeclickContext *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 = s->window_size * s->ar / 100.;
+    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->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, s->window_size);
+    if (!s->fifo)
+        return AVERROR(ENOMEM);
+
+    s->window_func_lut = av_realloc_f(s->window_func_lut, 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);
+    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->detection = av_calloc(s->window_size, sizeof(*s->detection));
+    s->auxiliary = av_calloc(s->ar_order + 1, sizeof(*s->auxiliary));
+    s->acoefficients = av_calloc(s->ar_order + 1, sizeof(*s->acoefficients));
+    s->acorrelation = av_calloc(s->ar_order + 1, sizeof(*s->acorrelation));
+    s->tmp = av_calloc(s->ar_order, sizeof(*s->tmp));
+    s->click = av_calloc(s->window_size, sizeof(*s->click));
+    s->index = av_calloc(s->window_size, sizeof(*s->index));
+    s->interpolated = av_calloc(s->window_size, sizeof(*s->interpolated));
+    if (!s->auxiliary || !s->acoefficients || !s->detection || !s->click ||
+        !s->index || !s->interpolated || !s->acorrelation || !s->tmp)
+        return AVERROR(ENOMEM);
+
+    return 0;
+}
+
+static void autocorrelation(const double *input, int order, int size, double *output)
+{
+    double scale = 1. / size;
+    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);
+
+    /* 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 = 0; i < ar_order; i++)
+        k[i + 1] = a[i];
+
+    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 cholesky_decomposition(DeclickContext *s, double *matrix, double *vector, int n, double *out)
+{
+    double *ltriangular, *diagonal, *y;
+    int i, j, k;
+
+    av_fast_malloc(&s->ltriangular, &s->ltriangular_size, n * n * sizeof(*s->ltriangular));
+    ltriangular = s->ltriangular;
+
+    av_fast_malloc(&s->diagonal, &s->d_size, n * sizeof(*s->diagonal));
+    diagonal = s->diagonal;
+
+    av_fast_malloc(&s->y, &s->y_size, n * sizeof(*s->y));
+    y = s->y;
+
+    if (!ltriangular || !diagonal || !y)
+        return AVERROR(ENOMEM);
+
+    memset(s->ltriangular, 0, n * n * sizeof(*s->ltriangular));
+    memset(s->diagonal, 0, n * sizeof(*s->diagonal));
+    memset(s->y, 0, n * sizeof(*s->y));
+
+    for (i = 0; i < n; i++) {
+        const int in = i * n;
+
+        diagonal[i] = matrix[in + i];
+        for (j = 0; j < i; j++)
+            diagonal[i] -= diagonal[j] * ltriangular[in + j] * ltriangular[in + j];
+
+        if (diagonal[i] == 0.) {
+            return -1;
+        }
+
+        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 -= diagonal[k] * ltriangular[in + k] * ltriangular[jn + k];
+            ltriangular[in + j] = ltriangular[jn + i] = x / diagonal[i];
+        }
+    }
+
+    for (i = 0; i < n; i++) {
+        const int in = i * n;
+
+        y[i] = vector[i];
+        for (j = 0; j <= i; j++)
+            y[i] -= ltriangular[in + j] * y[j];
+    }
+
+    for (i = n - 1; i >= 0; i--) {
+        const int in = i * n;
+
+        out[i] = y[i] / diagonal[i];
+        for (j = i; j < n; j++)
+            out[i] -= ltriangular[in + j] * out[j];
+    }
+
+    return 0;
+}
+
+static int interpolation(DeclickContext *s, const double *src, int ar_order,
+                         double *acoefficients, int *index, int nb_clicks,
+                         double *auxiliary, double *interpolated)
+{
+    double *vector, *matrix;
+    int i, j;
+
+    av_fast_malloc(&s->matrix, &s->matrix_size, nb_clicks * nb_clicks * sizeof(*s->matrix));
+    matrix = s->matrix;
+    if (!matrix)
+        return AVERROR(ENOMEM);
+
+    av_fast_malloc(&s->vector, &s->vector_size, nb_clicks * sizeof(*s->vector));
+    vector = s->vector;
+    if (!vector)
+        return AVERROR(ENOMEM);
+
+    for (i = 0; i <= ar_order; i++) {
+        auxiliary[i] = 0.;
+        for (j = i; j <= ar_order; j++)
+            auxiliary[i] += acoefficients[j] * acoefficients[j - i];
+    }
+
+    for (i = 0; i < nb_clicks; i++) {
+        const int im = i * nb_clicks;
+
+        for (j = i; j < nb_clicks; j++) {
+            if (abs(index[j] - index[i]) <= ar_order) {
+                matrix[j * nb_clicks + i] = matrix[im + j] = auxiliary[abs(index[j] - index[i])];
+            } else {
+                matrix[j * nb_clicks + i] = matrix[im + j] = 0;
+            }
+        }
+    }
+
+    for (i = 0; i < nb_clicks; i++) {
+        vector[i] = 0.;
+        for (j = -ar_order; j <= ar_order; j++)
+            if (find_index(index, index[i] - j, nb_clicks))
+                vector[i] -= src[index[i] - j] * auxiliary[abs(j)];
+    }
+
+    return cholesky_decomposition(s, matrix, vector, nb_clicks, interpolated);
+}
+
+static int detect_clicks(DeclickContext *s, 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;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AVFilterLink *outlink = ctx->outputs[0];
+    DeclickContext *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);
+    if (ret < 0) {
+        av_frame_free(&out);
+        return ret;
+    }
+
+    while (av_audio_fifo_size(s->fifo) >= s->window_size) {
+        int j, ch, detected_clicks = 0;
+
+        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;
+
+        for (ch = 0; ch < s->in->channels; ch++) {
+            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;
+            double sigmae;
+
+            sigmae = autoregression(src, s->ar_order, s->window_size, s->acoefficients, s->acorrelation, s->tmp);
+
+            if (isfinite_array(s->acoefficients, s->ar_order + 1)) {
+                double *interpolated = s->interpolated;
+                int *index = s->index;
+                int nb_clicks;
+
+                nb_clicks = detect_clicks(s, sigmae, s->detection, s->acoefficients,
+                                          s->click, index, src, dst);
+                if (nb_clicks > 0) {
+                    ret = interpolation(s, src, s->ar_order, s->acoefficients, index,
+                                        nb_clicks, s->auxiliary, interpolated);
+                    if (ret < 0)
+                        goto fail;
+
+                    for (j = 0; j < nb_clicks; j++) {
+                        dst[index[j]] = interpolated[j];
+                        is[index[j]] = 1;
+                    }
+                }
+            } else {
+                memcpy(dst, src, s->window_size * sizeof(*dst));
+            }
+
+            for (j = 0; j < s->window_size; j++)
+                buf[j] += dst[j] * w[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));
+
+            for (j = 0; j < s->hop_size; j++) {
+                if (is[j])
+                    detected_clicks++;
+            }
+        }
+
+        av_audio_fifo_drain(s->fifo, s->hop_size);
+
+        out->pts = s->pts;
+        s->pts += s->hop_size;
+
+        s->detected_clicks += detected_clicks;
+        s->nb_samples += s->hop_size * inlink->channels;
+
+        ret = ff_filter_frame(outlink, out);
+        if (ret < 0)
+            break;
+    }
+
+fail:
+    if (ret < 0)
+        av_frame_free(&out);
+    return ret;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    DeclickContext *s = ctx->priv;
+    int ret = 0;
+
+    ret = ff_request_frame(ctx->inputs[0]);
+
+    if (ret == AVERROR_EOF && av_audio_fifo_size(s->fifo) > 0) {
+        AVFrame *in;
+
+        if (!s->samples_left)
+            s->samples_left = av_audio_fifo_size(s->fifo);
+
+        in = ff_get_audio_buffer(outlink, s->window_size);
+        if (!in)
+            return AVERROR(ENOMEM);
+        ret = filter_frame(ctx->inputs[0], in);
+        if (s->samples_left) {
+            s->samples_left -= s->hop_size;
+            if (s->samples_left <= 0)
+                av_audio_fifo_drain(s->fifo, s->window_size);
+        }
+    }
+
+    return ret;
+}
+
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    DeclickContext *s = ctx->priv;
+
+    av_log(ctx, AV_LOG_INFO, "Detected clicks in %"PRId64" of %"PRId64" samples (%g%%).\n",
+           s->detected_clicks, s->nb_samples, 100. * s->detected_clicks / 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);
+    av_freep(&s->detection);
+    av_freep(&s->auxiliary);
+    av_freep(&s->acoefficients);
+    av_freep(&s->acorrelation);
+    av_freep(&s->tmp);
+    av_freep(&s->click);
+    av_freep(&s->index);
+    av_freep(&s->interpolated);
+    av_freep(&s->matrix);
+    s->matrix_size = 0;
+    av_freep(&s->vector);
+    s->vector_size = 0;
+    av_freep(&s->ltriangular);
+    s->ltriangular_size = 0;
+    av_freep(&s->diagonal);
+    s->d_size = 0;
+    av_freep(&s->y);
+    s->y_size = 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_declick = {
+    .name          = "declick",
+    .description   = NULL_IF_CONFIG_SMALL("Remove impulsive noise from input audio."),
+    .query_formats = query_formats,
+    .priv_size     = sizeof(DeclickContext),
+    .priv_class    = &declick_class,
+    .uninit        = uninit,
+    .inputs        = inputs,
+    .outputs       = outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 3f67e321bf..cf5016f2c1 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -98,6 +98,7 @@ static void register_all(void)
     REGISTER_FILTER(CROSSFEED,      crossfeed,      af);
     REGISTER_FILTER(CRYSTALIZER,    crystalizer,    af);
     REGISTER_FILTER(DCSHIFT,        dcshift,        af);
+    REGISTER_FILTER(DECLICK,        declick,        af);
     REGISTER_FILTER(DRMETER,        drmeter,        af);
     REGISTER_FILTER(DYNAUDNORM,     dynaudnorm,     af);
     REGISTER_FILTER(EARWAX,         earwax,         af);
-- 
2.11.0



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