diff --git a/doc/filters.texi b/doc/filters.texi
index e46839bfec..9c50b2e4b2 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -355,6 +355,34 @@ build.
 
 Below is a description of the currently available audio filters.
 
+@section acomb
+Apply comb audio filtering.
+
+Amplifies or attenuates certain frequencies by the superposition of a
+delayed version of the original audio signal onto itself.
+
+@table @option
+@item t
+Set comb filtering type.
+
+It accepts the following values:
+@table @option
+@item f
+set feedforward type
+@item b
+set feedback type
+@end table
+
+@item b0
+Set direct signal gain. Default is 1. Allowed range is from 0 to 1.
+
+@item xM
+Set delayed line gain. Default is 0.5. Allowed range is from 0 to 1.
+
+@item M
+Set delay in number of samples. Default is 10. Allowed range is from 1 to 327680.
+@end table
+
 @section acompressor
 
 A compressor is mainly used to reduce the dynamic range of a signal.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 182fe9df4b..d8a16d6e15 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -31,6 +31,7 @@ include $(SRC_PATH)/libavfilter/dnn/Makefile
 
 # audio filters
 OBJS-$(CONFIG_ABENCH_FILTER)                 += f_bench.o
+OBJS-$(CONFIG_ACOMB_FILTER)                  += af_acomb.o
 OBJS-$(CONFIG_ACOMPRESSOR_FILTER)            += af_sidechaincompress.o
 OBJS-$(CONFIG_ACONTRAST_FILTER)              += af_acontrast.o
 OBJS-$(CONFIG_ACOPY_FILTER)                  += af_acopy.o
diff --git a/libavfilter/af_acomb.c b/libavfilter/af_acomb.c
new file mode 100644
index 0000000000..3b0730c363
--- /dev/null
+++ b/libavfilter/af_acomb.c
@@ -0,0 +1,188 @@
+/*
+ * 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/opt.h"
+#include "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+
+typedef struct AudioCombContext {
+    const AVClass *class;
+
+    double b0, xM;
+    int t, M;
+
+    int head;
+    int tail;
+
+    AVFrame *delayframe;
+
+    void (*filter)(struct AudioCombContext *s, AVFrame *in, AVFrame *out);
+} AudioCombContext;
+
+#define OFFSET(x) offsetof(AudioCombContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption acomb_options[] = {
+    { "t",  "set comb filter type",           OFFSET(t),  AV_OPT_TYPE_INT,    {.i64=0}, 0, 1, A, "t" },
+    { "f",  "feedforward",                    0,          AV_OPT_TYPE_CONST,  {.i64=0}, 0, 0, A, "t" },
+    { "b",  "feedback",                       0,          AV_OPT_TYPE_CONST,  {.i64=1}, 0, 0, A, "t" },
+    { "b0", "set direct signal gain",         OFFSET(b0), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, A },
+    { "xM", "set delayed line gain",          OFFSET(xM), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, A },
+    { "M",  "set delay in number of samples", OFFSET(M),  AV_OPT_TYPE_INT,    {.i64=10}, 1, 327680, A },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(acomb);
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats = NULL;
+    AVFilterChannelLayouts *layouts = NULL;
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_FLTP,
+        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 COMB(name, type, dir, t)                                \
+static void acomb_## name ## _ ##dir(AudioCombContext *s,       \
+                                     AVFrame *in, AVFrame *out) \
+{                                                               \
+    const type b0 = s->b0;                                      \
+    const type xM = s->xM;                                      \
+    const int M = s->M;                                         \
+    int head;                                                   \
+                                                                \
+    for (int c = 0; c < in->channels; c++) {                    \
+        const type *src = (const type *)in->extended_data[c];   \
+        type *delay = (type *)s->delayframe->extended_data[c];  \
+        type *dst = (type *)out->extended_data[c];              \
+                                                                \
+        head = s->head;                                         \
+        for (int n = 0; n < in->nb_samples; n++) {              \
+            dst[n] = b0 * src[n] + t * xM * delay[head];        \
+            if (t == 1)                                         \
+                delay[head] = src[n];                           \
+            else                                                \
+                delay[head] = dst[n];                           \
+            head++;                                             \
+            if (head >= M)                                      \
+                head = 0;                                       \
+        }                                                       \
+    }                                                           \
+                                                                \
+    s->head = head;                                             \
+}
+
+COMB(fltp, float,  f,  1)
+COMB(dblp, double, f,  1)
+COMB(fltp, float,  b, -1)
+COMB(dblp, double, b, -1)
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AudioCombContext *s = ctx->priv;
+
+    s->delayframe = ff_get_audio_buffer(inlink, s->M);
+    if (!s->delayframe)
+        return AVERROR(ENOMEM);
+
+    switch (inlink->format) {
+    case AV_SAMPLE_FMT_FLTP: s->filter = s->t ? acomb_fltp_b : acomb_fltp_f; break;
+    case AV_SAMPLE_FMT_DBLP: s->filter = s->t ? acomb_dblp_b : acomb_dblp_f; break;
+    }
+
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AudioCombContext *s = ctx->priv;
+    AVFilterLink *outlink = ctx->outputs[0];
+    AVFrame *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->filter(s, in, out);
+
+    av_frame_free(&in);
+    return ff_filter_frame(outlink, out);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    AudioCombContext *s = ctx->priv;
+
+    av_frame_free(&s->delayframe);
+}
+
+static const AVFilterPad acomb_inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .filter_frame = filter_frame,
+        .config_props = config_input,
+    },
+    { NULL }
+};
+
+static const AVFilterPad acomb_outputs[] = {
+    {
+        .name = "default",
+        .type = AVMEDIA_TYPE_AUDIO,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_acomb = {
+    .name          = "acomb",
+    .description   = NULL_IF_CONFIG_SMALL("Apply comb audio filter."),
+    .query_formats = query_formats,
+    .priv_size     = sizeof(AudioCombContext),
+    .priv_class    = &acomb_class,
+    .uninit        = uninit,
+    .inputs        = acomb_inputs,
+    .outputs       = acomb_outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 1a26129069..7417f9656d 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -24,6 +24,7 @@
 #include "config.h"
 
 extern AVFilter ff_af_abench;
+extern AVFilter ff_af_acomb;
 extern AVFilter ff_af_acompressor;
 extern AVFilter ff_af_acontrast;
 extern AVFilter ff_af_acopy;
