diff --git a/doc/filters.texi b/doc/filters.texi
index f651f1234d..ff911ad92e 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1059,6 +1059,20 @@ the reduction.
 Default is @code{average}. Can be @code{average} or @code{maximum}.
 @end table
 
+@section aiir
+
+Apply an arbitrary Infinite Impulse Response filter.
+
+It accepts the following parameters:
+
+@table @option
+@item a
+Set denominator coefficients.
+
+@item b
+Set nominator coefficients.
+@end table
+
 @section alimiter
 
 The limiter prevents an input signal from rising over a desired threshold.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 8bde542163..1fe58ed3d2 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -43,6 +43,7 @@ OBJS-$(CONFIG_AFFTFILT_FILTER)               += af_afftfilt.o
 OBJS-$(CONFIG_AFIR_FILTER)                   += af_afir.o
 OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o
 OBJS-$(CONFIG_AGATE_FILTER)                  += af_agate.o
+OBJS-$(CONFIG_AIIR_FILTER)                   += af_aiir.o
 OBJS-$(CONFIG_AINTERLEAVE_FILTER)            += f_interleave.o
 OBJS-$(CONFIG_ALIMITER_FILTER)               += af_alimiter.o
 OBJS-$(CONFIG_ALLPASS_FILTER)                += af_biquads.o
diff --git a/libavfilter/af_aiir.c b/libavfilter/af_aiir.c
new file mode 100644
index 0000000000..d1be9afa5e
--- /dev/null
+++ b/libavfilter/af_aiir.c
@@ -0,0 +1,232 @@
+/*
+ * 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/avassert.h"
+#include "libavutil/avstring.h"
+#include "libavutil/opt.h"
+#include "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+
+typedef struct AudioIIRContext {
+    const AVClass *class;
+    char *a_str, *b_str;
+
+    int nb_a, nb_b;
+    double *a, *b;
+    AVFrame *input, *output;
+} AudioIIRContext;
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats;
+    AVFilterChannelLayouts *layouts;
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_DBLP,
+        AV_SAMPLE_FMT_NONE
+    };
+    int 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_make_format_list(sample_fmts);
+    if (!formats)
+        return AVERROR(ENOMEM);
+    ret = ff_set_common_formats(ctx, formats);
+    if (ret < 0)
+        return ret;
+
+    formats = ff_all_samplerates();
+    if (!formats)
+        return AVERROR(ENOMEM);
+    return ff_set_common_samplerates(ctx, formats);
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    AudioIIRContext *s = ctx->priv;
+    AVFilterLink *inlink = ctx->inputs[0];
+
+    s->input  = ff_get_audio_buffer(inlink, s->nb_b);
+    s->output = ff_get_audio_buffer(inlink, s->nb_a);
+    if (!s->input || !s->output)
+            return AVERROR(ENOMEM);
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext  *ctx = inlink->dst;
+    AudioIIRContext *s     = ctx->priv;
+    AVFilterLink *outlink = ctx->outputs[0];
+    AVFrame *out;
+    int ch, n;
+
+    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);
+    }
+
+    for (ch = 0; ch < out->channels; ch++) {
+        const double *src = (const double *)in->extended_data[ch];
+        double *ic = (double *)s->input->extended_data[ch];
+        double *oc = (double *)s->output->extended_data[ch];
+        double *dst = (double *)out->extended_data[ch];
+        const double *a = s->a;
+        const double *b = s->b;
+
+        for (n = 0; n < in->nb_samples; n++) {
+            double sample = 0.;
+            int x;
+
+            memmove(&ic[1], &ic[0], (s->nb_b - 1) * sizeof(*ic));
+            memmove(&oc[1], &oc[0], (s->nb_a - 1) * sizeof(*oc));
+            ic[0] = src[n];
+            for (x = 0; x < s->nb_b; x++)
+                sample += b[x] * ic[x];
+
+            for (x = 1; x < s->nb_a; x++)
+                sample -= a[x] * oc[x];
+
+            oc[0] = dst[n] = sample;
+        }
+    }
+
+    if (in != out)
+        av_frame_free(&in);
+
+    return ff_filter_frame(outlink, out);
+}
+
+static void count_items(char *item_str, int *nb_items)
+{
+    char *p;
+
+    *nb_items = 1;
+    for (p = item_str; *p; p++) {
+        if (*p == ' ' || *p == '|')
+            (*nb_items)++;
+    }
+}
+
+static int read_items(char *item_str, int nb_items, double *dst)
+{
+    char *p, *arg, *saveptr = NULL;
+    int i;
+
+    p = item_str;
+    for (i = 0; i < nb_items; i++) {
+        if (!(arg = av_strtok(p, " |", &saveptr)))
+            break;
+
+        p = NULL;
+        sscanf(arg, "%lf", &dst[i]);
+    }
+
+    return 0;
+}
+
+static av_cold int init(AVFilterContext *ctx)
+{
+    AudioIIRContext *s = ctx->priv;
+    int i;
+
+    count_items(s->a_str, &s->nb_a);
+    count_items(s->b_str, &s->nb_b);
+
+    s->a = av_calloc(s->nb_a, sizeof(*s->a));
+    s->b = av_calloc(s->nb_b, sizeof(*s->b));
+    if (!s->a || !s->b)
+        return AVERROR(ENOMEM);
+
+    read_items(s->a_str, s->nb_a, s->a);
+    read_items(s->b_str, s->nb_b, s->b);
+
+    for (i = 1; i < s->nb_a; i++)
+        s->a[i] /= s->a[0];
+
+    for (i = 0; i < s->nb_b; i++)
+        s->b[i] /= s->a[0];
+
+    return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    AudioIIRContext *s = ctx->priv;
+
+    av_freep(&s->a);
+    av_freep(&s->b);
+    av_frame_free(&s->input);
+    av_frame_free(&s->output);
+}
+
+static const AVFilterPad inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .filter_frame = filter_frame,
+    },
+    { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .config_props = config_output,
+    },
+    { NULL }
+};
+
+#define OFFSET(x) offsetof(AudioIIRContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption aiir_options[] = {
+    { "a", "set A coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, .flags = FLAGS },
+    { "b", "set B coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, .flags = FLAGS },
+    { NULL },
+};
+
+AVFILTER_DEFINE_CLASS(aiir);
+
+AVFilter ff_af_aiir = {
+    .name          = "aiir",
+    .description   = NULL_IF_CONFIG_SMALL("Apply Infinite Impulse Response filter with supplied coefficients."),
+    .priv_size     = sizeof(AudioIIRContext),
+    .init          = init,
+    .uninit        = uninit,
+    .query_formats = query_formats,
+    .inputs        = inputs,
+    .outputs       = outputs,
+    .priv_class    = &aiir_class,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 67c073091f..705c03c22c 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -54,6 +54,7 @@ static void register_all(void)
     REGISTER_FILTER(AFIR,           afir,           af);
     REGISTER_FILTER(AFORMAT,        aformat,        af);
     REGISTER_FILTER(AGATE,          agate,          af);
+    REGISTER_FILTER(AIIR,           aiir,           af);
     REGISTER_FILTER(AINTERLEAVE,    ainterleave,    af);
     REGISTER_FILTER(ALIMITER,       alimiter,       af);
     REGISTER_FILTER(ALLPASS,        allpass,        af);
