diff --git a/configure b/configure
index b3cb5b0..7fc7af4 100755
--- a/configure
+++ b/configure
@@ -3078,6 +3078,8 @@ unix_protocol_select="network"
 # filters
 afftfilt_filter_deps="avcodec"
 afftfilt_filter_select="fft"
+afirfilter_filter_deps="avcodec"
+afirfilter_filter_select="fft"
 amovie_filter_deps="avcodec avformat"
 aresample_filter_deps="swresample"
 ass_filter_deps="libass"
diff --git a/doc/filters.texi b/doc/filters.texi
index 119e747..ea343d1 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -878,6 +878,16 @@ afftfilt="1-clip((b/nb)*b,0,1)"
 @end example
 @end itemize
 
+@section afirfilter
+
+Apply an Arbitary Frequency Impulse Response filter.
+
+This filter uses second stream as FIR coefficients.
+If second stream holds single channel, it will be used
+for all input channels in first stream, otherwise
+number of channels in second stream must be same as
+number of channels in first stream.
+
 @anchor{aformat}
 @section aformat
 
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 66c36e4..1a0f24b 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -38,6 +38,7 @@ OBJS-$(CONFIG_AEMPHASIS_FILTER)              += af_aemphasis.o
 OBJS-$(CONFIG_AEVAL_FILTER)                  += aeval.o
 OBJS-$(CONFIG_AFADE_FILTER)                  += af_afade.o
 OBJS-$(CONFIG_AFFTFILT_FILTER)               += af_afftfilt.o window_func.o
+OBJS-$(CONFIG_AFIRFILTER_FILTER)             += af_afirfilter.o
 OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o
 OBJS-$(CONFIG_AGATE_FILTER)                  += af_agate.o
 OBJS-$(CONFIG_AINTERLEAVE_FILTER)            += f_interleave.o
diff --git a/libavfilter/af_afirfilter.c b/libavfilter/af_afirfilter.c
new file mode 100644
index 0000000..ef2488a
--- /dev/null
+++ b/libavfilter/af_afirfilter.c
@@ -0,0 +1,409 @@
+/*
+ * Copyright (c) 2017 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
+ */
+
+/**
+ * @file
+ * An arbitrary audio FIR filter
+ */
+
+#include "libavutil/audio_fifo.h"
+#include "libavutil/avassert.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/common.h"
+#include "libavutil/opt.h"
+#include "libavcodec/avfft.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "formats.h"
+#include "internal.h"
+
+typedef struct FIRContext {
+    const AVClass *class;
+
+    int n;
+    int eof_coeffs;
+    int have_coeffs;
+    int nb_taps;
+    int fft_length;
+    int nb_channels;
+    int one2many;
+
+    FFTContext *fft, *ifft;
+    FFTComplex **fft_data;
+    FFTComplex **fft_coef;
+
+    AVAudioFifo *fifo[2];
+    AVFrame *in[2];
+    AVFrame *buffer;
+    int64_t pts;
+    int hop_size;
+    int start, end;
+} FIRContext;
+
+static int fir_filter(FIRContext *s, AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    int start = s->start, end = s->end;
+    int ret = 0, n, ch, j, k;
+    int nb_samples;
+    AVFrame *out;
+
+    nb_samples = FFMIN(s->fft_length, av_audio_fifo_size(s->fifo[0]));
+
+    s->in[0] = ff_get_audio_buffer(ctx->inputs[0], nb_samples);
+    if (!s->in[0])
+        return AVERROR(ENOMEM);
+
+    av_audio_fifo_peek(s->fifo[0], (void **)s->in[0]->extended_data, nb_samples);
+
+    for (ch = 0; ch < outlink->channels; ch++) {
+        const float *src = (float *)s->in[0]->extended_data[ch];
+        float *buf = (float *)s->buffer->extended_data[ch];
+        FFTComplex *fft_data = s->fft_data[ch];
+        FFTComplex *fft_coef = s->fft_coef[ch];
+
+        memset(fft_data, 0, sizeof(*fft_data) * s->fft_length);
+        for (n = 0; n < nb_samples; n++) {
+            fft_data[n].re = src[n];
+            fft_data[n].im = 0;
+        }
+
+        av_fft_permute(s->fft, fft_data);
+        av_fft_calc(s->fft, fft_data);
+
+        fft_data[0].re *= fft_coef[0].re;
+        fft_data[0].im *= fft_coef[0].im;
+        for (n = 1; n < s->fft_length; n++) {
+            const float re = fft_data[n].re;
+            const float im = fft_data[n].im;
+
+            fft_data[n].re = re * fft_coef[n].re - im * fft_coef[n].im;
+            fft_data[n].im = re * fft_coef[n].im + im * fft_coef[n].re;
+        }
+
+        av_fft_permute(s->ifft, fft_data);
+        av_fft_calc(s->ifft, fft_data);
+
+        start = s->start;
+        end = s->end;
+        k = end;
+
+        for (n = 0, j = start; j < k && n < s->fft_length; n++, j++) {
+            buf[j] = fft_data[n].re;
+        }
+
+        for (; n < s->fft_length; n++, j++) {
+            buf[j] = fft_data[n].re;
+        }
+
+        start += s->hop_size;
+        end = j;
+    }
+
+    s->start = start;
+    s->end   = end;
+
+    if (start >= nb_samples) {
+        float *dst, *buf;
+
+        start -= nb_samples;
+        end   -= nb_samples;
+
+        s->start = start;
+        s->end = end;
+
+        out = ff_get_audio_buffer(outlink, nb_samples);
+        if (!out)
+            return AVERROR(ENOMEM);
+
+        out->pts = s->pts;
+        s->pts += nb_samples;
+
+        for (ch = 0; ch < s->nb_channels; ch++) {
+            dst = (float *)out->extended_data[ch];
+            buf = (float *)s->buffer->extended_data[ch];
+
+            for (n = 0; n < nb_samples; n++)
+                dst[n] = buf[n];
+            memmove(buf, buf + nb_samples, nb_samples * 4);
+        }
+
+        ret = ff_filter_frame(outlink, out);
+    }
+
+    av_audio_fifo_drain(s->fifo[0], FFMIN(nb_samples, s->hop_size));
+    av_frame_free(&s->in[0]);
+
+    return ret;
+}
+
+static int convert_coeffs(AVFilterContext *ctx)
+{
+    FIRContext *s = ctx->priv;
+    int ch, n;
+
+    s->nb_taps = av_audio_fifo_size(s->fifo[1]);
+    if (s->nb_taps > 32768) {
+        av_log(ctx, AV_LOG_ERROR, "Too big number of taps: %d > 32768.\n", s->nb_taps);
+        return AVERROR(EINVAL);
+    }
+
+    for (n = 1; (1 << n) < s->nb_taps; n++);
+    s->n = n + 2;
+    s->fft_length = 1 << s->n;
+
+    for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
+        s->fft_data[ch] = av_calloc(s->fft_length, sizeof(**s->fft_data));
+        if (!s->fft_data[ch])
+            return AVERROR(ENOMEM);
+    }
+
+    for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
+        s->fft_coef[ch] = av_calloc(s->fft_length, sizeof(**s->fft_coef));
+        if (!s->fft_coef[ch])
+            return AVERROR(ENOMEM);
+    }
+
+    s->hop_size = s->fft_length - s->nb_taps + 1;
+    if (s->hop_size <= 0) {
+        av_log(ctx, AV_LOG_ERROR, "Too big number of taps.\n");
+        return AVERROR(EINVAL);
+    }
+
+    s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->fft_length * 2);
+    if (!s->buffer)
+        return AVERROR(ENOMEM);
+
+    s->fft  = av_fft_init(s->n, 0);
+    s->ifft = av_fft_init(s->n, 1);
+    if (!s->fft || !s->ifft)
+        return AVERROR(ENOMEM);
+
+    s->in[1] = ff_get_audio_buffer(ctx->inputs[1], s->nb_taps);
+    if (!s->in[1])
+        return AVERROR(ENOMEM);
+
+    av_audio_fifo_read(s->fifo[1], (void **)s->in[1]->extended_data, s->nb_taps);
+    for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
+        FFTComplex *fft_coef = s->fft_coef[ch];
+        const float *re = (const float *)s->in[1]->extended_data[!s->one2many * ch];
+        const float scale = 1.f / s->fft_length;
+        const int offset = (s->fft_length - s->nb_taps);
+
+        memset(fft_coef, 0, sizeof(*fft_coef) * s->fft_length);
+        for (n = 0; n < s->nb_taps; n++) {
+            fft_coef[n + offset].re = re[n] * scale;
+        }
+        av_fft_permute(s->fft, fft_coef);
+        av_fft_calc(s->fft, fft_coef);
+    }
+
+    av_frame_free(&s->in[1]);
+    s->have_coeffs = 1;
+
+    return 0;
+}
+
+static int read_coeffs(AVFilterLink *link, AVFrame *frame)
+{
+    AVFilterContext *ctx = link->dst;
+    FIRContext *s = ctx->priv;
+
+    av_audio_fifo_write(s->fifo[1], (void **)frame->extended_data,
+                        frame->nb_samples);
+    av_frame_free(&frame);
+
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *link, AVFrame *frame)
+{
+    AVFilterContext *ctx = link->dst;
+    FIRContext *s = ctx->priv;
+    AVFilterLink *outlink = ctx->outputs[0];
+    int ret = 0;
+
+    av_audio_fifo_write(s->fifo[0], (void **)frame->extended_data,
+                        frame->nb_samples);
+    av_frame_free(&frame);
+
+    if (!s->have_coeffs && s->eof_coeffs) {
+        ret = convert_coeffs(ctx);
+        if (ret < 0)
+            return ret;
+    }
+
+    if (s->have_coeffs) {
+        while (av_audio_fifo_size(s->fifo[0]) >= s->fft_length) {
+            ret = fir_filter(s, outlink);
+            if (ret < 0)
+                break;
+        }
+    }
+    return ret;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    FIRContext *s = ctx->priv;
+    int ret;
+
+    if (!s->eof_coeffs) {
+        ret = ff_request_frame(ctx->inputs[1]);
+        if (ret == AVERROR_EOF) {
+            s->eof_coeffs = 1;
+            ret = 0;
+        }
+        return ret;
+    }
+    ret = ff_request_frame(ctx->inputs[0]);
+    if (ret == AVERROR_EOF && s->have_coeffs) {
+        while (av_audio_fifo_size(s->fifo[0]) > 0) {
+            ret = fir_filter(s, outlink);
+            if (ret < 0)
+                return ret;
+        }
+        ret = AVERROR_EOF;
+    }
+    return ret;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats;
+    AVFilterChannelLayouts *layouts = NULL;
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_FLTP,
+        AV_SAMPLE_FMT_NONE
+    };
+    int ret, i;
+
+    layouts = ff_all_channel_counts();
+    if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
+        return ret;
+
+    for (i = 0; i < 2; i++) {
+        layouts = ff_all_channel_counts();
+        if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0)
+            return ret;
+    }
+
+    formats = ff_make_format_list(sample_fmts);
+    if ((ret = ff_set_common_formats(ctx, formats)) < 0)
+        return ret;
+
+    formats = ff_all_samplerates();
+    return ff_set_common_samplerates(ctx, formats);
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    FIRContext *s = ctx->priv;
+
+    if (ctx->inputs[0]->channels != ctx->inputs[1]->channels &&
+        ctx->inputs[1]->channels != 1) {
+        av_log(ctx, AV_LOG_ERROR,
+               "Second input must have same number of channels as first input or "
+               "exactly 1 channel.\n");
+        return AVERROR(EINVAL);
+    }
+
+    s->one2many = ctx->inputs[1]->channels == 1;
+    outlink->sample_rate = ctx->inputs[0]->sample_rate;
+    outlink->time_base   = ctx->inputs[0]->time_base;
+    outlink->channel_layout = ctx->inputs[0]->channel_layout;
+    outlink->channels = ctx->inputs[0]->channels;
+
+    s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024);
+    s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024);
+    if (!s->fifo[0] || !s->fifo[1])
+        return AVERROR(ENOMEM);
+
+    s->fft_data = av_calloc(outlink->channels, sizeof(*s->fft_data));
+    s->fft_coef = av_calloc(ctx->inputs[1]->channels, sizeof(*s->fft_coef));
+    if (!s->fft_data || !s->fft_coef)
+        return AVERROR(ENOMEM);
+    s->nb_channels = outlink->channels;
+
+    return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    FIRContext *s = ctx->priv;
+    int ch;
+
+    for (ch = 0; ch < s->nb_channels; ch++) {
+        if (s->fft_data)
+            av_freep(&s->fft_data[ch]);
+    }
+    av_freep(&s->fft_data);
+
+    for (ch = 0; ch < s->nb_channels; ch++) {
+        if (s->fft_coef)
+            av_freep(&s->fft_coef[ch]);
+    }
+    av_freep(&s->fft_coef);
+
+    av_fft_end(s->fft);
+    av_fft_end(s->ifft);
+
+    av_frame_free(&s->in[0]);
+    av_frame_free(&s->in[1]);
+
+    av_audio_fifo_free(s->fifo[0]);
+    av_audio_fifo_free(s->fifo[1]);
+}
+
+static const AVFilterPad afirfilter_inputs[] = {
+    {
+        .name           = "main",
+        .type           = AVMEDIA_TYPE_AUDIO,
+        .filter_frame   = filter_frame,
+    },{
+        .name           = "coefficients",
+        .type           = AVMEDIA_TYPE_AUDIO,
+        .filter_frame   = read_coeffs,
+    },
+    { NULL }
+};
+
+static const AVFilterPad afirfilter_outputs[] = {
+    {
+        .name          = "default",
+        .type          = AVMEDIA_TYPE_AUDIO,
+        .config_props  = config_output,
+        .request_frame = request_frame,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_afirfilter = {
+    .name           = "afirfilter",
+    .description    = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
+    .priv_size      = sizeof(FIRContext),
+    .query_formats  = query_formats,
+    .uninit         = uninit,
+    .inputs         = afirfilter_inputs,
+    .outputs        = afirfilter_outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 8fb87eb..8bfe1ae 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -50,6 +50,7 @@ static void register_all(void)
     REGISTER_FILTER(AEVAL,          aeval,          af);
     REGISTER_FILTER(AFADE,          afade,          af);
     REGISTER_FILTER(AFFTFILT,       afftfilt,       af);
+    REGISTER_FILTER(AFIRFILTER,     afirfilter,     af);
     REGISTER_FILTER(AFORMAT,        aformat,        af);
     REGISTER_FILTER(AGATE,          agate,          af);
     REGISTER_FILTER(AINTERLEAVE,    ainterleave,    af);
