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avfilter: add spectrumsynth filter

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Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol 2016-01-10 14:48:12 +01:00
parent cc538e9dbd
commit 653f9d84ae
7 changed files with 603 additions and 1 deletions
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1
Changelog
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@ -52,6 +52,7 @@ version <next>:
- automatic bitstream filtering
- showspectrumpic filter
- libstagefright support removed
- spectrumsynth filter
version 2.8:

3
configure vendored
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@ -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"

63
doc/filters.texi
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@ -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.

1
libavfilter/Makefile
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@ -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

1
libavfilter/allfilters.c
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@ -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);

533
libavfilter/vaf_spectrumsynth.c Normal file
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@ -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,
};

2
libavfilter/version.h
View File

@ -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, \