diff --git a/Changelog b/Changelog
index cf0adc90e4..cd91f63cb3 100644
--- a/Changelog
+++ b/Changelog
@@ -19,6 +19,7 @@ version <next>:
 - surround audio filter
 - sofalizer filter switched to libmysofa
 - Gremlin Digital Video demuxer and decoder
+- headphone audio filter
 
 version 3.3:
 - CrystalHD decoder moved to new decode API
diff --git a/doc/filters.texi b/doc/filters.texi
index 9cc356b4df..023096f4e0 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2789,6 +2789,49 @@ Samples where the target gain does not match between channels
 @end table
 @end table
 
+@section headphone
+
+Apply head-related transfer functions (HRTFs) to create virtual
+loudspeakers around the user for binaural listening via headphones.
+The HRIRs are provided via additional streams, for each channel
+one stereo input stream is needed.
+
+The filter accepts the following options:
+
+@table @option
+@item map
+Set mapping of input streams for convolution.
+The argument is a '|'-separated list of channel names in order as they
+are given as additional stream inputs for filter.
+This also specify number of input streams. Number of input streams
+must be not less than number of channels in first stream plus one.
+
+@item gain
+Set gain applied to audio. Value is in dB. Default is 0.
+
+@item type
+Set processing type. Can be @var{time} or @var{freq}. @var{time} is
+processing audio in time domain which is slow.
+@var{freq} is processing audio in frequency domain which is fast.
+Default is @var{freq}.
+
+@item lfe
+Set custom gain for LFE channels. Value is in dB. Default is 0.
+@end table
+
+@subsection Examples
+
+@itemize
+@item
+Full example using wav files as coefficients with amovie filters for 7.1 downmix,
+each amovie filter use stereo file with IR coefficients as input.
+The files give coefficients for each position of virtual loudspeaker:
+@example
+ffmpeg -i input.wav -lavfi-complex "amovie=azi_270_ele_0_DFC.wav[sr],amovie=azi_90_ele_0_DFC.wav[sl],amovie=azi_225_ele_0_DFC.wav[br],amovie=azi_135_ele_0_DFC.wav[bl],amovie=azi_0_ele_0_DFC.wav,asplit[fc][lfe],amovie=azi_35_ele_0_DFC.wav[fl],amovie=azi_325_ele_0_DFC.wav[fr],[a:0][fl][fr][fc][lfe][bl][br][sl][sr]headphone=FL|FR|FC|LFE|BL|BR|SL|SR"
+output.wav
+@end example
+@end itemize
+
 @section highpass
 
 Apply a high-pass filter with 3dB point frequency.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index c88dfb3264..04ec9b8b8f 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -92,6 +92,7 @@ OBJS-$(CONFIG_EXTRASTEREO_FILTER)            += af_extrastereo.o
 OBJS-$(CONFIG_FIREQUALIZER_FILTER)           += af_firequalizer.o
 OBJS-$(CONFIG_FLANGER_FILTER)                += af_flanger.o generate_wave_table.o
 OBJS-$(CONFIG_HDCD_FILTER)                   += af_hdcd.o
+OBJS-$(CONFIG_HEADPHONE_FILTER)              += af_headphone.o
 OBJS-$(CONFIG_HIGHPASS_FILTER)               += af_biquads.o
 OBJS-$(CONFIG_JOIN_FILTER)                   += af_join.o
 OBJS-$(CONFIG_LADSPA_FILTER)                 += af_ladspa.o
diff --git a/libavfilter/af_headphone.c b/libavfilter/af_headphone.c
new file mode 100644
index 0000000000..3dd5a0c396
--- /dev/null
+++ b/libavfilter/af_headphone.c
@@ -0,0 +1,811 @@
+/*
+ * Copyright (C) 2017 Paul B Mahol
+ * Copyright (C) 2013-2015 Andreas Fuchs, Wolfgang Hrauda
+ * 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 <math.h>
+
+#include "libavutil/audio_fifo.h"
+#include "libavutil/avstring.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/float_dsp.h"
+#include "libavutil/intmath.h"
+#include "libavutil/opt.h"
+#include "libavcodec/avfft.h"
+
+#include "avfilter.h"
+#include "internal.h"
+#include "audio.h"
+
+#define TIME_DOMAIN      0
+#define FREQUENCY_DOMAIN 1
+
+typedef struct HeadphoneContext {
+    const AVClass *class;
+
+    char *map;
+    int type;
+
+    int lfe_channel;
+
+    int have_hrirs;
+    int eof_hrirs;
+    int64_t pts;
+
+    int ir_len;
+
+    int mapping[64];
+
+    int nb_inputs;
+
+    int nb_irs;
+
+    float gain;
+    float lfe_gain, gain_lfe;
+
+    float *ringbuffer[2];
+    int write[2];
+
+    int buffer_length;
+    int n_fft;
+    int size;
+
+    int *delay[2];
+    float *data_ir[2];
+    float *temp_src[2];
+    FFTComplex *temp_fft[2];
+
+    FFTContext *fft[2], *ifft[2];
+    FFTComplex *data_hrtf[2];
+
+    AVFloatDSPContext *fdsp;
+    struct headphone_inputs {
+        AVAudioFifo *fifo;
+        AVFrame     *frame;
+        int          ir_len;
+        int          delay_l;
+        int          delay_r;
+        int          eof;
+    } *in;
+} HeadphoneContext;
+
+static int parse_channel_name(HeadphoneContext *s, int x, char **arg, int *rchannel, char *buf)
+{
+    int len, i, channel_id = 0;
+    int64_t layout, layout0;
+
+    if (sscanf(*arg, "%7[A-Z]%n", buf, &len)) {
+        layout0 = layout = av_get_channel_layout(buf);
+        if (layout == AV_CH_LOW_FREQUENCY)
+            s->lfe_channel = x;
+        for (i = 32; i > 0; i >>= 1) {
+            if (layout >= 1LL << i) {
+                channel_id += i;
+                layout >>= i;
+            }
+        }
+        if (channel_id >= 64 || layout0 != 1LL << channel_id)
+            return AVERROR(EINVAL);
+        *rchannel = channel_id;
+        *arg += len;
+        return 0;
+    }
+    return AVERROR(EINVAL);
+}
+
+static void parse_map(AVFilterContext *ctx)
+{
+    HeadphoneContext *s = ctx->priv;
+    char *arg, *tokenizer, *p, *args = av_strdup(s->map);
+    int i;
+
+    if (!args)
+        return;
+    p = args;
+
+    s->lfe_channel = -1;
+    s->nb_inputs = 1;
+
+    for (i = 0; i < 64; i++) {
+        s->mapping[i] = -1;
+    }
+
+    while ((arg = av_strtok(p, "|", &tokenizer))) {
+        int out_ch_id;
+        char buf[8];
+
+        p = NULL;
+        if (parse_channel_name(s, s->nb_inputs - 1, &arg, &out_ch_id, buf)) {
+            av_log(ctx, AV_LOG_WARNING, "Failed to parse \'%s\' as channel name.\n", buf);
+            continue;
+        }
+        s->mapping[s->nb_inputs - 1] = out_ch_id;
+        s->nb_inputs++;
+    }
+    s->nb_irs = s->nb_inputs - 1;
+
+    av_free(args);
+}
+
+typedef struct ThreadData {
+    AVFrame *in, *out;
+    int *write;
+    int **delay;
+    float **ir;
+    int *n_clippings;
+    float **ringbuffer;
+    float **temp_src;
+    FFTComplex **temp_fft;
+} ThreadData;
+
+static int headphone_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
+{
+    HeadphoneContext *s = ctx->priv;
+    ThreadData *td = arg;
+    AVFrame *in = td->in, *out = td->out;
+    int offset = jobnr;
+    int *write = &td->write[jobnr];
+    const int *const delay = td->delay[jobnr];
+    const float *const ir = td->ir[jobnr];
+    int *n_clippings = &td->n_clippings[jobnr];
+    float *ringbuffer = td->ringbuffer[jobnr];
+    float *temp_src = td->temp_src[jobnr];
+    const int ir_len = s->ir_len;
+    const float *src = (const float *)in->data[0];
+    float *dst = (float *)out->data[0];
+    const int in_channels = in->channels;
+    const int buffer_length = s->buffer_length;
+    const uint32_t modulo = (uint32_t)buffer_length - 1;
+    float *buffer[16];
+    int wr = *write;
+    int read;
+    int i, l;
+
+    dst += offset;
+    for (l = 0; l < in_channels; l++) {
+        buffer[l] = ringbuffer + l * buffer_length;
+    }
+
+    for (i = 0; i < in->nb_samples; i++) {
+        const float *temp_ir = ir;
+
+        *dst = 0;
+        for (l = 0; l < in_channels; l++) {
+            *(buffer[l] + wr) = src[l];
+        }
+
+        for (l = 0; l < in_channels; l++) {
+            const float *const bptr = buffer[l];
+
+            if (l == s->lfe_channel) {
+                *dst += *(buffer[s->lfe_channel] + wr) * s->gain_lfe;
+                temp_ir += FFALIGN(ir_len, 16);
+                continue;
+            }
+
+            read = (wr - *(delay + l) - (ir_len - 1) + buffer_length) & modulo;
+
+            if (read + ir_len < buffer_length) {
+                memcpy(temp_src, bptr + read, ir_len * sizeof(*temp_src));
+            } else {
+                int len = FFMIN(ir_len - (read % ir_len), buffer_length - read);
+
+                memcpy(temp_src, bptr + read, len * sizeof(*temp_src));
+                memcpy(temp_src + len, bptr, (ir_len - len) * sizeof(*temp_src));
+            }
+
+            dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, ir_len);
+            temp_ir += FFALIGN(ir_len, 16);
+        }
+
+        if (fabs(*dst) > 1)
+            *n_clippings += 1;
+
+        dst += 2;
+        src += in_channels;
+        wr   = (wr + 1) & modulo;
+    }
+
+    *write = wr;
+
+    return 0;
+}
+
+static int headphone_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
+{
+    HeadphoneContext *s = ctx->priv;
+    ThreadData *td = arg;
+    AVFrame *in = td->in, *out = td->out;
+    int offset = jobnr;
+    int *write = &td->write[jobnr];
+    FFTComplex *hrtf = s->data_hrtf[jobnr];
+    int *n_clippings = &td->n_clippings[jobnr];
+    float *ringbuffer = td->ringbuffer[jobnr];
+    const int ir_len = s->ir_len;
+    const float *src = (const float *)in->data[0];
+    float *dst = (float *)out->data[0];
+    const int in_channels = in->channels;
+    const int buffer_length = s->buffer_length;
+    const uint32_t modulo = (uint32_t)buffer_length - 1;
+    FFTComplex *fft_in = s->temp_fft[jobnr];
+    FFTContext *ifft = s->ifft[jobnr];
+    FFTContext *fft = s->fft[jobnr];
+    const int n_fft = s->n_fft;
+    const float fft_scale = 1.0f / s->n_fft;
+    FFTComplex *hrtf_offset;
+    int wr = *write;
+    int n_read;
+    int i, j;
+
+    dst += offset;
+
+    n_read = FFMIN(s->ir_len, in->nb_samples);
+    for (j = 0; j < n_read; j++) {
+        dst[2 * j]     = ringbuffer[wr];
+        ringbuffer[wr] = 0.0;
+        wr  = (wr + 1) & modulo;
+    }
+
+    for (j = n_read; j < in->nb_samples; j++) {
+        dst[2 * j] = 0;
+    }
+
+    for (i = 0; i < in_channels; i++) {
+        if (i == s->lfe_channel) {
+            for (j = 0; j < in->nb_samples; j++) {
+                dst[2 * j] += src[i + j * in_channels] * s->gain_lfe;
+            }
+            continue;
+        }
+
+        offset = i * n_fft;
+        hrtf_offset = hrtf + offset;
+
+        memset(fft_in, 0, sizeof(FFTComplex) * n_fft);
+
+        for (j = 0; j < in->nb_samples; j++) {
+            fft_in[j].re = src[j * in_channels + i];
+        }
+
+        av_fft_permute(fft, fft_in);
+        av_fft_calc(fft, fft_in);
+        for (j = 0; j < n_fft; j++) {
+            const FFTComplex *hcomplex = hrtf_offset + j;
+            const float re = fft_in[j].re;
+            const float im = fft_in[j].im;
+
+            fft_in[j].re = re * hcomplex->re - im * hcomplex->im;
+            fft_in[j].im = re * hcomplex->im + im * hcomplex->re;
+        }
+
+        av_fft_permute(ifft, fft_in);
+        av_fft_calc(ifft, fft_in);
+
+        for (j = 0; j < in->nb_samples; j++) {
+            dst[2 * j] += fft_in[j].re * fft_scale;
+        }
+
+        for (j = 0; j < ir_len - 1; j++) {
+            int write_pos = (wr + j) & modulo;
+
+            *(ringbuffer + write_pos) += fft_in[in->nb_samples + j].re * fft_scale;
+        }
+    }
+
+    for (i = 0; i < out->nb_samples; i++) {
+        if (fabs(*dst) > 1) {
+            n_clippings[0]++;
+        }
+
+        dst += 2;
+    }
+
+    *write = wr;
+
+    return 0;
+}
+
+static int read_ir(AVFilterLink *inlink, AVFrame *frame)
+{
+    AVFilterContext *ctx = inlink->dst;
+    HeadphoneContext *s = ctx->priv;
+    int ir_len, max_ir_len, input_number;
+
+    for (input_number = 0; input_number < s->nb_inputs; input_number++)
+        if (inlink == ctx->inputs[input_number])
+            break;
+
+    av_audio_fifo_write(s->in[input_number].fifo, (void **)frame->extended_data,
+                        frame->nb_samples);
+    av_frame_free(&frame);
+
+    ir_len = av_audio_fifo_size(s->in[input_number].fifo);
+    max_ir_len = 4096;
+    if (ir_len > max_ir_len) {
+        av_log(ctx, AV_LOG_ERROR, "Too big length of IRs: %d > %d.\n", ir_len, max_ir_len);
+        return AVERROR(EINVAL);
+    }
+    s->in[input_number].ir_len = ir_len;
+    s->ir_len = FFMAX(ir_len, s->ir_len);
+
+    return 0;
+}
+
+static int headphone_frame(HeadphoneContext *s, AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    AVFrame *in = s->in[0].frame;
+    int n_clippings[2] = { 0 };
+    ThreadData td;
+    AVFrame *out;
+
+    av_audio_fifo_read(s->in[0].fifo, (void **)in->extended_data, s->size);
+
+    out = ff_get_audio_buffer(outlink, in->nb_samples);
+    if (!out) {
+        av_frame_free(&in);
+        return AVERROR(ENOMEM);
+    }
+    out->pts = s->pts;
+    if (s->pts != AV_NOPTS_VALUE)
+        s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
+
+    td.in = in; td.out = out; td.write = s->write;
+    td.delay = s->delay; td.ir = s->data_ir; td.n_clippings = n_clippings;
+    td.ringbuffer = s->ringbuffer; td.temp_src = s->temp_src;
+    td.temp_fft = s->temp_fft;
+
+    if (s->type == TIME_DOMAIN) {
+        ctx->internal->execute(ctx, headphone_convolute, &td, NULL, 2);
+    } else {
+        ctx->internal->execute(ctx, headphone_fast_convolute, &td, NULL, 2);
+    }
+    emms_c();
+
+    if (n_clippings[0] + n_clippings[1] > 0) {
+        av_log(ctx, AV_LOG_WARNING, "%d of %d samples clipped. Please reduce gain.\n",
+               n_clippings[0] + n_clippings[1], out->nb_samples * 2);
+    }
+
+    return ff_filter_frame(outlink, out);
+}
+
+static int convert_coeffs(AVFilterContext *ctx, AVFilterLink *inlink)
+{
+    struct HeadphoneContext *s = ctx->priv;
+    const int ir_len = s->ir_len;
+    int nb_irs = s->nb_irs;
+    int nb_input_channels = ctx->inputs[0]->channels;
+    float gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10);
+    FFTComplex *data_hrtf_l = NULL;
+    FFTComplex *data_hrtf_r = NULL;
+    FFTComplex *fft_in_l = NULL;
+    FFTComplex *fft_in_r = NULL;
+    float *data_ir_l = NULL;
+    float *data_ir_r = NULL;
+    int offset = 0;
+    int n_fft;
+    int i, j;
+
+    s->buffer_length = 1 << (32 - ff_clz(s->ir_len));
+    s->n_fft = n_fft = 1 << (32 - ff_clz(s->ir_len + inlink->sample_rate));
+
+    if (s->type == FREQUENCY_DOMAIN) {
+        fft_in_l = av_calloc(n_fft, sizeof(*fft_in_l));
+        fft_in_r = av_calloc(n_fft, sizeof(*fft_in_r));
+        if (!fft_in_l || !fft_in_r) {
+            return AVERROR(ENOMEM);
+        }
+
+        av_fft_end(s->fft[0]);
+        av_fft_end(s->fft[1]);
+        s->fft[0] = av_fft_init(log2(s->n_fft), 0);
+        s->fft[1] = av_fft_init(log2(s->n_fft), 0);
+        av_fft_end(s->ifft[0]);
+        av_fft_end(s->ifft[1]);
+        s->ifft[0] = av_fft_init(log2(s->n_fft), 1);
+        s->ifft[1] = av_fft_init(log2(s->n_fft), 1);
+
+        if (!s->fft[0] || !s->fft[1] || !s->ifft[0] || !s->ifft[1]) {
+            av_log(ctx, AV_LOG_ERROR, "Unable to create FFT contexts of size %d.\n", s->n_fft);
+            return AVERROR(ENOMEM);
+        }
+    }
+
+    s->data_ir[0] = av_calloc(FFALIGN(s->ir_len, 16), sizeof(float) * s->nb_irs);
+    s->data_ir[1] = av_calloc(FFALIGN(s->ir_len, 16), sizeof(float) * s->nb_irs);
+    s->delay[0] = av_malloc_array(s->nb_irs, sizeof(float));
+    s->delay[1] = av_malloc_array(s->nb_irs, sizeof(float));
+
+    if (s->type == TIME_DOMAIN) {
+        s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);
+        s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);
+    } else {
+        s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float));
+        s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float));
+        s->temp_fft[0] = av_malloc_array(s->n_fft, sizeof(FFTComplex));
+        s->temp_fft[1] = av_malloc_array(s->n_fft, sizeof(FFTComplex));
+        if (!s->temp_fft[0] || !s->temp_fft[1])
+            return AVERROR(ENOMEM);
+    }
+
+    if (!s->data_ir[0] || !s->data_ir[1] ||
+        !s->ringbuffer[0] || !s->ringbuffer[1])
+        return AVERROR(ENOMEM);
+
+    s->in[0].frame = ff_get_audio_buffer(ctx->inputs[0], s->size);
+    if (!s->in[0].frame)
+        return AVERROR(ENOMEM);
+    for (i = 0; i < s->nb_irs; i++) {
+        s->in[i + 1].frame = ff_get_audio_buffer(ctx->inputs[i + 1], s->ir_len);
+        if (!s->in[i + 1].frame)
+            return AVERROR(ENOMEM);
+    }
+
+    if (s->type == TIME_DOMAIN) {
+        s->temp_src[0] = av_calloc(FFALIGN(ir_len, 16), sizeof(float));
+        s->temp_src[1] = av_calloc(FFALIGN(ir_len, 16), sizeof(float));
+
+        data_ir_l = av_calloc(nb_irs * FFALIGN(ir_len, 16), sizeof(*data_ir_l));
+        data_ir_r = av_calloc(nb_irs * FFALIGN(ir_len, 16), sizeof(*data_ir_r));
+        if (!data_ir_r || !data_ir_l || !s->temp_src[0] || !s->temp_src[1]) {
+            av_free(data_ir_l);
+            av_free(data_ir_r);
+            return AVERROR(ENOMEM);
+        }
+    } else {
+        data_hrtf_l = av_malloc_array(n_fft, sizeof(*data_hrtf_l) * nb_irs);
+        data_hrtf_r = av_malloc_array(n_fft, sizeof(*data_hrtf_r) * nb_irs);
+        if (!data_hrtf_r || !data_hrtf_l) {
+            av_free(data_hrtf_l);
+            av_free(data_hrtf_r);
+            return AVERROR(ENOMEM);
+        }
+    }
+
+    for (i = 0; i < s->nb_irs; i++) {
+        int len = s->in[i + 1].ir_len;
+        int delay_l = s->in[i + 1].delay_l;
+        int delay_r = s->in[i + 1].delay_r;
+        int idx = -1;
+        float *ptr;
+
+        for (j = 0; j < inlink->channels; j++) {
+            if (s->mapping[i] < 0) {
+                continue;
+            }
+
+            if ((av_channel_layout_extract_channel(inlink->channel_layout, j)) == (1LL << s->mapping[i])) {
+                idx = j;
+                break;
+            }
+        }
+        if (idx == -1)
+            continue;
+
+        av_audio_fifo_read(s->in[i + 1].fifo, (void **)s->in[i + 1].frame->extended_data, len);
+        ptr = (float *)s->in[i + 1].frame->extended_data[0];
+
+        if (s->type == TIME_DOMAIN) {
+            offset = idx * FFALIGN(len, 16);
+            for (j = 0; j < len; j++) {
+                data_ir_l[offset + j] = ptr[len * 2 - j * 2 - 2] * gain_lin;
+                data_ir_r[offset + j] = ptr[len * 2 - j * 2 - 1] * gain_lin;
+            }
+        } else {
+            memset(fft_in_l, 0, n_fft * sizeof(*fft_in_l));
+            memset(fft_in_r, 0, n_fft * sizeof(*fft_in_r));
+
+            offset = idx * n_fft;
+            for (j = 0; j < len; j++) {
+                fft_in_l[delay_l + j].re = ptr[j * 2    ] * gain_lin;
+                fft_in_r[delay_r + j].re = ptr[j * 2 + 1] * gain_lin;
+            }
+
+            av_fft_permute(s->fft[0], fft_in_l);
+            av_fft_calc(s->fft[0], fft_in_l);
+            memcpy(data_hrtf_l + offset, fft_in_l, n_fft * sizeof(*fft_in_l));
+            av_fft_permute(s->fft[0], fft_in_r);
+            av_fft_calc(s->fft[0], fft_in_r);
+            memcpy(data_hrtf_r + offset, fft_in_r, n_fft * sizeof(*fft_in_r));
+        }
+    }
+
+    if (s->type == TIME_DOMAIN) {
+        memcpy(s->data_ir[0], data_ir_l, sizeof(float) * nb_irs * FFALIGN(ir_len, 16));
+        memcpy(s->data_ir[1], data_ir_r, sizeof(float) * nb_irs * FFALIGN(ir_len, 16));
+
+        av_freep(&data_ir_l);
+        av_freep(&data_ir_r);
+    } else {
+        s->data_hrtf[0] = av_malloc_array(n_fft * s->nb_irs, sizeof(FFTComplex));
+        s->data_hrtf[1] = av_malloc_array(n_fft * s->nb_irs, sizeof(FFTComplex));
+        if (!s->data_hrtf[0] || !s->data_hrtf[1]) {
+            av_freep(&data_hrtf_l);
+            av_freep(&data_hrtf_r);
+            av_freep(&fft_in_l);
+            av_freep(&fft_in_r);
+            return AVERROR(ENOMEM);
+        }
+
+        memcpy(s->data_hrtf[0], data_hrtf_l,
+            sizeof(FFTComplex) * nb_irs * n_fft);
+        memcpy(s->data_hrtf[1], data_hrtf_r,
+            sizeof(FFTComplex) * nb_irs * n_fft);
+
+        av_freep(&data_hrtf_l);
+        av_freep(&data_hrtf_r);
+
+        av_freep(&fft_in_l);
+        av_freep(&fft_in_r);
+    }
+
+    s->have_hrirs = 1;
+
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    HeadphoneContext *s = ctx->priv;
+    AVFilterLink *outlink = ctx->outputs[0];
+    int ret = 0;
+
+    av_audio_fifo_write(s->in[0].fifo, (void **)in->extended_data,
+                        in->nb_samples);
+    if (s->pts == AV_NOPTS_VALUE)
+        s->pts = in->pts;
+
+    av_frame_free(&in);
+
+    if (!s->have_hrirs && s->eof_hrirs) {
+        ret = convert_coeffs(ctx, inlink);
+        if (ret < 0)
+            return ret;
+    }
+
+    if (s->have_hrirs) {
+        while (av_audio_fifo_size(s->in[0].fifo) >= s->size) {
+            ret = headphone_frame(s, outlink);
+            if (ret < 0)
+                break;
+        }
+    }
+    return ret;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    struct HeadphoneContext *s = ctx->priv;
+    AVFilterFormats *formats = NULL;
+    AVFilterChannelLayouts *layouts = NULL;
+    int ret, i;
+
+    ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT);
+    if (ret)
+        return ret;
+    ret = ff_set_common_formats(ctx, formats);
+    if (ret)
+        return ret;
+
+    layouts = ff_all_channel_layouts();
+    if (!layouts)
+        return AVERROR(ENOMEM);
+
+    ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts);
+    if (ret)
+        return ret;
+
+    layouts = NULL;
+    ret = ff_add_channel_layout(&layouts, AV_CH_LAYOUT_STEREO);
+    if (ret)
+        return ret;
+
+    for (i = 1; i < s->nb_inputs; i++) {
+        ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts);
+        if (ret)
+            return ret;
+    }
+
+    ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts);
+    if (ret)
+        return ret;
+
+    formats = ff_all_samplerates();
+    if (!formats)
+        return AVERROR(ENOMEM);
+    return ff_set_common_samplerates(ctx, formats);
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    HeadphoneContext *s = ctx->priv;
+
+    if (s->type == FREQUENCY_DOMAIN) {
+        inlink->partial_buf_size =
+        inlink->min_samples =
+        inlink->max_samples = inlink->sample_rate;
+    }
+
+    if (s->nb_irs < inlink->channels) {
+        av_log(ctx, AV_LOG_ERROR, "Number of inputs must be >= %d.\n", inlink->channels + 1);
+        return AVERROR(EINVAL);
+    }
+
+    return 0;
+}
+
+static av_cold int init(AVFilterContext *ctx)
+{
+    HeadphoneContext *s = ctx->priv;
+    int i;
+
+    AVFilterPad pad = {
+        .name         = "in0",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .config_props = config_input,
+        .filter_frame = filter_frame,
+    };
+    ff_insert_inpad(ctx, 0, &pad);
+
+    if (!s->map) {
+        av_log(ctx, AV_LOG_ERROR, "Valid mapping must be set.\n");
+        return AVERROR(EINVAL);
+    }
+
+    parse_map(ctx);
+
+    s->in = av_calloc(s->nb_inputs, sizeof(*s->in));
+    if (!s->in)
+        return AVERROR(ENOMEM);
+
+    for (i = 1; i < s->nb_inputs; i++) {
+        char *name = av_asprintf("hrir%d", i - 1);
+        AVFilterPad pad = {
+            .name         = name,
+            .type         = AVMEDIA_TYPE_AUDIO,
+            .filter_frame = read_ir,
+        };
+        if (!name)
+            return AVERROR(ENOMEM);
+        ff_insert_inpad(ctx, i, &pad);
+    }
+
+    s->fdsp = avpriv_float_dsp_alloc(0);
+    if (!s->fdsp)
+        return AVERROR(ENOMEM);
+    s->pts = AV_NOPTS_VALUE;
+
+    return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    HeadphoneContext *s = ctx->priv;
+    AVFilterLink *inlink = ctx->inputs[0];
+    int i;
+
+    if (s->type == TIME_DOMAIN)
+        s->size = 1024;
+    else
+        s->size = inlink->sample_rate;
+
+    for (i = 0; i < s->nb_inputs; i++) {
+        s->in[i].fifo = av_audio_fifo_alloc(ctx->inputs[i]->format, ctx->inputs[i]->channels, 1024);
+        if (!s->in[i].fifo)
+            return AVERROR(ENOMEM);
+    }
+    s->gain_lfe = expf((s->gain - 3 * inlink->channels - 6 + s->lfe_gain) / 20 * M_LN10);
+
+    return 0;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    HeadphoneContext *s = ctx->priv;
+    int i, ret;
+
+    for (i = 1; !s->eof_hrirs && i < s->nb_inputs; i++) {
+        if (!s->in[i].eof) {
+            ret = ff_request_frame(ctx->inputs[i]);
+            if (ret == AVERROR_EOF) {
+                s->in[i].eof = 1;
+                ret = 0;
+            }
+            return ret;
+        } else {
+            if (i == s->nb_inputs - 1)
+                s->eof_hrirs = 1;
+        }
+    }
+    return ff_request_frame(ctx->inputs[0]);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    HeadphoneContext *s = ctx->priv;
+    int i;
+
+    av_fft_end(s->ifft[0]);
+    av_fft_end(s->ifft[1]);
+    av_fft_end(s->fft[0]);
+    av_fft_end(s->fft[1]);
+    av_freep(&s->delay[0]);
+    av_freep(&s->delay[1]);
+    av_freep(&s->data_ir[0]);
+    av_freep(&s->data_ir[1]);
+    av_freep(&s->ringbuffer[0]);
+    av_freep(&s->ringbuffer[1]);
+    av_freep(&s->temp_src[0]);
+    av_freep(&s->temp_src[1]);
+    av_freep(&s->temp_fft[0]);
+    av_freep(&s->temp_fft[1]);
+    av_freep(&s->data_hrtf[0]);
+    av_freep(&s->data_hrtf[1]);
+    av_freep(&s->fdsp);
+
+    for (i = 0; i < s->nb_inputs; i++) {
+        av_frame_free(&s->in[i].frame);
+        av_audio_fifo_free(s->in[i].fifo);
+        if (ctx->input_pads && i)
+            av_freep(&ctx->input_pads[i].name);
+    }
+    av_freep(&s->in);
+}
+
+#define OFFSET(x) offsetof(HeadphoneContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption headphone_options[] = {
+    { "map",       "set channels convolution mappings",  OFFSET(map),      AV_OPT_TYPE_STRING, {.str=NULL},            .flags = FLAGS },
+    { "gain",      "set gain in dB",                     OFFSET(gain),     AV_OPT_TYPE_FLOAT,  {.dbl=0},     -20,  40, .flags = FLAGS },
+    { "lfe",       "set lfe gain in dB",                 OFFSET(lfe_gain), AV_OPT_TYPE_FLOAT,  {.dbl=0},     -20,  40, .flags = FLAGS },
+    { "type",      "set processing",                     OFFSET(type),     AV_OPT_TYPE_INT,    {.i64=1},       0,   1, .flags = FLAGS, "type" },
+    { "time",      "time domain",                        0,                AV_OPT_TYPE_CONST,  {.i64=0},       0,   0, .flags = FLAGS, "type" },
+    { "freq",      "frequency domain",                   0,                AV_OPT_TYPE_CONST,  {.i64=1},       0,   0, .flags = FLAGS, "type" },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(headphone);
+
+static const AVFilterPad outputs[] = {
+    {
+        .name          = "default",
+        .type          = AVMEDIA_TYPE_AUDIO,
+        .config_props  = config_output,
+        .request_frame = request_frame,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_headphone = {
+    .name          = "headphone",
+    .description   = NULL_IF_CONFIG_SMALL("Apply headphone binaural spatialization with HRTFs in additional streams."),
+    .priv_size     = sizeof(HeadphoneContext),
+    .priv_class    = &headphone_class,
+    .init          = init,
+    .uninit        = uninit,
+    .query_formats = query_formats,
+    .inputs        = NULL,
+    .outputs       = outputs,
+    .flags         = AVFILTER_FLAG_SLICE_THREADS | AVFILTER_FLAG_DYNAMIC_INPUTS,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 534c340fa9..94f7cf31a6 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -105,6 +105,7 @@ static void register_all(void)
     REGISTER_FILTER(FIREQUALIZER,   firequalizer,   af);
     REGISTER_FILTER(FLANGER,        flanger,        af);
     REGISTER_FILTER(HDCD,           hdcd,           af);
+    REGISTER_FILTER(HEADPHONE,      headphone,      af);
     REGISTER_FILTER(HIGHPASS,       highpass,       af);
     REGISTER_FILTER(JOIN,           join,           af);
     REGISTER_FILTER(LADSPA,         ladspa,         af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 11cfe514b8..1fa3cf7535 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   6
-#define LIBAVFILTER_VERSION_MINOR  91
+#define LIBAVFILTER_VERSION_MINOR  92
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \