diff --git a/doc/filters.texi b/doc/filters.texi
index 2d9af46a6b..8cca5d008a 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -3045,6 +3045,42 @@ 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
 
+@section deesser
+
+Apply de-essing to the audio samples.
+
+@table @option
+@item i
+Set intensity for triggering de-essing. Allowed range is from 0 to 1.
+Default is 0.
+
+@item m
+Set amount of ducking on treble part of sound. Allowed range is from 0 to 1.
+Default is 0.5.
+
+@item f
+How much of original frequency content to keep when de-essing. Allowed range is from 0 to 1.
+Default is 0.5.
+
+@item s
+Set the output mode.
+
+It accepts the following values:
+@table @option
+@item i
+Pass input unchanged.
+
+@item o
+Pass ess filtered out.
+
+@item e
+Pass only ess.
+
+Default value is @var{o}.
+@end table
+
+@end table
+
 @section drmeter
 Measure audio dynamic range.
 
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 07ea8d7edc..455c809b15 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -101,6 +101,7 @@ 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-$(CONFIG_DEESSER_FILTER)                += af_deesser.o
 OBJS-$(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_deesser.c b/libavfilter/af_deesser.c
new file mode 100644
index 0000000000..1775dc7d4c
--- /dev/null
+++ b/libavfilter/af_deesser.c
@@ -0,0 +1,240 @@
+/*
+ * Copyright (c) 2018 Chris Johnson
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "formats.h"
+
+typedef struct DeesserChannel {
+    double s1, s2, s3;
+    double m1, m2;
+    double ratioA, ratioB;
+    double iirSampleA, iirSampleB;
+    int flip;
+} DeesserChannel;
+
+typedef struct DeesserContext {
+    const AVClass *class;
+
+    double intensity;
+    double max;
+    double frequency;
+    int    mode;
+
+    DeesserChannel *chan;
+} DeesserContext;
+
+enum OutModes {
+    IN_MODE,
+    OUT_MODE,
+    ESS_MODE,
+    NB_MODES
+};
+
+#define OFFSET(x) offsetof(DeesserContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption deesser_options[] = {
+    { "i", "set intensity",    OFFSET(intensity), AV_OPT_TYPE_DOUBLE, {.dbl=0.0}, 0.0, 1.0, A },
+    { "m", "set max deessing", OFFSET(max),       AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.0, 1.0, A },
+    { "f", "set frequency",    OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.0, 1.0, A },
+    { "s", "set output mode",  OFFSET(mode),      AV_OPT_TYPE_INT,    {.i64=OUT_MODE}, 0, NB_MODES-1, A, "mode" },
+    {  "i", "input",           0,                 AV_OPT_TYPE_CONST,  {.i64=IN_MODE},  0, 0, A, "mode" },
+    {  "o", "output",          0,                 AV_OPT_TYPE_CONST,  {.i64=OUT_MODE}, 0, 0, A, "mode" },
+    {  "e", "ess",             0,                 AV_OPT_TYPE_CONST,  {.i64=ESS_MODE}, 0, 0, A, "mode" },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(deesser);
+
+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;
+    DeesserContext *s = ctx->priv;
+
+    s->chan = av_calloc(inlink->channels, sizeof(*s->chan));
+    if (!s->chan)
+        return AVERROR(ENOMEM);
+
+    for (int i = 0; i < inlink->channels; i++) {
+        DeesserChannel *chan = &s->chan[i];;
+
+        chan->ratioA = chan->ratioB = 1.0;
+    }
+
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AVFilterLink *outlink = ctx->outputs[0];
+    DeesserContext *s = ctx->priv;
+    AVFrame *out;
+
+    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 (int ch = 0; ch < inlink->channels; ch++) {
+        DeesserChannel *dec = &s->chan[ch];
+        double *src = (double *)in->extended_data[ch];
+        double *dst = (double *)out->extended_data[ch];
+        double overallscale = (1.0 / 44100.0) * inlink->sample_rate;
+        double intensity = pow(s->intensity, 5) * (8192 / overallscale);
+        double maxdess = 1.0 / pow(10.0,((s->max-1.0)*48.0) / 20);
+        double iirAmount = pow(s->frequency, 2) / overallscale;
+        double offset;
+        double sense;
+        double recovery;
+        double attackspeed;
+
+        for (int i = 0; i < in->nb_samples; i++) {
+            double sample = src[i];
+
+            dec->s3 = dec->s2;
+            dec->s2 = dec->s1;
+            dec->s1 = sample;
+            dec->m1 = (dec->s1 - dec->s2) * ((dec->s1 - dec->s2) / 1.3);
+            dec->m2 = (dec->s2 - dec->s3) * ((dec->s1 - dec->s2) / 1.3);
+            sense = fabs((dec->m1 - dec->m2) * ((dec->m1 - dec->m2) / 1.3));
+            attackspeed = 7.0 + sense * 1024;
+
+            sense = 1.0 + intensity * intensity * sense;
+            sense = FFMIN(sense, intensity);
+            recovery = 1.0 + (0.01 / sense);
+
+            offset = 1.0 - fabs(sample);
+
+            if (dec->flip) {
+                dec->iirSampleA = (dec->iirSampleA * (1.0 - (offset * iirAmount))) +
+                                  (sample * (offset * iirAmount));
+                if (dec->ratioA < sense) {
+                    dec->ratioA = ((dec->ratioA * attackspeed) + sense) / (attackspeed + 1.0);
+                } else {
+                    dec->ratioA = 1.0 + ((dec->ratioA - 1.0) / recovery);
+                }
+
+                dec->ratioA = FFMIN(dec->ratioA, maxdess);
+                sample = dec->iirSampleA + ((sample - dec->iirSampleA) / dec->ratioA);
+            } else {
+                dec->iirSampleB = (dec->iirSampleB * (1.0 - (offset * iirAmount))) +
+                                  (sample * (offset * iirAmount));
+                if (dec->ratioB < sense) {
+                    dec->ratioB = ((dec->ratioB * attackspeed) + sense) / (attackspeed + 1.0);
+                } else {
+                    dec->ratioB = 1.0 + ((dec->ratioB - 1.0) / recovery);
+                }
+
+                dec->ratioB = FFMIN(dec->ratioB, maxdess);
+                sample = dec->iirSampleB + ((sample - dec->iirSampleB) / dec->ratioB);
+            }
+
+            dec->flip = !dec->flip;
+
+            switch (s->mode) {
+            case IN_MODE:  dst[i] = src[i]; break;
+            case OUT_MODE: dst[i] = sample; break;
+            case ESS_MODE: dst[i] = src[i] - sample; break;
+            }
+        }
+    }
+
+    if (out != in)
+        av_frame_free(&in);
+
+    return ff_filter_frame(outlink, out);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    DeesserContext *s = ctx->priv;
+
+    av_freep(&s->chan);
+}
+
+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,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_deesser = {
+    .name           = "deesser",
+    .description    = NULL_IF_CONFIG_SMALL("Apply de-essing to the audio."),
+    .query_formats  = query_formats,
+    .priv_size      = sizeof(DeesserContext),
+    .priv_class     = &deesser_class,
+    .uninit         = uninit,
+    .inputs         = inputs,
+    .outputs        = outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 9c846b1ddd..04a3df7d56 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -93,6 +93,7 @@ extern AVFilter ff_af_compensationdelay;
 extern AVFilter ff_af_crossfeed;
 extern AVFilter ff_af_crystalizer;
 extern AVFilter ff_af_dcshift;
+extern AVFilter ff_af_deesser;
 extern AVFilter ff_af_drmeter;
 extern AVFilter ff_af_dynaudnorm;
 extern AVFilter ff_af_earwax;
