diff --git a/doc/filters.texi b/doc/filters.texi
index a0ab2fb..19e4bfe 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -3792,6 +3792,33 @@ channels. Default is 0.3.
 Set level of input signal of original channel. Default is 0.8.
 @end table
 
+@section surround
+Apply audio surround upmix filter.
+
+This filter allows to produce multichannel output from stereo audio stream.
+
+The filter accepts the following options:
+
+@table @option
+@item chl_out
+Set output channel layout. By default is @var{5.1}.
+
+@item level_in
+Set input volume level. By default is @var{1}.
+
+@item level_out
+Set output volume level. By default is @var{1}.
+
+@item lfe
+Enable LFE channel output if output channel layout have it. By default is enabled.
+
+@item lfe_low
+Set LFE low cut off frequency. By default is @var{128} Hz.
+
+@item lfe_high
+Set LFE high cut off frequency. By default is @var{256} Hz.
+@end table
+
 @section treble
 
 Boost or cut treble (upper) frequencies of the audio using a two-pole
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 434a989..c88dfb3 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -108,6 +108,7 @@ OBJS-$(CONFIG_SILENCEREMOVE_FILTER)          += af_silenceremove.o
 OBJS-$(CONFIG_SOFALIZER_FILTER)              += af_sofalizer.o
 OBJS-$(CONFIG_STEREOTOOLS_FILTER)            += af_stereotools.o
 OBJS-$(CONFIG_STEREOWIDEN_FILTER)            += af_stereowiden.o
+OBJS-$(CONFIG_SURROUND_FILTER)               += af_surround.o
 OBJS-$(CONFIG_TREBLE_FILTER)                 += af_biquads.o
 OBJS-$(CONFIG_TREMOLO_FILTER)                += af_tremolo.o
 OBJS-$(CONFIG_VIBRATO_FILTER)                += af_vibrato.o generate_wave_table.o
diff --git a/libavfilter/af_surround.c b/libavfilter/af_surround.c
new file mode 100644
index 0000000..310b8d7
--- /dev/null
+++ b/libavfilter/af_surround.c
@@ -0,0 +1,853 @@
+/*
+ * 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/audio_fifo.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+#include "libavcodec/avfft.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "formats.h"
+
+typedef struct AudioSurroundContext {
+    const AVClass *class;
+
+    char *out_channel_layout_str;
+    float level_in;
+    float level_out;
+    int output_lfe;
+    int lowcutf;
+    int highcutf;
+
+    float lowcut;
+    float highcut;
+
+    uint64_t out_channel_layout;
+    int nb_in_channels;
+    int nb_out_channels;
+
+    AVFrame *input;
+    AVFrame *output;
+    AVFrame *overlap_buffer;
+
+    int buf_size;
+    int hop_size;
+    AVAudioFifo *fifo;
+    RDFTContext **rdft, **irdft;
+    float *window_func_lut;
+
+    int64_t pts;
+
+    void (*upmix)(AVFilterContext *ctx,
+                  float l_phase,
+                  float r_phase,
+                  float c_phase,
+                  float mag_total,
+                  float x, float y,
+                  int n);
+} AudioSurroundContext;
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AudioSurroundContext *s = ctx->priv;
+    AVFilterFormats *formats = NULL;
+    AVFilterChannelLayouts *layouts = NULL;
+    int ret;
+
+    ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLTP);
+    if (ret)
+        return ret;
+    ret = ff_set_common_formats(ctx, formats);
+    if (ret)
+        return ret;
+
+    layouts = NULL;
+    ret = ff_add_channel_layout(&layouts, s->out_channel_layout);
+    if (ret)
+        return ret;
+
+    ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts);
+    if (ret)
+        return ret;
+
+    layouts = NULL;
+    ret = ff_add_channel_layout(&layouts, AV_CH_LAYOUT_STEREO);
+    if (ret)
+        return ret;
+
+    ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_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;
+    AudioSurroundContext *s = ctx->priv;
+    int ch;
+
+    s->rdft = av_calloc(inlink->channels, sizeof(*s->rdft));
+    if (!s->rdft)
+        return AVERROR(ENOMEM);
+
+    for (ch = 0; ch < inlink->channels; ch++) {
+        s->rdft[ch]  = av_rdft_init(ff_log2(s->buf_size), DFT_R2C);
+        if (!s->rdft[ch])
+            return AVERROR(ENOMEM);
+    }
+    s->nb_in_channels = inlink->channels;
+
+    s->input = ff_get_audio_buffer(inlink, s->buf_size * 2);
+    if (!s->input)
+        return AVERROR(ENOMEM);
+
+    s->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, s->buf_size);
+    if (!s->fifo)
+        return AVERROR(ENOMEM);
+
+    s->lowcut = 1.f * s->lowcutf / (inlink->sample_rate * 0.5) * (s->buf_size / 2);
+    s->highcut = 1.f * s->highcutf / (inlink->sample_rate * 0.5) * (s->buf_size / 2);
+
+    return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    AudioSurroundContext *s = ctx->priv;
+    int ch;
+
+    s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft));
+    if (!s->irdft)
+        return AVERROR(ENOMEM);
+
+    for (ch = 0; ch < outlink->channels; ch++) {
+        s->irdft[ch] = av_rdft_init(ff_log2(s->buf_size), IDFT_C2R);
+        if (!s->irdft[ch])
+            return AVERROR(ENOMEM);
+    }
+    s->nb_out_channels = outlink->channels;
+
+    s->output = ff_get_audio_buffer(outlink, s->buf_size * 2);
+    s->overlap_buffer = ff_get_audio_buffer(outlink, s->buf_size * 2);
+    if (!s->overlap_buffer || !s->output)
+        return AVERROR(ENOMEM);
+
+    return 0;
+}
+
+static void stereo_position(float a, float p, float *x, float *y)
+{
+      *x = av_clipf(a+FFMAX(0, sinf(p-M_PI_2))*FFDIFFSIGN(a,0), -1, 1);
+      *y = av_clipf(cosf(a*M_PI_2+M_PI)*cosf(M_PI_2-p/M_PI)*M_LN10+1, -1, 1);
+}
+
+static void upmix_1_0(AVFilterContext *ctx,
+                      float l_phase,
+                      float r_phase,
+                      float c_phase,
+                      float mag_total,
+                      float x, float y,
+                      int n)
+{
+    float mag, *dst;
+    AudioSurroundContext *s = ctx->priv;
+
+    dst = (float *)s->output->extended_data[0];
+
+    mag = sqrtf(1 - fabsf(x)) * ((y + 1) * .5) * mag_total;
+
+    dst[2 * n    ] = mag * cosf(c_phase);
+    dst[2 * n + 1] = mag * sinf(c_phase);
+}
+
+static void upmix_stereo(AVFilterContext *ctx,
+                         float l_phase,
+                         float r_phase,
+                         float c_phase,
+                         float mag_total,
+                         float x, float y,
+                         int n)
+{
+    float l_mag, r_mag, *dstl, *dstr;
+    AudioSurroundContext *s = ctx->priv;
+
+    dstl = (float *)s->output->extended_data[0];
+    dstr = (float *)s->output->extended_data[1];
+
+    l_mag = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total;
+    r_mag = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total;
+
+    dstl[2 * n    ] = l_mag * cosf(l_phase);
+    dstl[2 * n + 1] = l_mag * sinf(l_phase);
+
+    dstr[2 * n    ] = r_mag * cosf(r_phase);
+    dstr[2 * n + 1] = r_mag * sinf(r_phase);
+}
+
+static void upmix_2_1(AVFilterContext *ctx,
+                      float l_phase,
+                      float r_phase,
+                      float c_phase,
+                      float mag_total,
+                      float x, float y,
+                      int n)
+{
+    AudioSurroundContext *s = ctx->priv;
+    float lfe_mag, l_mag, r_mag, *dstl, *dstr, *dstlfe;
+
+    dstl = (float *)s->output->extended_data[0];
+    dstr = (float *)s->output->extended_data[1];
+    dstlfe = (float *)s->output->extended_data[2];
+
+    if (s->output_lfe && n < s->highcut) {
+        lfe_mag = n < s->lowcut ? 1.f : .5f*(1.f+cosf(M_PI*(s->lowcut-n)/(s->lowcut-s->highcut)));
+        lfe_mag *= mag_total;
+        mag_total -= lfe_mag;
+    } else {
+        lfe_mag = 0.f;
+    }
+
+    l_mag = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total;
+    r_mag = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total;
+
+    dstl[2 * n    ] = l_mag * cosf(l_phase);
+    dstl[2 * n + 1] = l_mag * sinf(l_phase);
+
+    dstr[2 * n    ] = r_mag * cosf(r_phase);
+    dstr[2 * n + 1] = r_mag * sinf(r_phase);
+
+    dstlfe[2 * n    ] = lfe_mag * cosf(c_phase);
+    dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
+}
+
+static void upmix_3_0(AVFilterContext *ctx,
+                      float l_phase,
+                      float r_phase,
+                      float c_phase,
+                      float mag_total,
+                      float x, float y,
+                      int n)
+{
+    AudioSurroundContext *s = ctx->priv;
+    float l_mag, r_mag, c_mag, *dstc, *dstl, *dstr;
+
+    dstl = (float *)s->output->extended_data[0];
+    dstr = (float *)s->output->extended_data[1];
+    dstc = (float *)s->output->extended_data[2];
+
+    c_mag = sqrtf(1 - fabsf(x)) * ((y + 1) * .5) * mag_total;
+    l_mag = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total;
+    r_mag = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total;
+
+    dstl[2 * n    ] = l_mag * cosf(l_phase);
+    dstl[2 * n + 1] = l_mag * sinf(l_phase);
+
+    dstr[2 * n    ] = r_mag * cosf(r_phase);
+    dstr[2 * n + 1] = r_mag * sinf(r_phase);
+
+    dstc[2 * n    ] = c_mag * cosf(c_phase);
+    dstc[2 * n + 1] = c_mag * sinf(c_phase);
+}
+
+static void upmix_3_1(AVFilterContext *ctx,
+                      float l_phase,
+                      float r_phase,
+                      float c_phase,
+                      float mag_total,
+                      float x, float y,
+                      int n)
+{
+    AudioSurroundContext *s = ctx->priv;
+    float lfe_mag, l_mag, r_mag, c_mag, *dstc, *dstl, *dstr, *dstlfe;
+
+    dstl = (float *)s->output->extended_data[0];
+    dstr = (float *)s->output->extended_data[1];
+    dstc = (float *)s->output->extended_data[2];
+    dstlfe = (float *)s->output->extended_data[3];
+
+    if (s->output_lfe && n < s->highcut) {
+        lfe_mag = n < s->lowcut ? 1.f : .5f*(1.f+cosf(M_PI*(s->lowcut-n)/(s->lowcut-s->highcut)));
+        lfe_mag *= mag_total;
+        mag_total -= lfe_mag;
+    } else {
+        lfe_mag = 0.f;
+    }
+
+    c_mag = sqrtf(1 - fabsf(x))   * ((y + 1) * .5) * mag_total;
+    l_mag = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total;
+    r_mag = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total;
+
+    dstl[2 * n    ] = l_mag * cosf(l_phase);
+    dstl[2 * n + 1] = l_mag * sinf(l_phase);
+
+    dstr[2 * n    ] = r_mag * cosf(r_phase);
+    dstr[2 * n + 1] = r_mag * sinf(r_phase);
+
+    dstc[2 * n    ] = c_mag * cosf(c_phase);
+    dstc[2 * n + 1] = c_mag * sinf(c_phase);
+
+    dstlfe[2 * n    ] = lfe_mag * cosf(c_phase);
+    dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
+}
+
+static void upmix_4_0(AVFilterContext *ctx,
+                      float l_phase,
+                      float r_phase,
+                      float c_phase,
+                      float mag_total,
+                      float x, float y,
+                      int n)
+{
+    float b_mag, l_mag, r_mag, c_mag, *dstc, *dstl, *dstr, *dstb;
+    AudioSurroundContext *s = ctx->priv;
+
+    dstl = (float *)s->output->extended_data[0];
+    dstr = (float *)s->output->extended_data[1];
+    dstc = (float *)s->output->extended_data[2];
+    dstb = (float *)s->output->extended_data[3];
+
+    c_mag = sqrtf(1 - fabsf(x))   * ((y + 1) * .5) * mag_total;
+    b_mag = sqrtf(1 - fabsf(x))   * ((1 - y) * .5) * mag_total;
+    l_mag = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total;
+    r_mag = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total;
+
+    dstl[2 * n    ] = l_mag * cosf(l_phase);
+    dstl[2 * n + 1] = l_mag * sinf(l_phase);
+
+    dstr[2 * n    ] = r_mag * cosf(r_phase);
+    dstr[2 * n + 1] = r_mag * sinf(r_phase);
+
+    dstc[2 * n    ] = c_mag * cosf(c_phase);
+    dstc[2 * n + 1] = c_mag * sinf(c_phase);
+
+    dstb[2 * n    ] = b_mag * cosf(c_phase);
+    dstb[2 * n + 1] = b_mag * sinf(c_phase);
+}
+
+static void upmix_4_1(AVFilterContext *ctx,
+                      float l_phase,
+                      float r_phase,
+                      float c_phase,
+                      float mag_total,
+                      float x, float y,
+                      int n)
+{
+    float lfe_mag, b_mag, l_mag, r_mag, c_mag, *dstc, *dstl, *dstr, *dstb, *dstlfe;
+    AudioSurroundContext *s = ctx->priv;
+
+    dstl = (float *)s->output->extended_data[0];
+    dstr = (float *)s->output->extended_data[1];
+    dstc = (float *)s->output->extended_data[2];
+    dstlfe = (float *)s->output->extended_data[3];
+    dstb = (float *)s->output->extended_data[4];
+
+    if (s->output_lfe && n < s->highcut) {
+        lfe_mag = n < s->lowcut ? 1.f : .5f*(1.f+cosf(M_PI*(s->lowcut-n)/(s->lowcut-s->highcut)));
+        lfe_mag *= mag_total;
+        mag_total -= lfe_mag;
+    } else {
+        lfe_mag = 0.f;
+    }
+
+    dstlfe[2 * n    ] = lfe_mag * cosf(c_phase);
+    dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
+
+    c_mag = sqrtf(1 - fabsf(x))   * ((y + 1) * .5) * mag_total;
+    b_mag = sqrtf(1 - fabsf(x))   * ((1 - y) * .5) * mag_total;
+    l_mag = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total;
+    r_mag = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total;
+
+    dstl[2 * n    ] = l_mag * cosf(l_phase);
+    dstl[2 * n + 1] = l_mag * sinf(l_phase);
+
+    dstr[2 * n    ] = r_mag * cosf(r_phase);
+    dstr[2 * n + 1] = r_mag * sinf(r_phase);
+
+    dstc[2 * n    ] = c_mag * cosf(c_phase);
+    dstc[2 * n + 1] = c_mag * sinf(c_phase);
+
+    dstb[2 * n    ] = b_mag * cosf(c_phase);
+    dstb[2 * n + 1] = b_mag * sinf(c_phase);
+}
+
+static void upmix_5_0_back(AVFilterContext *ctx,
+                           float l_phase,
+                           float r_phase,
+                           float c_phase,
+                           float mag_total,
+                           float x, float y,
+                           int n)
+{
+    float l_mag, r_mag, ls_mag, rs_mag, c_mag, *dstc, *dstl, *dstr, *dstls, *dstrs;
+    AudioSurroundContext *s = ctx->priv;
+
+    dstl  = (float *)s->output->extended_data[0];
+    dstr  = (float *)s->output->extended_data[1];
+    dstc  = (float *)s->output->extended_data[2];
+    dstls = (float *)s->output->extended_data[3];
+    dstrs = (float *)s->output->extended_data[4];
+
+    c_mag  = sqrtf(1 - fabsf(x))   * ((y + 1) * .5) * mag_total;
+    l_mag  = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total;
+    r_mag  = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total;
+    ls_mag = sqrtf(0.5 * ( x + 1)) * (1 - ((y + 1) * .5)) * mag_total;
+    rs_mag = sqrtf(0.5 * (-x + 1)) * (1 - ((y + 1) * .5)) * mag_total;
+
+    dstl[2 * n    ] = l_mag * cosf(l_phase);
+    dstl[2 * n + 1] = l_mag * sinf(l_phase);
+
+    dstr[2 * n    ] = r_mag * cosf(r_phase);
+    dstr[2 * n + 1] = r_mag * sinf(r_phase);
+
+    dstc[2 * n    ] = c_mag * cosf(c_phase);
+    dstc[2 * n + 1] = c_mag * sinf(c_phase);
+
+    dstls[2 * n    ] = ls_mag * cosf(l_phase);
+    dstls[2 * n + 1] = ls_mag * sinf(l_phase);
+
+    dstrs[2 * n    ] = rs_mag * cosf(r_phase);
+    dstrs[2 * n + 1] = rs_mag * sinf(r_phase);
+}
+
+static void upmix_5_1_back(AVFilterContext *ctx,
+                           float l_phase,
+                           float r_phase,
+                           float c_phase,
+                           float mag_total,
+                           float x, float y,
+                           int n)
+{
+    float lfe_mag, l_mag, r_mag, ls_mag, rs_mag, c_mag, *dstc, *dstl, *dstr, *dstls, *dstrs, *dstlfe;
+    AudioSurroundContext *s = ctx->priv;
+
+    dstl  = (float *)s->output->extended_data[0];
+    dstr  = (float *)s->output->extended_data[1];
+    dstc  = (float *)s->output->extended_data[2];
+    dstlfe = (float *)s->output->extended_data[3];
+    dstls = (float *)s->output->extended_data[4];
+    dstrs = (float *)s->output->extended_data[5];
+
+    if (s->output_lfe && n < s->highcut) {
+        lfe_mag = n < s->lowcut ? 1.f : .5f*(1.f+cosf(M_PI*(s->lowcut-n)/(s->lowcut-s->highcut)));
+        lfe_mag *= mag_total;
+        mag_total -= lfe_mag;
+    } else {
+        lfe_mag = 0.f;
+    }
+
+    c_mag  = sqrtf(1 - fabsf(x))   * ((y + 1) * .5) * mag_total;
+    l_mag  = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total;
+    r_mag  = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total;
+    ls_mag = sqrtf(0.5 * ( x + 1)) * (1 - ((y + 1) * .5)) * mag_total;
+    rs_mag = sqrtf(0.5 * (-x + 1)) * (1 - ((y + 1) * .5)) * mag_total;
+
+    dstl[2 * n    ] = l_mag * cosf(l_phase);
+    dstl[2 * n + 1] = l_mag * sinf(l_phase);
+
+    dstr[2 * n    ] = r_mag * cosf(r_phase);
+    dstr[2 * n + 1] = r_mag * sinf(r_phase);
+
+    dstc[2 * n    ] = c_mag * cosf(c_phase);
+    dstc[2 * n + 1] = c_mag * sinf(c_phase);
+
+    dstlfe[2 * n    ] = lfe_mag * cosf(c_phase);
+    dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
+
+    dstls[2 * n    ] = ls_mag * cosf(l_phase);
+    dstls[2 * n + 1] = ls_mag * sinf(l_phase);
+
+    dstrs[2 * n    ] = rs_mag * cosf(r_phase);
+    dstrs[2 * n + 1] = rs_mag * sinf(r_phase);
+}
+
+static void upmix_7_0(AVFilterContext *ctx,
+                      float l_phase,
+                      float r_phase,
+                      float c_phase,
+                      float mag_total,
+                      float x, float y,
+                      int n)
+{
+    float l_mag, r_mag, ls_mag, rs_mag, c_mag, lb_mag, rb_mag;
+    float *dstc, *dstl, *dstr, *dstls, *dstrs, *dstlb, *dstrb;
+    AudioSurroundContext *s = ctx->priv;
+
+    dstl  = (float *)s->output->extended_data[0];
+    dstr  = (float *)s->output->extended_data[1];
+    dstc  = (float *)s->output->extended_data[2];
+    dstlb = (float *)s->output->extended_data[3];
+    dstrb = (float *)s->output->extended_data[4];
+    dstls = (float *)s->output->extended_data[5];
+    dstrs = (float *)s->output->extended_data[6];
+
+    c_mag  = sqrtf(1 - fabsf(x))   * ((y + 1) * .5) * mag_total;
+    l_mag  = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total;
+    r_mag  = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total;
+    lb_mag = sqrtf(0.5 * ( x + 1)) * (1 - ((y + 1) * .5)) * mag_total;
+    rb_mag = sqrtf(0.5 * (-x + 1)) * (1 - ((y + 1) * .5)) * mag_total;
+    ls_mag = sqrtf(0.5 * ( x + 1)) * (1 - fabsf(y)) * mag_total;
+    rs_mag = sqrtf(0.5 * (-x + 1)) * (1 - fabsf(y)) * mag_total;
+
+    dstl[2 * n    ] = l_mag * cosf(l_phase);
+    dstl[2 * n + 1] = l_mag * sinf(l_phase);
+
+    dstr[2 * n    ] = r_mag * cosf(r_phase);
+    dstr[2 * n + 1] = r_mag * sinf(r_phase);
+
+    dstc[2 * n    ] = c_mag * cosf(c_phase);
+    dstc[2 * n + 1] = c_mag * sinf(c_phase);
+
+    dstlb[2 * n    ] = lb_mag * cosf(l_phase);
+    dstlb[2 * n + 1] = lb_mag * sinf(l_phase);
+
+    dstrb[2 * n    ] = rb_mag * cosf(r_phase);
+    dstrb[2 * n + 1] = rb_mag * sinf(r_phase);
+
+    dstls[2 * n    ] = ls_mag * cosf(l_phase);
+    dstls[2 * n + 1] = ls_mag * sinf(l_phase);
+
+    dstrs[2 * n    ] = rs_mag * cosf(r_phase);
+    dstrs[2 * n + 1] = rs_mag * sinf(r_phase);
+}
+
+static void upmix_7_1(AVFilterContext *ctx,
+                      float l_phase,
+                      float r_phase,
+                      float c_phase,
+                      float mag_total,
+                      float x, float y,
+                      int n)
+{
+    float lfe_mag, l_mag, r_mag, ls_mag, rs_mag, c_mag, lb_mag, rb_mag;
+    float *dstc, *dstl, *dstr, *dstls, *dstrs, *dstlb, *dstrb, *dstlfe;
+    AudioSurroundContext *s = ctx->priv;
+
+    dstl  = (float *)s->output->extended_data[0];
+    dstr  = (float *)s->output->extended_data[1];
+    dstc  = (float *)s->output->extended_data[2];
+    dstlfe = (float *)s->output->extended_data[3];
+    dstlb = (float *)s->output->extended_data[4];
+    dstrb = (float *)s->output->extended_data[5];
+    dstls = (float *)s->output->extended_data[6];
+    dstrs = (float *)s->output->extended_data[7];
+
+    if (s->output_lfe && n < s->highcut) {
+        lfe_mag = n < s->lowcut ? 1.f : .5f*(1.f+cosf(M_PI*(s->lowcut-n)/(s->lowcut-s->highcut)));
+        lfe_mag *= mag_total;
+        mag_total -= lfe_mag;
+    } else {
+        lfe_mag = 0.f;
+    }
+
+    c_mag  = sqrtf(1 - fabsf(x))   * ((y + 1) * .5) * mag_total;
+    l_mag  = sqrtf(0.5 * ( x + 1)) * ((y + 1) * .5) * mag_total;
+    r_mag  = sqrtf(0.5 * (-x + 1)) * ((y + 1) * .5) * mag_total;
+    lb_mag = sqrtf(0.5 * ( x + 1)) * (1 - ((y + 1) * .5)) * mag_total;
+    rb_mag = sqrtf(0.5 * (-x + 1)) * (1 - ((y + 1) * .5)) * mag_total;
+    ls_mag = sqrtf(0.5 * ( x + 1)) * (1 - fabsf(y)) * mag_total;
+    rs_mag = sqrtf(0.5 * (-x + 1)) * (1 - fabsf(y)) * mag_total;
+
+    dstl[2 * n    ] = l_mag * cosf(l_phase);
+    dstl[2 * n + 1] = l_mag * sinf(l_phase);
+
+    dstr[2 * n    ] = r_mag * cosf(r_phase);
+    dstr[2 * n + 1] = r_mag * sinf(r_phase);
+
+    dstc[2 * n    ] = c_mag * cosf(c_phase);
+    dstc[2 * n + 1] = c_mag * sinf(c_phase);
+
+    dstlfe[2 * n    ] = lfe_mag * cosf(c_phase);
+    dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
+
+    dstlb[2 * n    ] = lb_mag * cosf(l_phase);
+    dstlb[2 * n + 1] = lb_mag * sinf(l_phase);
+
+    dstrb[2 * n    ] = rb_mag * cosf(r_phase);
+    dstrb[2 * n + 1] = rb_mag * sinf(r_phase);
+
+    dstls[2 * n    ] = ls_mag * cosf(l_phase);
+    dstls[2 * n + 1] = ls_mag * sinf(l_phase);
+
+    dstrs[2 * n    ] = rs_mag * cosf(r_phase);
+    dstrs[2 * n + 1] = rs_mag * sinf(r_phase);
+}
+
+static int init(AVFilterContext *ctx)
+{
+    AudioSurroundContext *s = ctx->priv;
+    float overlap;
+    int i;
+
+    if (!(s->out_channel_layout = av_get_channel_layout(s->out_channel_layout_str))) {
+        av_log(ctx, AV_LOG_ERROR, "Error parsing channel layout '%s'.\n",
+               s->out_channel_layout_str);
+        return AVERROR(EINVAL);
+    }
+
+    if (s->lowcutf >= s->highcutf) {
+        av_log(ctx, AV_LOG_ERROR, "Low cut-off '%d' should be less than high cut-off '%d'.\n",
+               s->lowcutf, s->highcutf);
+        return AVERROR(EINVAL);
+    }
+
+    switch (s->out_channel_layout) {
+    case AV_CH_LAYOUT_MONO:
+        s->upmix = upmix_1_0;
+        break;
+    case AV_CH_LAYOUT_STEREO:
+        s->upmix = upmix_stereo;
+        break;
+    case AV_CH_LAYOUT_2POINT1:
+        s->upmix = upmix_2_1;
+        break;
+    case AV_CH_LAYOUT_SURROUND:
+        s->upmix = upmix_3_0;
+        break;
+    case AV_CH_LAYOUT_3POINT1:
+        s->upmix = upmix_3_1;
+        break;
+    case AV_CH_LAYOUT_4POINT0:
+        s->upmix = upmix_4_0;
+        break;
+    case AV_CH_LAYOUT_4POINT1:
+        s->upmix = upmix_4_1;
+        break;
+    case AV_CH_LAYOUT_5POINT0_BACK:
+        s->upmix = upmix_5_0_back;
+        break;
+    case AV_CH_LAYOUT_5POINT1_BACK:
+        s->upmix = upmix_5_1_back;
+        break;
+    case AV_CH_LAYOUT_7POINT0:
+        s->upmix = upmix_7_0;
+        break;
+    case AV_CH_LAYOUT_7POINT1:
+        s->upmix = upmix_7_1;
+        break;
+    default:
+        av_log(ctx, AV_LOG_ERROR, "Unsupported output channel layout '%s'.\n",
+               s->out_channel_layout_str);
+        return AVERROR(EINVAL);
+    }
+
+    s->buf_size = 4096;
+    s->pts = AV_NOPTS_VALUE;
+
+    s->window_func_lut = av_calloc(s->buf_size, sizeof(*s->window_func_lut));
+    if (!s->window_func_lut)
+        return AVERROR(ENOMEM);
+
+    for (i = 0; i < s->buf_size; i++)
+        s->window_func_lut[i] = sqrtf(0.5 * (1 - cosf(2 * M_PI * i / s->buf_size)) / s->buf_size);
+    overlap = .5;
+    s->hop_size = s->buf_size * (1. - overlap);
+
+    return 0;
+}
+
+static int fft_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
+{
+    AudioSurroundContext *s = ctx->priv;
+    const float level_in = s->level_in;
+    float *dst;
+    int n;
+
+    memset(s->input->extended_data[ch] + s->buf_size * sizeof(float), 0, s->buf_size * sizeof(float));
+
+    dst = (float *)s->input->extended_data[ch];
+    for (n = 0; n < s->buf_size; n++) {
+        dst[n] *= s->window_func_lut[n] * level_in;
+    }
+
+    av_rdft_calc(s->rdft[ch], (float *)s->input->extended_data[ch]);
+
+    return 0;
+}
+
+static int ifft_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
+{
+    AudioSurroundContext *s = ctx->priv;
+    const float level_out = s->level_out;
+    AVFrame *out = arg;
+    float *dst, *ptr;
+    int n;
+
+    av_rdft_calc(s->irdft[ch], (float *)s->output->extended_data[ch]);
+
+    dst = (float *)s->output->extended_data[ch];
+    ptr = (float *)s->overlap_buffer->extended_data[ch];
+
+    memmove(s->overlap_buffer->extended_data[ch],
+            s->overlap_buffer->extended_data[ch] + s->hop_size * sizeof(float),
+            s->buf_size * sizeof(float));
+    memset(s->overlap_buffer->extended_data[ch] + s->buf_size * sizeof(float),
+           0, s->hop_size * sizeof(float));
+
+    for (n = 0; n < s->buf_size; n++) {
+        ptr[n] += dst[n] * s->window_func_lut[n] * level_out;
+    }
+
+    ptr = (float *)s->overlap_buffer->extended_data[ch];
+    dst = (float *)out->extended_data[ch];
+    memcpy(dst, ptr, s->hop_size * sizeof(float));
+
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AVFilterLink *outlink = ctx->outputs[0];
+    AudioSurroundContext *s = ctx->priv;
+
+    av_audio_fifo_write(s->fifo, (void **)in->extended_data,
+                        in->nb_samples);
+
+    if (s->pts == AV_NOPTS_VALUE)
+        s->pts = in->pts;
+
+    av_frame_free(&in);
+
+    while (av_audio_fifo_size(s->fifo) >= s->buf_size) {
+        float *srcl, *srcr;
+        AVFrame *out;
+        int n, ret;
+
+        ret = av_audio_fifo_peek(s->fifo, (void **)s->input->extended_data, s->buf_size);
+        if (ret < 0)
+            return ret;
+
+        ctx->internal->execute(ctx, fft_channel, NULL, NULL, inlink->channels);
+
+        srcl = (float *)s->input->extended_data[0];
+        srcr = (float *)s->input->extended_data[1];
+
+        for (n = 0; n < s->buf_size; n++) {
+            float l_re = srcl[2 * n], r_re = srcr[2 * n];
+            float l_im = srcl[2 * n + 1], r_im = srcr[2 * n + 1];
+            float c_phase = atan2f(l_im + r_im, l_re + r_re);
+            float l_mag = hypotf(l_re, l_im);
+            float r_mag = hypotf(r_re, r_im);
+            float l_phase = atan2f(l_im, l_re);
+            float r_phase = atan2f(r_im, r_re);
+            float phase_dif = fabsf(l_phase - r_phase);
+            float mag_dif = (l_mag - r_mag) / (l_mag + r_mag);
+            float mag_total = hypotf(l_mag, r_mag);
+            float x, y;
+
+            if (phase_dif > M_PI)
+                phase_dif = 2 * M_PI - phase_dif;
+
+            stereo_position(mag_dif, phase_dif, &x, &y);
+
+            s->upmix(ctx, l_phase, r_phase, c_phase, mag_total, x, y, n);
+        }
+
+        out = ff_get_audio_buffer(outlink, s->hop_size);
+        if (!out)
+            return AVERROR(ENOMEM);
+
+        ctx->internal->execute(ctx, ifft_channel, out, NULL, outlink->channels);
+
+        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);
+        av_audio_fifo_drain(s->fifo, s->hop_size);
+        ret = ff_filter_frame(outlink, out);
+        if (ret < 0)
+            return ret;
+    }
+
+    return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    AudioSurroundContext *s = ctx->priv;
+    int ch;
+
+    av_frame_free(&s->input);
+    av_frame_free(&s->output);
+    av_frame_free(&s->overlap_buffer);
+
+    for (ch = 0; ch < s->nb_in_channels; ch++) {
+        av_rdft_end(s->rdft[ch]);
+    }
+    for (ch = 0; ch < s->nb_out_channels; ch++) {
+        av_rdft_end(s->irdft[ch]);
+    }
+    av_freep(&s->rdft);
+    av_freep(&s->irdft);
+    av_audio_fifo_free(s->fifo);
+    av_freep(&s->window_func_lut);
+}
+
+#define OFFSET(x) offsetof(AudioSurroundContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption surround_options[] = {
+    { "chl_out",   "set output channel layout", OFFSET(out_channel_layout_str), AV_OPT_TYPE_STRING, {.str="5.1"}, 0,   0, FLAGS },
+    { "level_in",  "set input level",           OFFSET(level_in),               AV_OPT_TYPE_FLOAT,  {.dbl=1},     0,  10, FLAGS },
+    { "level_out", "set output level",          OFFSET(level_out),              AV_OPT_TYPE_FLOAT,  {.dbl=1},     0,  10, FLAGS },
+    { "lfe",       "output LFE",                OFFSET(output_lfe),             AV_OPT_TYPE_BOOL,   {.i64=1},     0,   1, FLAGS },
+    { "lfe_low",   "LFE low cut off",           OFFSET(lowcutf),                AV_OPT_TYPE_INT,    {.i64=128},   0, 256, FLAGS },
+    { "lfe_high",  "LFE high cut off",          OFFSET(highcutf),               AV_OPT_TYPE_INT,    {.i64=256},   0, 512, FLAGS },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(surround);
+
+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,
+        .config_props = config_output,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_surround = {
+    .name           = "surround",
+    .description    = NULL_IF_CONFIG_SMALL("Apply audio surround upmix filter."),
+    .query_formats  = query_formats,
+    .priv_size      = sizeof(AudioSurroundContext),
+    .priv_class     = &surround_class,
+    .init           = init,
+    .uninit         = uninit,
+    .inputs         = inputs,
+    .outputs        = outputs,
+    .flags          = AVFILTER_FLAG_SLICE_THREADS,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index f8cd193..534c340 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -121,6 +121,7 @@ static void register_all(void)
     REGISTER_FILTER(SOFALIZER,      sofalizer,      af);
     REGISTER_FILTER(STEREOTOOLS,    stereotools,    af);
     REGISTER_FILTER(STEREOWIDEN,    stereowiden,    af);
+    REGISTER_FILTER(SURROUND,       surround,       af);
     REGISTER_FILTER(TREBLE,         treble,         af);
     REGISTER_FILTER(TREMOLO,        tremolo,        af);
     REGISTER_FILTER(VIBRATO,        vibrato,        af);
