@@ -52,6 +52,7 @@ version <next>:

 - automatic bitstream filtering

 - showspectrumpic filter

 - libstagefright support removed

+- spectrumsynth filter

 version 2.8:

@@ -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"

@@ -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.

@@ -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

@@ -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);

new file mode 100644

@@ -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,

+};

@@ -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, \