[FFmpeg-cvslog] avfilter: add audio spectral stats filter
Paul B Mahol
git at videolan.org
Thu Dec 2 10:41:57 EET 2021
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Mon Nov 22 11:44:24 2021 +0100| [11b11577fe801b6afb7b4f46a5e5df853f7c1557] | committer: Paul B Mahol
avfilter: add audio spectral stats filter
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=11b11577fe801b6afb7b4f46a5e5df853f7c1557
---
Changelog | 1 +
doc/filters.texi | 63 +++++
libavfilter/Makefile | 1 +
libavfilter/af_aspectralstats.c | 605 ++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
6 files changed, 672 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index 648079ab64..f6b9802034 100644
--- a/Changelog
+++ b/Changelog
@@ -36,6 +36,7 @@ version <next>:
- VideoToolbox VP9 hwaccel
- VideoToolbox ProRes hwaccel
- support loongarch.
+- aspectralstats audio filter
version 4.4:
diff --git a/doc/filters.texi b/doc/filters.texi
index a1b212d38a..9ddff533de 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2695,6 +2695,69 @@ Set oversampling factor.
This filter supports the all above options as @ref{commands}.
+ at section aspectralstats
+
+Display frequency domain statistical information about the audio channels.
+Statistics are calculated and stored as metadata for each audio channel and for each audio frame.
+
+It accepts the following option:
+ at table @option
+ at item win_size
+Set the window length in samples. Default value is 2048.
+Allowed range is from 32 to 65536.
+
+ at item win_func
+Set window function.
+
+It accepts the following values:
+ at table @samp
+ at item rect
+ at item bartlett
+ at item hann, hanning
+ at item hamming
+ at item blackman
+ at item welch
+ at item flattop
+ at item bharris
+ at item bnuttall
+ at item bhann
+ at item sine
+ at item nuttall
+ at item lanczos
+ at item gauss
+ at item tukey
+ at item dolph
+ at item cauchy
+ at item parzen
+ at item poisson
+ at item bohman
+ at end table
+Default is @code{hann}.
+
+ at item overlap
+Set window overlap. Allowed range is from @code{0}
+to @code{1}. Default value is @code{0.5}.
+
+ at end table
+
+A list of each metadata key follows:
+
+ at table @option
+ at item mean
+ at item variance
+ at item centroid
+ at item spread
+ at item skewness
+ at item kurtosis
+ at item entropy
+ at item flatness
+ at item crest
+ at item flux
+ at item slope
+ at item decrease
+ at item rolloff
+ at end table
+
@section asr
Automatic Speech Recognition
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 0e27aeeff6..551d13aadc 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -92,6 +92,7 @@ OBJS-$(CONFIG_ASETTB_FILTER) += settb.o
OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
OBJS-$(CONFIG_ASIDEDATA_FILTER) += f_sidedata.o
OBJS-$(CONFIG_ASOFTCLIP_FILTER) += af_asoftclip.o
+OBJS-$(CONFIG_ASPECTRALSTATS_FILTER) += af_aspectralstats.o
OBJS-$(CONFIG_ASPLIT_FILTER) += split.o
OBJS-$(CONFIG_ASR_FILTER) += af_asr.o
OBJS-$(CONFIG_ASTATS_FILTER) += af_astats.o
diff --git a/libavfilter/af_aspectralstats.c b/libavfilter/af_aspectralstats.c
new file mode 100644
index 0000000000..da418d22bf
--- /dev/null
+++ b/libavfilter/af_aspectralstats.c
@@ -0,0 +1,605 @@
+/*
+ * Copyright (c) 2021 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 <float.h>
+#include <math.h>
+
+#include "libavutil/audio_fifo.h"
+#include "libavutil/opt.h"
+#include "libavutil/tx.h"
+#include "audio.h"
+#include "avfilter.h"
+#include "filters.h"
+#include "internal.h"
+#include "window_func.h"
+
+typedef struct ChannelSpectralStats {
+ float mean;
+ float variance;
+ float centroid;
+ float spread;
+ float skewness;
+ float kurtosis;
+ float entropy;
+ float flatness;
+ float crest;
+ float flux;
+ float slope;
+ float decrease;
+ float rolloff;
+} ChannelSpectralStats;
+
+typedef struct AudioSpectralStatsContext {
+ const AVClass *class;
+ int win_size;
+ int win_func;
+ float overlap;
+ int nb_channels;
+ int hop_size;
+ ChannelSpectralStats *stats;
+ AVAudioFifo *fifo;
+ float *window_func_lut;
+ int64_t pts;
+ int eof;
+ av_tx_fn tx_fn;
+ AVTXContext **fft;
+ AVComplexFloat **fft_in;
+ AVComplexFloat **fft_out;
+ float **prev_magnitude;
+ float **magnitude;
+} AudioSpectralStatsContext;
+
+#define OFFSET(x) offsetof(AudioSpectralStatsContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption aspectralstats_options[] = {
+ { "win_size", "set the window size", OFFSET(win_size), AV_OPT_TYPE_INT, {.i64=2048}, 32, 65536, A },
+ WIN_FUNC_OPTION("win_func", OFFSET(win_func), A, WFUNC_HANNING),
+ { "overlap", "set window overlap", OFFSET(overlap), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, 0, 1, A },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(aspectralstats);
+
+static int config_output(AVFilterLink *outlink)
+{
+ AudioSpectralStatsContext *s = outlink->src->priv;
+ float overlap, scale;
+ int ret;
+
+ s->nb_channels = outlink->channels;
+ s->fifo = av_audio_fifo_alloc(outlink->format, s->nb_channels, s->win_size);
+ if (!s->fifo)
+ return AVERROR(ENOMEM);
+
+ 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);
+ generate_window_func(s->window_func_lut, s->win_size, s->win_func, &overlap);
+ if (s->overlap == 1.f)
+ s->overlap = overlap;
+
+ s->hop_size = s->win_size * (1.f - s->overlap);
+ if (s->hop_size <= 0)
+ return AVERROR(EINVAL);
+
+ s->stats = av_calloc(s->nb_channels, sizeof(*s->stats));
+ if (!s->stats)
+ return AVERROR(ENOMEM);
+
+ s->fft = av_calloc(s->nb_channels, sizeof(*s->fft));
+ if (!s->fft)
+ return AVERROR(ENOMEM);
+
+ s->magnitude = av_calloc(s->nb_channels, sizeof(*s->magnitude));
+ if (!s->magnitude)
+ return AVERROR(ENOMEM);
+
+ s->prev_magnitude = av_calloc(s->nb_channels, sizeof(*s->prev_magnitude));
+ if (!s->prev_magnitude)
+ return AVERROR(ENOMEM);
+
+ s->fft_in = av_calloc(s->nb_channels, sizeof(*s->fft_in));
+ if (!s->fft_in)
+ return AVERROR(ENOMEM);
+
+ s->fft_out = av_calloc(s->nb_channels, sizeof(*s->fft_out));
+ if (!s->fft_out)
+ return AVERROR(ENOMEM);
+
+ for (int ch = 0; ch < s->nb_channels; ch++) {
+ ret = av_tx_init(&s->fft[ch], &s->tx_fn, AV_TX_FLOAT_FFT, 0, s->win_size, &scale, 0);
+ if (ret < 0)
+ return ret;
+
+ s->fft_in[ch] = av_calloc(s->win_size, sizeof(**s->fft_in));
+ if (!s->fft_in[ch])
+ return AVERROR(ENOMEM);
+
+ s->fft_out[ch] = av_calloc(s->win_size, sizeof(**s->fft_out));
+ if (!s->fft_out[ch])
+ return AVERROR(ENOMEM);
+
+ s->magnitude[ch] = av_calloc(s->win_size, sizeof(**s->magnitude));
+ if (!s->magnitude[ch])
+ return AVERROR(ENOMEM);
+
+ s->prev_magnitude[ch] = av_calloc(s->win_size, sizeof(**s->prev_magnitude));
+ if (!s->prev_magnitude[ch])
+ return AVERROR(ENOMEM);
+ }
+
+ return 0;
+}
+
+static void set_meta(AVDictionary **metadata, int chan, const char *key,
+ const char *fmt, float val)
+{
+ uint8_t value[128];
+ uint8_t key2[128];
+
+ snprintf(value, sizeof(value), fmt, val);
+ if (chan)
+ snprintf(key2, sizeof(key2), "lavfi.aspectralstats.%d.%s", chan, key);
+ else
+ snprintf(key2, sizeof(key2), "lavfi.aspectralstats.%s", key);
+ av_dict_set(metadata, key2, value, 0);
+}
+
+static void set_metadata(AudioSpectralStatsContext *s, AVDictionary **metadata)
+{
+ for (int ch = 0; ch < s->nb_channels; ch++) {
+ ChannelSpectralStats *stats = &s->stats[ch];
+
+ set_meta(metadata, ch + 1, "mean", "%g", stats->mean);
+ set_meta(metadata, ch + 1, "variance", "%g", stats->variance);
+ set_meta(metadata, ch + 1, "centroid", "%g", stats->centroid);
+ set_meta(metadata, ch + 1, "spread", "%g", stats->spread);
+ set_meta(metadata, ch + 1, "skewness", "%g", stats->skewness);
+ set_meta(metadata, ch + 1, "kurtosis", "%g", stats->kurtosis);
+ set_meta(metadata, ch + 1, "entropy", "%g", stats->entropy);
+ set_meta(metadata, ch + 1, "flatness", "%g", stats->flatness);
+ set_meta(metadata, ch + 1, "crest", "%g", stats->crest);
+ set_meta(metadata, ch + 1, "flux", "%g", stats->flux);
+ set_meta(metadata, ch + 1, "slope", "%g", stats->slope);
+ set_meta(metadata, ch + 1, "decrease", "%g", stats->decrease);
+ set_meta(metadata, ch + 1, "rolloff", "%g", stats->rolloff);
+ }
+}
+
+static float spectral_mean(const float *const spectral, int size, int max_freq)
+{
+ float sum = 0.f;
+
+ for (int n = 0; n < size; n++)
+ sum += spectral[n];
+
+ return sum / size;
+}
+
+static float sqrf(float a)
+{
+ return a * a;
+}
+
+static float spectral_variance(const float *const spectral, int size, int max_freq, float mean)
+{
+ float sum = 0.f;
+
+ for (int n = 0; n < size; n++)
+ sum += sqrf(spectral[n] - mean);
+
+ return sum / size;
+}
+
+static float spectral_centroid(const float *const spectral, int size, int max_freq)
+{
+ const float scale = max_freq / (float)size;
+ float num = 0.f, den = 0.f;
+
+ for (int n = 0; n < size; n++) {
+ num += spectral[n] * n * scale;
+ den += spectral[n];
+ }
+
+ if (den <= FLT_EPSILON)
+ return 1.f;
+ return num / den;
+}
+
+static float spectral_spread(const float *const spectral, int size, int max_freq, float centroid)
+{
+ const float scale = max_freq / (float)size;
+ float num = 0.f, den = 0.f;
+
+ for (int n = 0; n < size; n++) {
+ num += spectral[n] * sqrf(n * scale - centroid);
+ den += spectral[n];
+ }
+
+ if (den <= FLT_EPSILON)
+ return 1.f;
+ return sqrtf(num / den);
+}
+
+static float cbrf(float a)
+{
+ return a * a * a;
+}
+
+static float spectral_skewness(const float *const spectral, int size, int max_freq, float centroid, float spread)
+{
+ const float scale = max_freq / (float)size;
+ float num = 0.f, den = 0.f;
+
+ for (int n = 0; n < size; n++) {
+ num += spectral[n] * cbrf(n * scale - centroid);
+ den += spectral[n];
+ }
+
+ den *= cbrf(spread);
+ if (den <= FLT_EPSILON)
+ return 1.f;
+ return num / den;
+}
+
+static float spectral_kurtosis(const float *const spectral, int size, int max_freq, float centroid, float spread)
+{
+ const float scale = max_freq / (float)size;
+ float num = 0.f, den = 0.f;
+
+ for (int n = 0; n < size; n++) {
+ num += spectral[n] * sqrf(sqrf(n * scale - centroid));
+ den += spectral[n];
+ }
+
+ den *= sqrf(sqrf(spread));
+ if (den <= FLT_EPSILON)
+ return 1.f;
+ return num / den;
+}
+
+static float spectral_entropy(const float *const spectral, int size, int max_freq)
+{
+ float num = 0.f, den = 0.f;
+
+ for (int n = 0; n < size; n++) {
+ num += spectral[n] * logf(spectral[n] + FLT_EPSILON);
+ }
+
+ den = logf(size);
+ if (den <= FLT_EPSILON)
+ return 1.f;
+ return -num / den;
+}
+
+static float spectral_flatness(const float *const spectral, int size, int max_freq)
+{
+ float num = 0.f, den = 0.f;
+
+ for (int n = 0; n < size; n++) {
+ float v = FLT_EPSILON + spectral[n];
+ num += logf(v);
+ den += v;
+ }
+
+ num /= size;
+ den /= size;
+ num = expf(num);
+ if (den <= FLT_EPSILON)
+ return 0.f;
+ return num / den;
+}
+
+static float spectral_crest(const float *const spectral, int size, int max_freq)
+{
+ float max = 0.f, mean = 0.f;
+
+ for (int n = 0; n < size; n++) {
+ max = fmaxf(max, spectral[n]);
+ mean += spectral[n];
+ }
+
+ mean /= size;
+ if (mean <= FLT_EPSILON)
+ return 0.f;
+ return max / mean;
+}
+
+static float spectral_flux(const float *const spectral, const float *const prev_spectral,
+ int size, int max_freq)
+{
+ float sum = 0.f;
+
+ for (int n = 0; n < size; n++)
+ sum += sqrf(spectral[n] - prev_spectral[n]);
+
+ return sqrtf(sum);
+}
+
+static float spectral_slope(const float *const spectral, int size, int max_freq)
+{
+ const float mean_freq = size * 0.5f;
+ float mean_spectral = 0.f, num = 0.f, den = 0.f;
+
+ for (int n = 0; n < size; n++)
+ mean_spectral += spectral[n];
+ mean_spectral /= size;
+
+ for (int n = 0; n < size; n++) {
+ num += ((n - mean_freq) / mean_freq) * (spectral[n] - mean_spectral);
+ den += sqrf((n - mean_freq) / mean_freq);
+ }
+
+ if (fabsf(den) <= FLT_EPSILON)
+ return 0.f;
+ return num / den;
+}
+
+static float spectral_decrease(const float *const spectral, int size, int max_freq)
+{
+ float num = 0.f, den = 0.f;
+
+ for (int n = 1; n < size; n++) {
+ num += (spectral[n] - spectral[0]) / n;
+ den += spectral[n];
+ }
+
+ if (den <= FLT_EPSILON)
+ return 0.f;
+ return num / den;
+}
+
+static float spectral_rolloff(const float *const spectral, int size, int max_freq)
+{
+ const float scale = max_freq / (float)size;
+ float norm = 0.f, sum = 0.f;
+ int idx = 0.f;
+
+ for (int n = 0; n < size; n++)
+ norm += spectral[n];
+ norm *= 0.85f;
+
+ for (int n = 0; n < size; n++) {
+ sum += spectral[n];
+ if (sum >= norm) {
+ idx = n;
+ break;
+ }
+ }
+
+ return idx * scale;
+}
+
+static int filter_channel(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
+{
+ AudioSpectralStatsContext *s = ctx->priv;
+ AVFrame *in = arg;
+ const int channels = s->nb_channels;
+ const int samples = in->nb_samples;
+ const int start = (channels * jobnr) / nb_jobs;
+ const int end = (channels * (jobnr+1)) / nb_jobs;
+
+ for (int ch = start; ch < end; ch++) {
+ const float *const src = (const float *const)in->extended_data[ch];
+ ChannelSpectralStats *stats = &s->stats[ch];
+ AVComplexFloat *fft_out = s->fft_out[ch];
+ AVComplexFloat *fft_in = s->fft_in[ch];
+ float *magnitude = s->magnitude[ch];
+ float *prev_magnitude = s->prev_magnitude[ch];
+ const float scale = 1.f / s->win_size;
+
+ for (int n = 0; n < samples; n++) {
+ fft_in[n].re = src[n] * s->window_func_lut[n];
+ fft_in[n].im = 0;
+ }
+
+ for (int n = in->nb_samples; n < s->win_size; n++) {
+ fft_in[n].re = 0;
+ fft_in[n].im = 0;
+ }
+
+ s->tx_fn(s->fft[ch], fft_out, fft_in, sizeof(float));
+
+ for (int n = 0; n < s->win_size / 2; n++) {
+ fft_out[n].re *= scale;
+ fft_out[n].im *= scale;
+ }
+
+ for (int n = 0; n < s->win_size / 2; n++)
+ magnitude[n] = hypotf(fft_out[n].re, fft_out[n].im);
+
+ stats->mean = spectral_mean(magnitude, s->win_size / 2, in->sample_rate / 2);
+ stats->variance = spectral_variance(magnitude, s->win_size / 2, in->sample_rate / 2, stats->mean);
+ stats->centroid = spectral_centroid(magnitude, s->win_size / 2, in->sample_rate / 2);
+ stats->spread = spectral_spread(magnitude, s->win_size / 2, in->sample_rate / 2, stats->centroid);
+ stats->skewness = spectral_skewness(magnitude, s->win_size / 2, in->sample_rate / 2, stats->centroid, stats->spread);
+ stats->kurtosis = spectral_kurtosis(magnitude, s->win_size / 2, in->sample_rate / 2, stats->centroid, stats->spread);
+ stats->entropy = spectral_entropy(magnitude, s->win_size / 2, in->sample_rate / 2);
+ stats->flatness = spectral_flatness(magnitude, s->win_size / 2, in->sample_rate / 2);
+ stats->crest = spectral_crest(magnitude, s->win_size / 2, in->sample_rate / 2);
+ stats->flux = spectral_flux(magnitude, prev_magnitude, s->win_size / 2, in->sample_rate / 2);
+ stats->slope = spectral_slope(magnitude, s->win_size / 2, in->sample_rate / 2);
+ stats->decrease = spectral_decrease(magnitude, s->win_size / 2, in->sample_rate / 2);
+ stats->rolloff = spectral_rolloff(magnitude, s->win_size / 2, in->sample_rate / 2);
+
+ memcpy(prev_magnitude, magnitude, s->win_size * sizeof(float));
+ }
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AVFilterLink *outlink = ctx->outputs[0];
+ AudioSpectralStatsContext *s = ctx->priv;
+ AVDictionary **metadata;
+ AVFrame *out, *in = NULL;
+ int ret = 0;
+
+ out = ff_get_audio_buffer(outlink, s->hop_size);
+ if (!out) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
+
+ if (!in) {
+ in = ff_get_audio_buffer(outlink, s->win_size);
+ if (!in)
+ return AVERROR(ENOMEM);
+ }
+
+ ret = av_audio_fifo_peek(s->fifo, (void **)in->extended_data, s->win_size);
+ if (ret < 0)
+ goto fail;
+
+ metadata = &out->metadata;
+ ff_filter_execute(ctx, filter_channel, in, NULL,
+ FFMIN(inlink->channels, ff_filter_get_nb_threads(ctx)));
+
+ set_metadata(s, metadata);
+
+ out->pts = s->pts;
+ s->pts += av_rescale_q(s->hop_size, (AVRational){1, outlink->sample_rate}, outlink->time_base);
+
+ av_audio_fifo_read(s->fifo, (void **)out->extended_data, s->hop_size);
+
+ av_frame_free(&in);
+ return ff_filter_frame(outlink, out);
+fail:
+ av_frame_free(&in);
+ return ret < 0 ? ret : 0;
+}
+
+static int activate(AVFilterContext *ctx)
+{
+ AVFilterLink *inlink = ctx->inputs[0];
+ AVFilterLink *outlink = ctx->outputs[0];
+ AudioSpectralStatsContext *s = ctx->priv;
+ AVFrame *in = NULL;
+ int ret = 0, status;
+ int64_t pts;
+
+ FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
+
+ if (!s->eof && av_audio_fifo_size(s->fifo) < s->win_size) {
+ ret = ff_inlink_consume_frame(inlink, &in);
+ if (ret < 0)
+ return ret;
+
+ if (ret > 0) {
+ ret = av_audio_fifo_write(s->fifo, (void **)in->extended_data,
+ in->nb_samples);
+ if (ret >= 0 && s->pts == AV_NOPTS_VALUE)
+ s->pts = in->pts;
+
+ av_frame_free(&in);
+ if (ret < 0)
+ return ret;
+ }
+ }
+
+ if ((av_audio_fifo_size(s->fifo) >= s->win_size) ||
+ (av_audio_fifo_size(s->fifo) > 0 && s->eof)) {
+ ret = filter_frame(inlink);
+ if (av_audio_fifo_size(s->fifo) >= s->win_size)
+ ff_filter_set_ready(ctx, 100);
+ return ret;
+ }
+
+ if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {
+ if (status == AVERROR_EOF) {
+ s->eof = 1;
+ if (av_audio_fifo_size(s->fifo) >= 0) {
+ ff_filter_set_ready(ctx, 100);
+ return 0;
+ }
+ }
+ }
+
+ if (s->eof && av_audio_fifo_size(s->fifo) <= 0) {
+ ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
+ return 0;
+ }
+
+ if (!s->eof)
+ FF_FILTER_FORWARD_WANTED(outlink, inlink);
+
+ return FFERROR_NOT_READY;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ AudioSpectralStatsContext *s = ctx->priv;
+
+ for (int ch = 0; ch < s->nb_channels; ch++) {
+ if (s->fft)
+ av_tx_uninit(&s->fft[ch]);
+ if (s->fft_in)
+ av_freep(&s->fft_in[ch]);
+ if (s->fft_out)
+ av_freep(&s->fft_out[ch]);
+ if (s->magnitude)
+ av_freep(&s->magnitude[ch]);
+ if (s->prev_magnitude)
+ av_freep(&s->prev_magnitude[ch]);
+ }
+
+ av_freep(&s->fft);
+ av_freep(&s->magnitude);
+ av_freep(&s->prev_magnitude);
+ av_freep(&s->fft_in);
+ av_freep(&s->fft_out);
+ av_freep(&s->stats);
+
+ av_freep(&s->window_func_lut);
+ av_audio_fifo_free(s->fifo);
+ s->fifo = NULL;
+}
+
+static const AVFilterPad aspectralstats_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+};
+
+static const AVFilterPad aspectralstats_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
+ },
+};
+
+const AVFilter ff_af_aspectralstats = {
+ .name = "aspectralstats",
+ .description = NULL_IF_CONFIG_SMALL("Show frequency domain statistics about audio frames."),
+ .priv_size = sizeof(AudioSpectralStatsContext),
+ .priv_class = &aspectralstats_class,
+ .uninit = uninit,
+ .activate = activate,
+ FILTER_INPUTS(aspectralstats_inputs),
+ FILTER_OUTPUTS(aspectralstats_outputs),
+ FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_FLTP),
+ .flags = AVFILTER_FLAG_SLICE_THREADS,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 4bf17ef292..00c36c3f63 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -85,6 +85,7 @@ extern const AVFilter ff_af_asettb;
extern const AVFilter ff_af_ashowinfo;
extern const AVFilter ff_af_asidedata;
extern const AVFilter ff_af_asoftclip;
+extern const AVFilter ff_af_aspectralstats;
extern const AVFilter ff_af_asplit;
extern const AVFilter ff_af_asr;
extern const AVFilter ff_af_astats;
diff --git a/libavfilter/version.h b/libavfilter/version.h
index b9e610ea1f..e0bdcb836d 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 8
-#define LIBAVFILTER_VERSION_MINOR 17
+#define LIBAVFILTER_VERSION_MINOR 18
#define LIBAVFILTER_VERSION_MICRO 100
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