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FFmpeg/libavfilter/af_aspectralstats.c
2021-12-02 09:35:36 +01:00

606 lines
18 KiB
C

/*
* 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,
};