diff --git a/doc/filters.texi b/doc/filters.texi
index cadf78c93c..54b85c4bb9 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -5370,6 +5370,49 @@ Set number of samples per each frame.
 Set window function to be used when generating FIR coefficients.
 @end table
 
+@section sinc
+
+Generate a sinc kaiser-windowed low-pass, high-pass, band-pass, or band-reject FIR coefficients.
+
+The resulting stream can be used with @ref{afir} filter for filtering the audio signal.
+
+The filter accepts the following options:
+
+@table @option
+@item sample_rate, r
+Set sample rate, default is 44100.
+
+@item nb_samples, n
+Set number of samples per each frame. Default is 1024.
+
+@item hp
+Set high-pass frequency. Default is 0.
+
+@item lp
+Set low-pass frequency. Default is 0.
+If high-pass frequency is lower than low-pass frequency and low-pass frequency
+is higher than 0 then filter will create band-pass filter coefficients,
+otherwise band-reject filter coefficients.
+
+@item phase
+Set filter phase response. Default is 50. Allowed range is from 0 to 100.
+
+@item beta
+Set Kaiser window beta.
+
+@item att
+Set stop-band attenuation. Default is 120dB, allowed range is from 40 to 180 dB.
+
+@item round
+Enable rounding, by default is disabled.
+
+@item hptaps
+Set number of taps for high-pass filter.
+
+@item lptaps
+Set number of taps for low-pass filter.
+@end table
+
 @section sine
 
 Generate an audio signal made of a sine wave with amplitude 1/8.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 62cc2f561f..b03b2457eb 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -141,6 +141,7 @@ OBJS-$(CONFIG_ANOISESRC_FILTER)              += asrc_anoisesrc.o
 OBJS-$(CONFIG_ANULLSRC_FILTER)               += asrc_anullsrc.o
 OBJS-$(CONFIG_FLITE_FILTER)                  += asrc_flite.o
 OBJS-$(CONFIG_HILBERT_FILTER)                += asrc_hilbert.o
+OBJS-$(CONFIG_SINC_FILTER)                   += asrc_sinc.o
 OBJS-$(CONFIG_SINE_FILTER)                   += asrc_sine.o
 
 OBJS-$(CONFIG_ANULLSINK_FILTER)              += asink_anullsink.o
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 5e72803b13..725bac94a0 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -134,6 +134,7 @@ extern AVFilter ff_asrc_anoisesrc;
 extern AVFilter ff_asrc_anullsrc;
 extern AVFilter ff_asrc_flite;
 extern AVFilter ff_asrc_hilbert;
+extern AVFilter ff_asrc_sinc;
 extern AVFilter ff_asrc_sine;
 
 extern AVFilter ff_asink_anullsink;
diff --git a/libavfilter/asrc_sinc.c b/libavfilter/asrc_sinc.c
new file mode 100644
index 0000000000..b0145cfaa1
--- /dev/null
+++ b/libavfilter/asrc_sinc.c
@@ -0,0 +1,455 @@
+/*
+ * Copyright (c) 2008-2009 Rob Sykes <robs@users.sourceforge.net>
+ * 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
+ */
+
+#include "libavutil/avassert.h"
+#include "libavutil/opt.h"
+
+#include "libavcodec/avfft.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+
+typedef struct SincContext {
+    const AVClass *class;
+
+    int sample_rate, nb_samples;
+    float att, beta, phase, Fc0, Fc1, tbw0, tbw1;
+    int num_taps[2];
+    int round;
+
+    int n, rdft_len;
+    float *coeffs;
+    int64_t pts;
+
+    RDFTContext *rdft, *irdft;
+} SincContext;
+
+static int request_frame(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    SincContext *s = ctx->priv;
+    const float *coeffs = s->coeffs;
+    AVFrame *frame = NULL;
+    int nb_samples;
+
+    nb_samples = FFMIN(s->nb_samples, s->n - s->pts);
+    if (nb_samples <= 0)
+        return AVERROR_EOF;
+
+    if (!(frame = ff_get_audio_buffer(outlink, nb_samples)))
+        return AVERROR(ENOMEM);
+
+    memcpy(frame->data[0], coeffs + s->pts, nb_samples * sizeof(float));
+
+    frame->pts = s->pts;
+    s->pts    += nb_samples;
+
+    return ff_filter_frame(outlink, frame);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    SincContext *s = ctx->priv;
+    static const int64_t chlayouts[] = { AV_CH_LAYOUT_MONO, -1 };
+    int sample_rates[] = { s->sample_rate, -1 };
+    static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLT,
+                                                       AV_SAMPLE_FMT_NONE };
+    AVFilterFormats *formats;
+    AVFilterChannelLayouts *layouts;
+    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 = avfilter_make_format64_list(chlayouts);
+    if (!layouts)
+        return AVERROR(ENOMEM);
+    ret = ff_set_common_channel_layouts(ctx, layouts);
+    if (ret < 0)
+        return ret;
+
+    formats = ff_make_format_list(sample_rates);
+    if (!formats)
+        return AVERROR(ENOMEM);
+    return ff_set_common_samplerates(ctx, formats);
+}
+
+static float bessel_I_0(float x)
+{
+    float term = 1, sum = 1, last_sum, x2 = x / 2;
+    int i = 1;
+
+    do {
+        float y = x2 / i++;
+
+        last_sum = sum;
+        sum += term *= y * y;
+    } while (sum != last_sum);
+
+    return sum;
+}
+
+static float *make_lpf(int num_taps, float Fc, float beta, float rho,
+                       float scale, int dc_norm)
+{
+    int i, m = num_taps - 1;
+    float *h = av_calloc(num_taps, sizeof(*h)), sum = 0;
+    float mult = scale / bessel_I_0(beta), mult1 = 1.f / (.5f * m + rho);
+
+    av_assert0(Fc >= 0 && Fc <= 1);
+
+    for (i = 0; i <= m / 2; i++) {
+        float z = i - .5f * m, x = z * M_PI, y = z * mult1;
+        h[i] = x ? sinf(Fc * x) / x : Fc;
+        sum += h[i] *= bessel_I_0(beta * sqrtf(1.f - y * y)) * mult;
+        if (m - i != i) {
+            h[m - i] = h[i];
+            sum += h[i];
+        }
+    }
+
+    for (i = 0; dc_norm && i < num_taps; i++)
+        h[i] *= scale / sum;
+
+    return h;
+}
+
+static float kaiser_beta(float att, float tr_bw)
+{
+    if (att >= 60.f) {
+        static const float coefs[][4] = {
+            {-6.784957e-10, 1.02856e-05, 0.1087556, -0.8988365 + .001},
+            {-6.897885e-10, 1.027433e-05, 0.10876, -0.8994658 + .002},
+            {-1.000683e-09, 1.030092e-05, 0.1087677, -0.9007898 + .003},
+            {-3.654474e-10, 1.040631e-05, 0.1087085, -0.8977766 + .006},
+            {8.106988e-09, 6.983091e-06, 0.1091387, -0.9172048 + .015},
+            {9.519571e-09, 7.272678e-06, 0.1090068, -0.9140768 + .025},
+            {-5.626821e-09, 1.342186e-05, 0.1083999, -0.9065452 + .05},
+            {-9.965946e-08, 5.073548e-05, 0.1040967, -0.7672778 + .085},
+            {1.604808e-07, -5.856462e-05, 0.1185998, -1.34824 + .1},
+            {-1.511964e-07, 6.363034e-05, 0.1064627, -0.9876665 + .18},
+        };
+        float realm = logf(tr_bw / .0005f) / logf(2.f);
+        float const *c0 = coefs[av_clip((int)realm, 0, FF_ARRAY_ELEMS(coefs) - 1)];
+        float const *c1 = coefs[av_clip(1 + (int)realm, 0, FF_ARRAY_ELEMS(coefs) - 1)];
+        float b0 = ((c0[0] * att + c0[1]) * att + c0[2]) * att + c0[3];
+        float b1 = ((c1[0] * att + c1[1]) * att + c1[2]) * att + c1[3];
+
+        return b0 + (b1 - b0) * (realm - (int)realm);
+    }
+    if (att > 50.f)
+        return .1102f * (att - 8.7f);
+    if (att > 20.96f)
+        return .58417f * powf(att - 20.96f, .4f) + .07886f * (att - 20.96f);
+    return 0;
+}
+
+static void kaiser_params(float att, float Fc, float tr_bw, float *beta, int *num_taps)
+{
+    *beta = *beta < 0.f ? kaiser_beta(att, tr_bw * .5f / Fc): *beta;
+    att = att < 60.f ? (att - 7.95f) / (2.285f * M_PI * 2.f) :
+        ((.0007528358f-1.577737e-05 * *beta) * *beta + 0.6248022f) * *beta + .06186902f;
+    *num_taps = !*num_taps ? ceilf(att/tr_bw + 1) : *num_taps;
+}
+
+static float *lpf(float Fn, float Fc, float tbw, int *num_taps, float att, float *beta, int round)
+{
+    int n = *num_taps;
+
+    if ((Fc /= Fn) <= 0.f || Fc >= 1.f) {
+        *num_taps = 0;
+        return NULL;
+    }
+
+    att = att ? att : 120.f;
+
+    kaiser_params(att, Fc, (tbw ? tbw / Fn : .05f) * .5f, beta, num_taps);
+
+    if (!n) {
+        n = *num_taps;
+        *num_taps = av_clip(n, 11, 32767);
+        if (round)
+            *num_taps = 1 + 2 * (int)((int)((*num_taps / 2) * Fc + .5f) / Fc + .5f);
+    }
+
+    return make_lpf(*num_taps |= 1, Fc, *beta, 0.f, 1.f, 0);
+}
+
+static void invert(float *h, int n)
+{
+    for (int i = 0; i < n; i++)
+        h[i] = -h[i];
+
+    h[(n - 1) / 2] += 1;
+}
+
+#define PACK(h, n)   h[1] = h[n]
+#define UNPACK(h, n) h[n] = h[1], h[n + 1] = h[1] = 0;
+#define SQR(a) ((a) * (a))
+
+static float safe_log(float x)
+{
+    av_assert0(x >= 0);
+    if (x)
+        return logf(x);
+    return -26;
+}
+
+static int fir_to_phase(SincContext *s, float **h, int *len, int *post_len, float phase)
+{
+    float *pi_wraps, *work, phase1 = (phase > 50.f ? 100.f - phase : phase) / 50.f;
+    int i, work_len, begin, end, imp_peak = 0, peak = 0;
+    float imp_sum = 0, peak_imp_sum = 0;
+    float prev_angle2 = 0, cum_2pi = 0, prev_angle1 = 0, cum_1pi = 0;
+
+    for (i = *len, work_len = 2 * 2 * 8; i > 1; work_len <<= 1, i >>= 1);
+
+    work = av_calloc(work_len + 2, sizeof(*work));    /* +2: (UN)PACK */
+    pi_wraps = av_calloc(((work_len + 2) / 2), sizeof(*pi_wraps));
+    if (!work || !pi_wraps)
+        return AVERROR(ENOMEM);
+
+    memcpy(work, *h, *len * sizeof(*work));
+
+    av_rdft_end(s->rdft);
+    av_rdft_end(s->irdft);
+    s->rdft = s->irdft = NULL;
+    s->rdft  = av_rdft_init(av_log2(work_len), DFT_R2C);
+    s->irdft = av_rdft_init(av_log2(work_len), IDFT_C2R);
+    if (!s->rdft || !s->irdft)
+        return AVERROR(ENOMEM);
+
+    av_rdft_calc(s->rdft, work);   /* Cepstral: */
+    UNPACK(work, work_len);
+
+    for (i = 0; i <= work_len; i += 2) {
+        float angle = atan2f(work[i + 1], work[i]);
+        float detect = 2 * M_PI;
+        float delta = angle - prev_angle2;
+        float adjust = detect * ((delta < -detect * .7f) - (delta > detect * .7f));
+
+        prev_angle2 = angle;
+        cum_2pi += adjust;
+        angle += cum_2pi;
+        detect = M_PI;
+        delta = angle - prev_angle1;
+        adjust = detect * ((delta < -detect * .7f) - (delta > detect * .7f));
+        prev_angle1 = angle;
+        cum_1pi += fabsf(adjust);        /* fabs for when 2pi and 1pi have combined */
+        pi_wraps[i >> 1] = cum_1pi;
+
+        work[i] = safe_log(sqrtf(SQR(work[i]) + SQR(work[i + 1])));
+        work[i + 1] = 0;
+    }
+
+    PACK(work, work_len);
+    av_rdft_calc(s->irdft, work);
+
+    for (i = 0; i < work_len; i++)
+        work[i] *= 2.f / work_len;
+
+    for (i = 1; i < work_len / 2; i++) {        /* Window to reject acausal components */
+        work[i] *= 2;
+        work[i + work_len / 2] = 0;
+    }
+    av_rdft_calc(s->rdft, work);
+
+    for (i = 2; i < work_len; i += 2)   /* Interpolate between linear & min phase */
+        work[i + 1] = phase1 * i / work_len * pi_wraps[work_len >> 1] + (1 - phase1) * (work[i + 1] + pi_wraps[i >> 1]) - pi_wraps[i >> 1];
+
+    work[0] = exp(work[0]);
+    work[1] = exp(work[1]);
+    for (i = 2; i < work_len; i += 2) {
+        float x = expf(work[i]);
+
+        work[i    ] = x * cosf(work[i + 1]);
+        work[i + 1] = x * sinf(work[i + 1]);
+    }
+
+    av_rdft_calc(s->irdft, work);
+    for (i = 0; i < work_len; i++)
+        work[i] *= 2.f / work_len;
+
+    /* Find peak pos. */
+    for (i = 0; i <= (int) (pi_wraps[work_len >> 1] / M_PI + .5f); i++) {
+        imp_sum += work[i];
+        if (fabs(imp_sum) > fabs(peak_imp_sum)) {
+            peak_imp_sum = imp_sum;
+            peak = i;
+        }
+        if (work[i] > work[imp_peak])   /* For debug check only */
+            imp_peak = i;
+    }
+
+    while (peak && fabsf(work[peak - 1]) > fabsf(work[peak]) && (work[peak - 1] * work[peak] > 0)) {
+        peak--;
+    }
+
+    if (!phase1) {
+        begin = 0;
+    } else if (phase1 == 1) {
+        begin = peak - *len / 2;
+    } else {
+        begin = (.997f - (2 - phase1) * .22f) * *len + .5f;
+        end = (.997f + (0 - phase1) * .22f) * *len + .5f;
+        begin = peak - (begin & ~3);
+        end = peak + 1 + ((end + 3) & ~3);
+        *len = end - begin;
+        *h = av_realloc(*h, *len * sizeof(**h));
+        if (!*h) {
+            av_free(pi_wraps);
+            av_free(work);
+            return AVERROR(ENOMEM);
+        }
+    }
+
+    for (i = 0; i < *len; i++) {
+        (*h)[i] = work[(begin + (phase > 50.f ? *len - 1 - i : i) + work_len) & (work_len - 1)];
+    }
+    *post_len = phase > 50 ? peak - begin : begin + *len - (peak + 1);
+
+    av_log(s, AV_LOG_DEBUG, "%d nPI=%g peak-sum@%i=%g (val@%i=%g); len=%i post=%i (%g%%)\n",
+           work_len, pi_wraps[work_len >> 1] / M_PI, peak, peak_imp_sum, imp_peak,
+           work[imp_peak], *len, *post_len, 100.f - 100.f * *post_len / (*len - 1));
+
+    av_free(pi_wraps);
+    av_free(work);
+
+    return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    SincContext *s = ctx->priv;
+    float Fn = s->sample_rate * .5f;
+    float *h[2];
+    int i, n, post_peak, longer;
+
+    outlink->sample_rate = s->sample_rate;
+    s->pts = 0;
+
+    if (s->Fc0 >= Fn || s->Fc1 >= Fn) {
+        av_log(ctx, AV_LOG_ERROR,
+               "filter frequency must be less than %d/2.\n", s->sample_rate);
+        return AVERROR(EINVAL);
+    }
+
+    h[0] = lpf(Fn, s->Fc0, s->tbw0, &s->num_taps[0], s->att, &s->beta, s->round);
+    h[1] = lpf(Fn, s->Fc1, s->tbw1, &s->num_taps[1], s->att, &s->beta, s->round);
+
+    if (h[0])
+        invert(h[0], s->num_taps[0]);
+
+    longer = s->num_taps[1] > s->num_taps[0];
+    n = s->num_taps[longer];
+
+    if (h[0] && h[1]) {
+        for (i = 0; i < s->num_taps[!longer]; i++)
+            h[longer][i + (n - s->num_taps[!longer]) / 2] += h[!longer][i];
+
+        if (s->Fc0 < s->Fc1)
+            invert(h[longer], n);
+
+        av_free(h[!longer]);
+    }
+
+    if (s->phase != 50.f) {
+        int ret = fir_to_phase(s, &h[longer], &n, &post_peak, s->phase);
+        if (ret < 0)
+            return ret;
+    } else {
+        post_peak = n >> 1;
+    }
+
+    s->n = 1 << (av_log2(n) + 1);
+    s->rdft_len = 1 << av_log2(n);
+    s->coeffs = av_calloc(s->n, sizeof(*s->coeffs));
+    if (!s->coeffs)
+        return AVERROR(ENOMEM);
+
+    for (i = 0; i < n; i++)
+        s->coeffs[i] = h[longer][i];
+    av_free(h[longer]);
+
+    av_rdft_end(s->rdft);
+    av_rdft_end(s->irdft);
+    s->rdft = s->irdft = NULL;
+
+    return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    SincContext *s = ctx->priv;
+
+    av_freep(&s->coeffs);
+    av_rdft_end(s->rdft);
+    av_rdft_end(s->irdft);
+    s->rdft = s->irdft = NULL;
+}
+
+static const AVFilterPad sinc_outputs[] = {
+    {
+        .name          = "default",
+        .type          = AVMEDIA_TYPE_AUDIO,
+        .config_props  = config_output,
+        .request_frame = request_frame,
+    },
+    { NULL }
+};
+
+#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+#define OFFSET(x) offsetof(SincContext, x)
+
+static const AVOption sinc_options[] = {
+    { "sample_rate", "set sample rate",                               OFFSET(sample_rate), AV_OPT_TYPE_INT,   {.i64=44100},  1, INT_MAX, AF },
+    { "r",           "set sample rate",                               OFFSET(sample_rate), AV_OPT_TYPE_INT,   {.i64=44100},  1, INT_MAX, AF },
+    { "nb_samples",  "set the number of samples per requested frame", OFFSET(nb_samples),  AV_OPT_TYPE_INT,   {.i64=1024},   1, INT_MAX, AF },
+    { "n",           "set the number of samples per requested frame", OFFSET(nb_samples),  AV_OPT_TYPE_INT,   {.i64=1024},   1, INT_MAX, AF },
+    { "hp",          "set high-pass filter frequency",                OFFSET(Fc0),         AV_OPT_TYPE_FLOAT, {.dbl=0},      0, INT_MAX, AF },
+    { "lp",          "set low-pass filter frequency",                 OFFSET(Fc1),         AV_OPT_TYPE_FLOAT, {.dbl=0},      0, INT_MAX, AF },
+    { "phase",       "set filter phase response",                     OFFSET(phase),       AV_OPT_TYPE_FLOAT, {.dbl=50},     0,     100, AF },
+    { "beta",        "set kaiser window beta",                        OFFSET(beta),        AV_OPT_TYPE_FLOAT, {.dbl=-1},    -1,     256, AF },
+    { "att",         "set stop-band attenuation",                     OFFSET(att),         AV_OPT_TYPE_FLOAT, {.dbl=120},   40,     180, AF },
+    { "round",       "enable rounding",                               OFFSET(round),       AV_OPT_TYPE_BOOL,  {.i64=0},      0,       1, AF },
+    { "hptaps",      "set number of taps for high-pass filter",       OFFSET(num_taps[0]), AV_OPT_TYPE_INT,   {.i64=0},      0,   32768, AF },
+    { "lptaps",      "set number of taps for low-pass filter",        OFFSET(num_taps[1]), AV_OPT_TYPE_INT,   {.i64=0},      0,   32768, AF },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(sinc);
+
+AVFilter ff_asrc_sinc = {
+    .name          = "sinc",
+    .description   = NULL_IF_CONFIG_SMALL("Generate a sinc kaiser-windowed low-pass, high-pass, band-pass, or band-reject FIR coefficients."),
+    .priv_size     = sizeof(SincContext),
+    .priv_class    = &sinc_class,
+    .query_formats = query_formats,
+    .uninit        = uninit,
+    .inputs        = NULL,
+    .outputs       = sinc_outputs,
+};
