avfilter: add spectrumsynth filter

Signed-off-by: Paul B Mahol <onemda@gmail.com>

7 files changed, 603 insertions, 1 deletions

diff --git a/Changelog b/Changelog
index 7b19557702..df2a30a6ab 100644
--- a/Changelog
+++ b/Changelog
@@ -52,6 +52,7 @@ version <next>:
 - automatic bitstream filtering
 - showspectrumpic filter
 - libstagefright support removed
+- spectrumsynth filter
 
 
 version 2.8:
diff --git a/configure b/configure
index 28ec5bf01e..7cef6f50ab 100755
--- a/configure
+++ b/configure
@@ -2903,6 +2903,8 @@ showspectrumpic_filter_deps="avcodec"
 showspectrumpic_filter_select="fft"
 sofalizer_filter_deps="netcdf avcodec"
 sofalizer_filter_select="fft"
+spectrumsynth_filter_deps="avcodec"
+spectrumsynth_filter_select="fft"
 spp_filter_deps="gpl avcodec"
 spp_filter_select="fft idctdsp fdctdsp me_cmp pixblockdsp"
 stereo3d_filter_deps="gpl"
@@ -6081,6 +6083,7 @@ enabled sofalizer_filter    && prepend avfilter_deps "avcodec"
 enabled showfreqs_filter    && prepend avfilter_deps "avcodec"
 enabled showspectrum_filter && prepend avfilter_deps "avcodec"
 enabled smartblur_filter    && prepend avfilter_deps "swscale"
+enabled spectrumsynth_filter && prepend avfilter_deps "avcodec"
 enabled subtitles_filter    && prepend avfilter_deps "avformat avcodec"
 enabled uspp_filter         && prepend avfilter_deps "avcodec"
 
diff --git a/doc/filters.texi b/doc/filters.texi
index 45d22f49db..9b3acc977e 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -14578,6 +14578,7 @@ Default is @code{combined}.
 
 @end table
 
+@anchor{showspectrum}
 @section showspectrum
 
 Convert input audio to a video output, representing the audio frequency
@@ -15003,6 +15004,68 @@ ffmpeg -i audio.mp3 -filter_complex "showwavespic,colorchannelmixer=rr=66/255:gg
 @end example
 @end itemize
 
+@section spectrumsynth
+
+Sythesize audio from 2 input video spectrums, first input stream represents
+magnitude across time and second represents phase across time.
+The filter will transform from frequency domain as displayed in videos back
+to time domain as presented in audio output.
+
+This filter is primarly created for reversing processed @ref{showspectrum}
+filter outputs, but can synthesize sound from other spectrograms too.
+But in such case results are going to be poor if the phase data is not
+available, because in such cases phase data need to be recreated, usually
+its just recreated from random noise.
+For best results use gray only output (@code{channel} color mode in
+@ref{showspectrum} filter) and @code{log} scale for magnitude video and
+@code{lin} scale for phase video. To produce phase, for 2nd video, use
+@code{data} option. Inputs videos should generally use @code{fullframe}
+slide mode as that saves resources needed for decoding video.
+
+The filter accepts the following options:
+
+@table @option
+@item sample_rate
+Specify sample rate of output audio, the sample rate of audio from which
+spectrum was generated may differ.
+
+@item channels
+Set number of channels represented in input video spectrums.
+
+@item scale
+Set scale which was used when generating magnitude input spectrum.
+Can be @code{lin} or @code{log}. Default is @code{log}.
+
+@item slide
+Set slide which was used when generating inputs spectrums.
+Can be @code{replace}, @code{scroll}, @code{fullframe} or @code{rscroll}.
+Default is @code{fullframe}.
+
+@item win_func
+Set window function used for resynthesis.
+
+@item overlap
+Set window overlap. In range @code{[0, 1]}. Default is @code{1},
+which means optimal overlap for selected window function will be picked.
+
+@item orientation
+Set orientation of input videos. Can be @code{vertical} or @code{horizontal}.
+Default is @code{vertical}.
+@end table
+
+@subsection Examples
+
+@itemize
+@item
+First create magnitude and phase videos from audio, assuming audio is stereo with 44100 sample rate,
+then resynthesize videos back to audio with spectrumsynth:
+@example
+ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=log:overlap=0.875:color=channel:slide=fullframe:data=magnitude -an -c:v rawvideo magnitude.nut
+ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=lin:overlap=0.875:color=channel:slide=fullframe:data=phase -an -c:v rawvideo phase.nut
+ffmpeg -i magnitude.nut -i phase.nut -lavfi spectrumsynth=channels=2:sample_rate=44100:win_fun=hann:overlap=0.875:slide=fullframe output.flac
+@end example
+@end itemize
+
 @section split, asplit
 
 Split input into several identical outputs.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 689da73aa3..9257a92284 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -290,6 +290,7 @@ OBJS-$(CONFIG_SHOWSPECTRUMPIC_FILTER)        += avf_showspectrum.o window_func.o
 OBJS-$(CONFIG_SHOWVOLUME_FILTER)             += avf_showvolume.o
 OBJS-$(CONFIG_SHOWWAVES_FILTER)              += avf_showwaves.o
 OBJS-$(CONFIG_SHOWWAVESPIC_FILTER)           += avf_showwaves.o
+OBJS-$(CONFIG_SPECTRUMSYNTH_FILTER)          += vaf_spectrumsynth.o window_func.o
 
 # multimedia sources
 OBJS-$(CONFIG_AMOVIE_FILTER)                 += src_movie.o
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 2267e88005..d4815d6585 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -310,6 +310,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER(SHOWVOLUME,     showvolume,     avf);
     REGISTER_FILTER(SHOWWAVES,      showwaves,      avf);
     REGISTER_FILTER(SHOWWAVESPIC,   showwavespic,   avf);
+    REGISTER_FILTER(SPECTRUMSYNTH,  spectrumsynth,  vaf);
 
     /* multimedia sources */
     REGISTER_FILTER(AMOVIE,         amovie,         avsrc);
diff --git a/libavfilter/vaf_spectrumsynth.c b/libavfilter/vaf_spectrumsynth.c
new file mode 100644
index 0000000000..76788e1f2c
--- /dev/null
+++ b/libavfilter/vaf_spectrumsynth.c
@@ -0,0 +1,533 @@
+/*
+ * Copyright (c) 2016 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
+ * SpectrumSynth filter
+ * @todo support float pixel format
+ */
+
+#include "libavcodec/avfft.h"
+#include "libavutil/avassert.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+#include "libavutil/parseutils.h"
+#include "avfilter.h"
+#include "formats.h"
+#include "audio.h"
+#include "video.h"
+#include "internal.h"
+#include "window_func.h"
+
+enum MagnitudeScale { LINEAR, LOG, NB_SCALES };
+enum SlideMode      { REPLACE, SCROLL, FULLFRAME, RSCROLL, NB_SLIDES };
+enum Orientation    { VERTICAL, HORIZONTAL, NB_ORIENTATIONS };
+
+typedef struct SpectrumSynthContext {
+    const AVClass *class;
+    int sample_rate;
+    int channels;
+    int scale;
+    int sliding;
+    int win_func;
+    float overlap;
+    int orientation;
+
+    AVFrame *magnitude, *phase;
+    FFTContext *fft;            ///< Fast Fourier Transform context
+    int fft_bits;               ///< number of bits (FFT window size = 1<<fft_bits)
+    FFTComplex **fft_data;      ///< bins holder for each (displayed) channels
+    int win_size;
+    int size;
+    int nb_freq;
+    int hop_size;
+    int start, end;
+    int xpos;
+    int xend;
+    int64_t pts;
+    float factor;
+    AVFrame *buffer;
+    float *window_func_lut;     ///< Window function LUT
+} SpectrumSynthContext;
+
+#define OFFSET(x) offsetof(SpectrumSynthContext, x)
+#define A AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_AUDIO_PARAM
+#define V AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_VIDEO_PARAM
+
+static const AVOption spectrumsynth_options[] = {
+    { "sample_rate", "set sample rate",  OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64 = 44100}, 15,  INT_MAX, A },
+    { "channels",    "set channels",     OFFSET(channels), AV_OPT_TYPE_INT, {.i64 = 1}, 1, 8, A },
+    { "scale",       "set input amplitude scale", OFFSET(scale), AV_OPT_TYPE_INT, {.i64 = LOG}, 0, NB_SCALES-1, V, "scale" },
+        { "lin",  "linear",      0, AV_OPT_TYPE_CONST, {.i64=LINEAR}, 0, 0, V, "scale" },
+        { "log",  "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64=LOG},    0, 0, V, "scale" },
+    { "slide", "set input sliding mode", OFFSET(sliding), AV_OPT_TYPE_INT, {.i64 = FULLFRAME}, 0, NB_SLIDES-1, V, "slide" },
+        { "replace",   "consume old columns with new",   0, AV_OPT_TYPE_CONST, {.i64=REPLACE},   0, 0, V, "slide" },
+        { "scroll",    "consume only most right column", 0, AV_OPT_TYPE_CONST, {.i64=SCROLL},    0, 0, V, "slide" },
+        { "fullframe", "consume full frames",            0, AV_OPT_TYPE_CONST, {.i64=FULLFRAME}, 0, 0, V, "slide" },
+        { "rscroll",   "consume only most left column",  0, AV_OPT_TYPE_CONST, {.i64=RSCROLL},   0, 0, V, "slide" },
+    { "win_func", "set window function", OFFSET(win_func), AV_OPT_TYPE_INT, {.i64 = 0}, 0, NB_WFUNC-1, A, "win_func" },
+        { "rect",     "Rectangular",      0, AV_OPT_TYPE_CONST, {.i64=WFUNC_RECT},     0, 0, A, "win_func" },
+        { "bartlett", "Bartlett",         0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BARTLETT}, 0, 0, A, "win_func" },
+        { "hann",     "Hann",             0, AV_OPT_TYPE_CONST, {.i64=WFUNC_HANNING},  0, 0, A, "win_func" },
+        { "hanning",  "Hanning",          0, AV_OPT_TYPE_CONST, {.i64=WFUNC_HANNING},  0, 0, A, "win_func" },
+        { "hamming",  "Hamming",          0, AV_OPT_TYPE_CONST, {.i64=WFUNC_HAMMING},  0, 0, A, "win_func" },
+        { "sine",     "Sine",             0, AV_OPT_TYPE_CONST, {.i64=WFUNC_SINE},     0, 0, A, "win_func" },
+    { "overlap", "set window overlap",  OFFSET(overlap), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0,  1, A },
+    { "orientation", "set orientation", OFFSET(orientation), AV_OPT_TYPE_INT, {.i64=VERTICAL}, 0, NB_ORIENTATIONS-1, V, "orientation" },
+        { "vertical",   NULL, 0, AV_OPT_TYPE_CONST, {.i64=VERTICAL},   0, 0, V, "orientation" },
+        { "horizontal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=HORIZONTAL}, 0, 0, V, "orientation" },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(spectrumsynth);
+
+static int query_formats(AVFilterContext *ctx)
+{
+    SpectrumSynthContext *s = ctx->priv;
+    AVFilterFormats *formats = NULL;
+    AVFilterChannelLayouts *layout = NULL;
+    AVFilterLink *magnitude = ctx->inputs[0];
+    AVFilterLink *phase = ctx->inputs[1];
+    AVFilterLink *outlink = ctx->outputs[0];
+    static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE };
+    static const enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_GRAY16,
+                                                   AV_PIX_FMT_YUV444P, AV_PIX_FMT_YUVJ444P,
+                                                   AV_PIX_FMT_YUV444P16, AV_PIX_FMT_NONE };
+    int ret, sample_rates[] = { 48000, -1 };
+
+    formats = ff_make_format_list(sample_fmts);
+    if ((ret = ff_formats_ref         (formats, &outlink->in_formats        )) < 0 ||
+        (ret = ff_add_channel_layout  (&layout, FF_COUNT2LAYOUT(s->channels))) < 0 ||
+        (ret = ff_channel_layouts_ref (layout , &outlink->in_channel_layouts)) < 0)
+        return ret;
+
+    sample_rates[0] = s->sample_rate;
+    formats = ff_make_format_list(sample_rates);
+    if (!formats)
+        return AVERROR(ENOMEM);
+    if ((ret = ff_formats_ref(formats, &outlink->in_samplerates)) < 0)
+        return ret;
+
+    formats = ff_make_format_list(pix_fmts);
+    if (!formats)
+        return AVERROR(ENOMEM);
+    if ((ret = ff_formats_ref(formats, &magnitude->out_formats)) < 0)
+        return ret;
+
+    formats = ff_make_format_list(pix_fmts);
+    if (!formats)
+        return AVERROR(ENOMEM);
+    if ((ret = ff_formats_ref(formats, &phase->out_formats)) < 0)
+        return ret;
+
+    return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    SpectrumSynthContext *s = ctx->priv;
+    int width = ctx->inputs[0]->w;
+    int height = ctx->inputs[0]->h;
+    AVRational time_base  = ctx->inputs[0]->time_base;
+    AVRational frame_rate = ctx->inputs[0]->frame_rate;
+    int i, ch, fft_bits;
+    float factor, overlap;
+
+    outlink->sample_rate = s->sample_rate;
+    outlink->time_base = (AVRational){1, s->sample_rate};
+
+    if (width  != ctx->inputs[1]->w ||
+        height != ctx->inputs[1]->h) {
+        av_log(ctx, AV_LOG_ERROR,
+               "Magnitude and Phase sizes differ (%dx%d vs %dx%d).\n",
+               width, height,
+               ctx->inputs[1]->w, ctx->inputs[1]->h);
+        return AVERROR_INVALIDDATA;
+    } else if (av_cmp_q(time_base, ctx->inputs[1]->time_base) != 0) {
+        av_log(ctx, AV_LOG_ERROR,
+               "Magnitude and Phase time bases differ (%d/%d vs %d/%d).\n",
+               time_base.num, time_base.den,
+               ctx->inputs[1]->time_base.num,
+               ctx->inputs[1]->time_base.den);
+        return AVERROR_INVALIDDATA;
+    } else if (av_cmp_q(frame_rate, ctx->inputs[1]->frame_rate) != 0) {
+        av_log(ctx, AV_LOG_ERROR,
+               "Magnitude and Phase framerates differ (%d/%d vs %d/%d).\n",
+               frame_rate.num, frame_rate.den,
+               ctx->inputs[1]->frame_rate.num,
+               ctx->inputs[1]->frame_rate.den);
+        return AVERROR_INVALIDDATA;
+    }
+
+    s->size = s->orientation == VERTICAL ? height / s->channels : width / s->channels;
+    s->xend = s->orientation == VERTICAL ? width : height;
+
+    for (fft_bits = 1; 1 << fft_bits < 2 * s->size; fft_bits++);
+
+    s->win_size = 1 << fft_bits;
+    s->nb_freq = 1 << (fft_bits - 1);
+
+    s->fft = av_fft_init(fft_bits, 1);
+    if (!s->fft) {
+        av_log(ctx, AV_LOG_ERROR, "Unable to create FFT context. "
+               "The window size might be too high.\n");
+        return AVERROR(EINVAL);
+    }
+    s->fft_data = av_calloc(s->channels, sizeof(*s->fft_data));
+    if (!s->fft_data)
+        return AVERROR(ENOMEM);
+    for (ch = 0; ch < s->channels; ch++) {
+        s->fft_data[ch] = av_calloc(s->win_size, sizeof(**s->fft_data));
+        if (!s->fft_data[ch])
+            return AVERROR(ENOMEM);
+    }
+
+    s->buffer = ff_get_audio_buffer(outlink, s->win_size * 2);
+    if (!s->buffer)
+        return AVERROR(ENOMEM);
+
+    /* pre-calc windowing function */
+    s->window_func_lut = av_realloc_f(s->window_func_lut, s->win_size,
+                                      sizeof(*s->window_func_lut));
+    if (!s->window_func_lut)
+        return AVERROR(ENOMEM);
+    ff_generate_window_func(s->window_func_lut, s->win_size, s->win_func, &overlap);
+    if (s->overlap == 1)
+        s->overlap = overlap;
+    s->hop_size = (1 - s->overlap) * s->win_size;
+    for (factor = 0, i = 0; i < s->win_size; i++) {
+        factor += s->window_func_lut[i] * s->window_func_lut[i];
+    }
+    s->factor = (factor / s->win_size) / FFMAX(1 / (1 - s->overlap) - 1, 1);
+
+    return 0;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    SpectrumSynthContext *s = ctx->priv;
+    int ret;
+
+    if (!s->magnitude) {
+        ret = ff_request_frame(ctx->inputs[0]);
+        if (ret < 0)
+            return ret;
+    }
+    if (!s->phase) {
+        ret = ff_request_frame(ctx->inputs[1]);
+        if (ret < 0)
+            return ret;
+    }
+    return 0;
+}
+
+static void read16_fft_bin(SpectrumSynthContext *s,
+                           int x, int y, int f, int ch)
+{
+    const int m_linesize = s->magnitude->linesize[0];
+    const int p_linesize = s->phase->linesize[0];
+    const uint16_t *m = (uint16_t *)(s->magnitude->data[0] + y * m_linesize);
+    const uint16_t *p = (uint16_t *)(s->phase->data[0] + y * p_linesize);
+    float magnitude, phase;
+
+    switch (s->scale) {
+    case LINEAR:
+        magnitude = m[x] / (double)UINT16_MAX;
+        break;
+    case LOG:
+        magnitude = ff_exp10(((m[x] / (double)UINT16_MAX) - 1.) * 6.);
+        break;
+    }
+    phase = ((p[x] / (double)UINT16_MAX) * 2. - 1.) * M_PI;
+
+    s->fft_data[ch][f].re = magnitude * cos(phase);
+    s->fft_data[ch][f].im = magnitude * sin(phase);
+}
+
+static void read8_fft_bin(SpectrumSynthContext *s,
+                          int x, int y, int f, int ch)
+{
+    const int m_linesize = s->magnitude->linesize[0];
+    const int p_linesize = s->phase->linesize[0];
+    const uint8_t *m = (uint8_t *)(s->magnitude->data[0] + y * m_linesize);
+    const uint8_t *p = (uint8_t *)(s->phase->data[0] + y * p_linesize);
+    float magnitude, phase;
+
+    switch (s->scale) {
+    case LINEAR:
+        magnitude = m[x] / (double)UINT8_MAX;
+        break;
+    case LOG:
+        magnitude = ff_exp10(((m[x] / (double)UINT8_MAX) - 1.) * 6.);
+        break;
+    }
+    phase = ((p[x] / (double)UINT8_MAX) * 2. - 1.) * M_PI;
+
+    s->fft_data[ch][f].re = magnitude * cos(phase);
+    s->fft_data[ch][f].im = magnitude * sin(phase);
+}
+
+static void read_fft_data(AVFilterContext *ctx, int x, int h, int ch)
+{
+    SpectrumSynthContext *s = ctx->priv;
+    AVFilterLink *inlink = ctx->inputs[0];
+    int start = h * (s->channels - ch) - 1;
+    int end = h * (s->channels - ch - 1);
+    int y, f;
+
+    switch (s->orientation) {
+    case VERTICAL:
+        switch (inlink->format) {
+        case AV_PIX_FMT_YUV444P16:
+        case AV_PIX_FMT_GRAY16:
+            for (y = start, f = 0; y >= end; y--, f++) {
+                read16_fft_bin(s, x, y, f, ch);
+            }
+            break;
+        case AV_PIX_FMT_YUVJ444P:
+        case AV_PIX_FMT_YUV444P:
+        case AV_PIX_FMT_GRAY8:
+            for (y = start, f = 0; y >= end; y--, f++) {
+                read8_fft_bin(s, x, y, f, ch);
+            }
+            break;
+        }
+        break;
+    case HORIZONTAL:
+        switch (inlink->format) {
+        case AV_PIX_FMT_YUV444P16:
+        case AV_PIX_FMT_GRAY16:
+            for (y = end, f = 0; y <= start; y++, f++) {
+                read16_fft_bin(s, y, x, f, ch);
+            }
+            break;
+        case AV_PIX_FMT_YUVJ444P:
+        case AV_PIX_FMT_YUV444P:
+        case AV_PIX_FMT_GRAY8:
+            for (y = end, f = 0; y <= start; y++, f++) {
+                read8_fft_bin(s, y, x, f, ch);
+            }
+            break;
+        }
+        break;
+    }
+}
+
+static void synth_window(AVFilterContext *ctx, int x)
+{
+    SpectrumSynthContext *s = ctx->priv;
+    const int h = s->size;
+    int nb = s->win_size;
+    int y, f, ch;
+
+    for (ch = 0; ch < s->channels; ch++) {
+        read_fft_data(ctx, x, h, ch);
+
+        for (y = h; y <= s->nb_freq; y++) {
+            s->fft_data[ch][y].re = 0;
+            s->fft_data[ch][y].im = 0;
+        }
+
+        for (y = s->nb_freq + 1, f = s->nb_freq - 1; y < nb; y++, f--) {
+            s->fft_data[ch][y].re =  s->fft_data[ch][f].re;
+            s->fft_data[ch][y].im = -s->fft_data[ch][f].im;
+        }
+
+        av_fft_permute(s->fft, s->fft_data[ch]);
+        av_fft_calc(s->fft, s->fft_data[ch]);
+    }
+}
+
+static int try_push_frame(AVFilterContext *ctx, int x)
+{
+    SpectrumSynthContext *s = ctx->priv;
+    AVFilterLink *outlink = ctx->outputs[0];
+    const float factor = s->factor;
+    int ch, n, i, ret;
+    int start, end;
+    AVFrame *out;
+
+    synth_window(ctx, x);
+
+    for (ch = 0; ch < s->channels; ch++) {
+        float *buf = (float *)s->buffer->extended_data[ch];
+        int j, k;
+
+        start = s->start;
+        end = s->end;
+        k = end;
+        for (i = 0, j = start; j < k && i < s->win_size; i++, j++) {
+            buf[j] += s->fft_data[ch][i].re;
+        }
+
+        for (; i < s->win_size; i++, j++) {
+            buf[j] = s->fft_data[ch][i].re;
+        }
+
+        start += s->hop_size;
+        end = j;
+
+        if (start >= s->win_size) {
+            start -= s->win_size;
+            end -= s->win_size;
+
+            if (ch == s->channels - 1) {
+                float *dst;
+
+                out = ff_get_audio_buffer(outlink, s->win_size);
+                if (!out) {
+                    av_frame_free(&s->magnitude);
+                    av_frame_free(&s->phase);
+                    return AVERROR(ENOMEM);
+                }
+
+                out->pts = s->pts;
+                s->pts += s->win_size;
+                for (int c = 0; c < s->channels; c++) {
+                    dst = (float *)out->extended_data[c];
+                    buf = (float *)s->buffer->extended_data[c];
+
+                    for (n = 0; n < s->win_size; n++) {
+                        dst[n] = buf[n] * factor;
+                    }
+                    memmove(buf, buf + s->win_size, s->win_size * 4);
+                }
+
+                ret = ff_filter_frame(outlink, out);
+            }
+        }
+    }
+
+    s->start = start;
+    s->end = end;
+
+    return 0;
+}
+
+static int try_push_frames(AVFilterContext *ctx)
+{
+    SpectrumSynthContext *s = ctx->priv;
+    int ret, x;
+
+    if (!(s->magnitude && s->phase))
+        return 0;
+
+    switch (s->sliding) {
+    case REPLACE:
+        ret = try_push_frame(ctx, s->xpos);
+        s->xpos++;
+        if (s->xpos >= s->xend)
+            s->xpos = 0;
+        break;
+    case SCROLL:
+        s->xpos = s->xend - 1;
+        ret = try_push_frame(ctx, s->xpos);
+    case RSCROLL:
+        s->xpos = 0;
+        ret = try_push_frame(ctx, s->xpos);
+        break;
+        break;
+    case FULLFRAME:
+        for (x = 0; x < s->xend; x++) {
+            ret = try_push_frame(ctx, x);
+            if (ret < 0)
+                break;
+        }
+        break;
+    }
+
+    av_frame_free(&s->magnitude);
+    av_frame_free(&s->phase);
+    return ret;
+}
+
+static int filter_frame_magnitude(AVFilterLink *inlink, AVFrame *magnitude)
+{
+    AVFilterContext *ctx = inlink->dst;
+    SpectrumSynthContext *s = ctx->priv;
+
+    s->magnitude = magnitude;
+    return try_push_frames(ctx);
+}
+
+static int filter_frame_phase(AVFilterLink *inlink, AVFrame *phase)
+{
+    AVFilterContext *ctx = inlink->dst;
+    SpectrumSynthContext *s = ctx->priv;
+
+    s->phase = phase;
+    return try_push_frames(ctx);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    SpectrumSynthContext *s = ctx->priv;
+    int i;
+
+    av_frame_free(&s->magnitude);
+    av_frame_free(&s->phase);
+    av_frame_free(&s->buffer);
+    av_fft_end(s->fft);
+    if (s->fft_data) {
+        for (i = 0; i < s->channels; i++)
+            av_freep(&s->fft_data[i]);
+    }
+    av_freep(&s->fft_data);
+    av_freep(&s->window_func_lut);
+}
+
+static const AVFilterPad spectrumsynth_inputs[] = {
+    {
+        .name         = "magnitude",
+        .type         = AVMEDIA_TYPE_VIDEO,
+        .filter_frame = filter_frame_magnitude,
+        .needs_fifo   = 1,
+    },
+    {
+        .name         = "phase",
+        .type         = AVMEDIA_TYPE_VIDEO,
+        .filter_frame = filter_frame_phase,
+        .needs_fifo   = 1,
+    },
+    { NULL }
+};
+
+static const AVFilterPad spectrumsynth_outputs[] = {
+    {
+        .name          = "default",
+        .type          = AVMEDIA_TYPE_AUDIO,
+        .config_props  = config_output,
+        .request_frame = request_frame,
+    },
+    { NULL }
+};
+
+AVFilter ff_vaf_spectrumsynth = {
+    .name          = "spectrumsynth",
+    .description   = NULL_IF_CONFIG_SMALL("Convert input spectrum videos to audio output."),
+    .uninit        = uninit,
+    .query_formats = query_formats,
+    .priv_size     = sizeof(SpectrumSynthContext),
+    .inputs        = spectrumsynth_inputs,
+    .outputs       = spectrumsynth_outputs,
+    .priv_class    = &spectrumsynth_class,
+};
diff --git a/libavfilter/version.h b/libavfilter/version.h
index a7f213de3f..d13929ed33 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  23
+#define LIBAVFILTER_VERSION_MINOR  24
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \