diff --git a/doc/filters.texi b/doc/filters.texi
index b902aca12d..66b86d4655 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -745,6 +745,20 @@ Select overlap-save method. Not interpolated samples remain unchanged.
 Default value is @code{a}.
 @end table
 
+@section adecorrelate
+Apply decorrelation to input audio stream.
+
+The filter accepts the following options:
+
+@table @option
+@item stages
+Set decorrelation stages of filtering. Allowed
+range is from 1 to 16. Default value is 6.
+
+@item seed
+Set random seed used for setting delay in samples across channels.
+@end table
+
 @section adelay
 
 Delay one or more audio channels.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 102ce7beff..f71c59ba8b 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -40,6 +40,7 @@ OBJS-$(CONFIG_ACRUSHER_FILTER)               += af_acrusher.o
 OBJS-$(CONFIG_ACUE_FILTER)                   += f_cue.o
 OBJS-$(CONFIG_ADECLICK_FILTER)               += af_adeclick.o
 OBJS-$(CONFIG_ADECLIP_FILTER)                += af_adeclick.o
+OBJS-$(CONFIG_ADECORRELATE_FILTER)           += af_adecorrelate.o
 OBJS-$(CONFIG_ADELAY_FILTER)                 += af_adelay.o
 OBJS-$(CONFIG_ADENORM_FILTER)                += af_adenorm.o
 OBJS-$(CONFIG_ADERIVATIVE_FILTER)            += af_aderivative.o
diff --git a/libavfilter/af_adecorrelate.c b/libavfilter/af_adecorrelate.c
new file mode 100644
index 0000000000..6113574125
--- /dev/null
+++ b/libavfilter/af_adecorrelate.c
@@ -0,0 +1,268 @@
+/*
+ * Copyright (c) 2013-2020 Michael Barbour <barbour.michael.0@gmail.com>
+ * Copyright (c) 2021 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/channel_layout.h"
+#include "libavutil/ffmath.h"
+#include "libavutil/lfg.h"
+#include "libavutil/random_seed.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "formats.h"
+
+#define MAX_STAGES 16
+#define FILTER_FC  1100.0
+#define RT60_LF    0.1
+#define RT60_HF    0.008
+
+typedef struct APContext {
+    int len, p;
+    double *mx, *my;
+    double b0, b1, a0, a1;
+} APContext;
+
+typedef struct ADecorrelateContext {
+    const AVClass *class;
+
+    int stages;
+    int64_t seed;
+
+    int nb_channels;
+    APContext (*ap)[MAX_STAGES];
+
+    AVLFG c;
+
+    void (*filter_channel)(AVFilterContext *ctx,
+                           int channel,
+                           AVFrame *in, AVFrame *out);
+} ADecorrelateContext;
+
+static int query_formats(AVFilterContext *ctx)
+{
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_DBLP,
+        AV_SAMPLE_FMT_NONE
+    };
+    int ret = ff_set_common_formats_from_list(ctx, sample_fmts);
+    if (ret < 0)
+        return ret;
+
+    ret = ff_set_common_all_channel_counts(ctx);
+    if (ret < 0)
+        return ret;
+
+    return ff_set_common_all_samplerates(ctx);
+}
+
+static int ap_init(APContext *ap, int fs, double delay)
+{
+    const int delay_samples = lrint(round(delay * fs));
+    const double gain_lf = -60.0 / (RT60_LF * fs) * delay_samples;
+    const double gain_hf = -60.0 / (RT60_HF * fs) * delay_samples;
+    const double w0 = 2.0 * M_PI * FILTER_FC / fs;
+    const double t = tan(w0 / 2.0);
+    const double g_hf = ff_exp10(gain_hf / 20.0);
+    const double gd = ff_exp10((gain_lf-gain_hf) / 20.0);
+    const double sgd = sqrt(gd);
+
+    ap->len = delay_samples + 1;
+    ap->p = 0;
+    ap->mx = av_calloc(ap->len, sizeof(*ap->mx));
+    ap->my = av_calloc(ap->len, sizeof(*ap->my));
+    if (!ap->mx || !ap->my)
+        return AVERROR(ENOMEM);
+
+    ap->a0 = t + sgd;
+    ap->a1 = (t - sgd) / ap->a0;
+    ap->b0 = (gd*t - sgd) / ap->a0 * g_hf;
+    ap->b1 = (gd*t + sgd) / ap->a0 * g_hf;
+    ap->a0 = 1.0;
+
+    return 0;
+}
+
+static void ap_free(APContext *ap)
+{
+    av_freep(&ap->mx);
+    av_freep(&ap->my);
+}
+
+static double ap_run(APContext *ap, double x)
+{
+    const int i0 = ((ap->p < 1) ? ap->len : ap->p)-1, i_n1 = ap->p, i_n2 = (ap->p+1 >= ap->len) ? 0 : ap->p+1;
+    const double r = ap->b1*x + ap->b0*ap->mx[i0] + ap->a1*ap->mx[i_n2] + ap->a0*ap->mx[i_n1] -
+                     ap->a1*ap->my[i0] - ap->b0*ap->my[i_n2] - ap->b1*ap->my[i_n1];
+
+    ap->mx[ap->p] = x;
+    ap->my[ap->p] = r;
+    ap->p = (ap->p+1 >= ap->len) ? 0 : ap->p+1;
+
+    return r;
+}
+
+static void filter_channel_dbl(AVFilterContext *ctx, int ch,
+                               AVFrame *in, AVFrame *out)
+{
+    ADecorrelateContext *s = ctx->priv;
+    const double *src = (const double *)in->extended_data[ch];
+    double *dst = (double *)out->extended_data[ch];
+    const int nb_samples = in->nb_samples;
+    const int stages = s->stages;
+    APContext *ap0 = &s->ap[ch][0];
+
+    for (int n = 0; n < nb_samples; n++) {
+        dst[n] = ap_run(ap0, src[n]);
+        for (int i = 1; i < stages; i++) {
+            APContext *ap = &s->ap[ch][i];
+
+            dst[n] = ap_run(ap, dst[n]);
+        }
+    }
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    ADecorrelateContext *s = ctx->priv;
+    int ret;
+
+    if (s->seed == -1)
+        s->seed = av_get_random_seed();
+    av_lfg_init(&s->c, s->seed);
+
+    s->nb_channels = inlink->channels;
+    s->ap = av_calloc(inlink->channels, sizeof(*s->ap));
+    if (!s->ap)
+        return AVERROR(ENOMEM);
+
+    for (int i = 0; i < inlink->channels; i++) {
+        for (int j = 0; j < s->stages; j++) {
+            ret = ap_init(&s->ap[i][j], inlink->sample_rate,
+                          (double)av_lfg_get(&s->c) / 0xffffffff * 2.2917e-3 + 0.83333e-3);
+            if (ret < 0)
+                return ret;
+        }
+    }
+
+    s->filter_channel = filter_channel_dbl;
+
+    return 0;
+}
+
+typedef struct ThreadData {
+    AVFrame *in, *out;
+} ThreadData;
+
+static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
+{
+    ADecorrelateContext *s = ctx->priv;
+    ThreadData *td = arg;
+    AVFrame *out = td->out;
+    AVFrame *in = td->in;
+    const int start = (in->channels * jobnr) / nb_jobs;
+    const int end = (in->channels * (jobnr+1)) / nb_jobs;
+
+    for (int ch = start; ch < end; ch++)
+        s->filter_channel(ctx, ch, in, out);
+
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AVFilterLink *outlink = ctx->outputs[0];
+    AVFrame *out;
+    ThreadData td;
+
+    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);
+    }
+
+    td.in = in; td.out = out;
+    ff_filter_execute(ctx, filter_channels, &td, NULL,
+                      FFMIN(inlink->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)
+{
+    ADecorrelateContext *s = ctx->priv;
+
+    if (s->ap) {
+        for (int ch = 0; ch < s->nb_channels; ch++) {
+            for (int stage = 0; stage < s->stages; stage++)
+                ap_free(&s->ap[ch][stage]);
+        }
+    }
+
+    av_freep(&s->ap);
+}
+
+#define OFFSET(x) offsetof(ADecorrelateContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption adecorrelate_options[] = {
+    { "stages", "set filtering stages", OFFSET(stages), AV_OPT_TYPE_INT,    {.i64=6},   1, MAX_STAGES, FLAGS },
+    { "seed",   "set random seed",      OFFSET(seed),   AV_OPT_TYPE_INT64,  {.i64=-1}, -1,   UINT_MAX, FLAGS },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(adecorrelate);
+
+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_adecorrelate = {
+    .name            = "adecorrelate",
+    .description     = NULL_IF_CONFIG_SMALL("Apply decorrelation to input audio."),
+    .query_formats   = query_formats,
+    .priv_size       = sizeof(ADecorrelateContext),
+    .priv_class      = &adecorrelate_class,
+    .uninit          = uninit,
+    FILTER_INPUTS(inputs),
+    FILTER_OUTPUTS(outputs),
+    .flags           = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
+                       AVFILTER_FLAG_SLICE_THREADS,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 73040d2824..68bd3c432a 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -33,6 +33,7 @@ extern const AVFilter ff_af_acrossover;
 extern const AVFilter ff_af_acrusher;
 extern const AVFilter ff_af_adeclick;
 extern const AVFilter ff_af_adeclip;
+extern const AVFilter ff_af_adecorrelate;
 extern const AVFilter ff_af_adelay;
 extern const AVFilter ff_af_adenorm;
 extern const AVFilter ff_af_aderivative;
