606 lines
18 KiB
C
606 lines
18 KiB
C
/*
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* Copyright (c) 2021 Paul B Mahol
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <float.h>
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#include <math.h>
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#include "libavutil/audio_fifo.h"
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#include "libavutil/opt.h"
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#include "libavutil/tx.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "filters.h"
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#include "internal.h"
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#include "window_func.h"
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typedef struct ChannelSpectralStats {
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float mean;
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float variance;
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float centroid;
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float spread;
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float skewness;
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float kurtosis;
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float entropy;
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float flatness;
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float crest;
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float flux;
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float slope;
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float decrease;
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float rolloff;
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} ChannelSpectralStats;
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typedef struct AudioSpectralStatsContext {
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const AVClass *class;
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int win_size;
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int win_func;
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float overlap;
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int nb_channels;
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int hop_size;
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ChannelSpectralStats *stats;
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AVAudioFifo *fifo;
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float *window_func_lut;
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int64_t pts;
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int eof;
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av_tx_fn tx_fn;
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AVTXContext **fft;
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AVComplexFloat **fft_in;
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AVComplexFloat **fft_out;
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float **prev_magnitude;
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float **magnitude;
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} AudioSpectralStatsContext;
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#define OFFSET(x) offsetof(AudioSpectralStatsContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption aspectralstats_options[] = {
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{ "win_size", "set the window size", OFFSET(win_size), AV_OPT_TYPE_INT, {.i64=2048}, 32, 65536, A },
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WIN_FUNC_OPTION("win_func", OFFSET(win_func), A, WFUNC_HANNING),
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{ "overlap", "set window overlap", OFFSET(overlap), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, 0, 1, A },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(aspectralstats);
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static int config_output(AVFilterLink *outlink)
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{
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AudioSpectralStatsContext *s = outlink->src->priv;
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float overlap, scale;
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int ret;
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s->nb_channels = outlink->channels;
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s->fifo = av_audio_fifo_alloc(outlink->format, s->nb_channels, s->win_size);
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if (!s->fifo)
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return AVERROR(ENOMEM);
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s->window_func_lut = av_realloc_f(s->window_func_lut, s->win_size,
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sizeof(*s->window_func_lut));
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if (!s->window_func_lut)
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return AVERROR(ENOMEM);
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generate_window_func(s->window_func_lut, s->win_size, s->win_func, &overlap);
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if (s->overlap == 1.f)
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s->overlap = overlap;
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s->hop_size = s->win_size * (1.f - s->overlap);
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if (s->hop_size <= 0)
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return AVERROR(EINVAL);
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s->stats = av_calloc(s->nb_channels, sizeof(*s->stats));
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if (!s->stats)
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return AVERROR(ENOMEM);
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s->fft = av_calloc(s->nb_channels, sizeof(*s->fft));
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if (!s->fft)
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return AVERROR(ENOMEM);
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s->magnitude = av_calloc(s->nb_channels, sizeof(*s->magnitude));
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if (!s->magnitude)
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return AVERROR(ENOMEM);
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s->prev_magnitude = av_calloc(s->nb_channels, sizeof(*s->prev_magnitude));
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if (!s->prev_magnitude)
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return AVERROR(ENOMEM);
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s->fft_in = av_calloc(s->nb_channels, sizeof(*s->fft_in));
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if (!s->fft_in)
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return AVERROR(ENOMEM);
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s->fft_out = av_calloc(s->nb_channels, sizeof(*s->fft_out));
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if (!s->fft_out)
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return AVERROR(ENOMEM);
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for (int ch = 0; ch < s->nb_channels; ch++) {
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ret = av_tx_init(&s->fft[ch], &s->tx_fn, AV_TX_FLOAT_FFT, 0, s->win_size, &scale, 0);
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if (ret < 0)
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return ret;
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s->fft_in[ch] = av_calloc(s->win_size, sizeof(**s->fft_in));
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if (!s->fft_in[ch])
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return AVERROR(ENOMEM);
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s->fft_out[ch] = av_calloc(s->win_size, sizeof(**s->fft_out));
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if (!s->fft_out[ch])
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return AVERROR(ENOMEM);
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s->magnitude[ch] = av_calloc(s->win_size, sizeof(**s->magnitude));
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if (!s->magnitude[ch])
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return AVERROR(ENOMEM);
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s->prev_magnitude[ch] = av_calloc(s->win_size, sizeof(**s->prev_magnitude));
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if (!s->prev_magnitude[ch])
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return AVERROR(ENOMEM);
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}
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return 0;
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}
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static void set_meta(AVDictionary **metadata, int chan, const char *key,
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const char *fmt, float val)
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{
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uint8_t value[128];
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uint8_t key2[128];
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snprintf(value, sizeof(value), fmt, val);
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if (chan)
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snprintf(key2, sizeof(key2), "lavfi.aspectralstats.%d.%s", chan, key);
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else
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snprintf(key2, sizeof(key2), "lavfi.aspectralstats.%s", key);
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av_dict_set(metadata, key2, value, 0);
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}
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static void set_metadata(AudioSpectralStatsContext *s, AVDictionary **metadata)
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{
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for (int ch = 0; ch < s->nb_channels; ch++) {
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ChannelSpectralStats *stats = &s->stats[ch];
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set_meta(metadata, ch + 1, "mean", "%g", stats->mean);
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set_meta(metadata, ch + 1, "variance", "%g", stats->variance);
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set_meta(metadata, ch + 1, "centroid", "%g", stats->centroid);
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set_meta(metadata, ch + 1, "spread", "%g", stats->spread);
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set_meta(metadata, ch + 1, "skewness", "%g", stats->skewness);
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set_meta(metadata, ch + 1, "kurtosis", "%g", stats->kurtosis);
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set_meta(metadata, ch + 1, "entropy", "%g", stats->entropy);
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set_meta(metadata, ch + 1, "flatness", "%g", stats->flatness);
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set_meta(metadata, ch + 1, "crest", "%g", stats->crest);
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set_meta(metadata, ch + 1, "flux", "%g", stats->flux);
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set_meta(metadata, ch + 1, "slope", "%g", stats->slope);
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set_meta(metadata, ch + 1, "decrease", "%g", stats->decrease);
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set_meta(metadata, ch + 1, "rolloff", "%g", stats->rolloff);
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}
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}
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static float spectral_mean(const float *const spectral, int size, int max_freq)
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{
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float sum = 0.f;
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for (int n = 0; n < size; n++)
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sum += spectral[n];
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return sum / size;
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}
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static float sqrf(float a)
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{
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return a * a;
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}
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static float spectral_variance(const float *const spectral, int size, int max_freq, float mean)
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{
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float sum = 0.f;
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for (int n = 0; n < size; n++)
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sum += sqrf(spectral[n] - mean);
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return sum / size;
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}
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static float spectral_centroid(const float *const spectral, int size, int max_freq)
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{
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const float scale = max_freq / (float)size;
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float num = 0.f, den = 0.f;
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for (int n = 0; n < size; n++) {
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num += spectral[n] * n * scale;
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den += spectral[n];
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}
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if (den <= FLT_EPSILON)
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return 1.f;
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return num / den;
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}
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static float spectral_spread(const float *const spectral, int size, int max_freq, float centroid)
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{
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const float scale = max_freq / (float)size;
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float num = 0.f, den = 0.f;
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for (int n = 0; n < size; n++) {
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num += spectral[n] * sqrf(n * scale - centroid);
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den += spectral[n];
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}
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if (den <= FLT_EPSILON)
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return 1.f;
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return sqrtf(num / den);
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}
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static float cbrf(float a)
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{
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return a * a * a;
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}
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static float spectral_skewness(const float *const spectral, int size, int max_freq, float centroid, float spread)
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{
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const float scale = max_freq / (float)size;
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float num = 0.f, den = 0.f;
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for (int n = 0; n < size; n++) {
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num += spectral[n] * cbrf(n * scale - centroid);
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den += spectral[n];
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}
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den *= cbrf(spread);
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if (den <= FLT_EPSILON)
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return 1.f;
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return num / den;
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}
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static float spectral_kurtosis(const float *const spectral, int size, int max_freq, float centroid, float spread)
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{
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const float scale = max_freq / (float)size;
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float num = 0.f, den = 0.f;
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for (int n = 0; n < size; n++) {
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num += spectral[n] * sqrf(sqrf(n * scale - centroid));
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den += spectral[n];
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}
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den *= sqrf(sqrf(spread));
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if (den <= FLT_EPSILON)
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return 1.f;
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return num / den;
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}
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static float spectral_entropy(const float *const spectral, int size, int max_freq)
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{
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float num = 0.f, den = 0.f;
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for (int n = 0; n < size; n++) {
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num += spectral[n] * logf(spectral[n] + FLT_EPSILON);
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}
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den = logf(size);
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if (den <= FLT_EPSILON)
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return 1.f;
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return -num / den;
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}
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static float spectral_flatness(const float *const spectral, int size, int max_freq)
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{
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float num = 0.f, den = 0.f;
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for (int n = 0; n < size; n++) {
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float v = FLT_EPSILON + spectral[n];
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num += logf(v);
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den += v;
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}
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num /= size;
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den /= size;
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num = expf(num);
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if (den <= FLT_EPSILON)
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return 0.f;
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return num / den;
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}
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static float spectral_crest(const float *const spectral, int size, int max_freq)
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{
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float max = 0.f, mean = 0.f;
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for (int n = 0; n < size; n++) {
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max = fmaxf(max, spectral[n]);
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mean += spectral[n];
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}
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mean /= size;
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if (mean <= FLT_EPSILON)
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return 0.f;
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return max / mean;
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}
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static float spectral_flux(const float *const spectral, const float *const prev_spectral,
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int size, int max_freq)
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{
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float sum = 0.f;
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for (int n = 0; n < size; n++)
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sum += sqrf(spectral[n] - prev_spectral[n]);
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return sqrtf(sum);
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}
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static float spectral_slope(const float *const spectral, int size, int max_freq)
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{
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const float mean_freq = size * 0.5f;
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float mean_spectral = 0.f, num = 0.f, den = 0.f;
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for (int n = 0; n < size; n++)
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mean_spectral += spectral[n];
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mean_spectral /= size;
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for (int n = 0; n < size; n++) {
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num += ((n - mean_freq) / mean_freq) * (spectral[n] - mean_spectral);
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den += sqrf((n - mean_freq) / mean_freq);
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}
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if (fabsf(den) <= FLT_EPSILON)
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return 0.f;
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return num / den;
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}
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static float spectral_decrease(const float *const spectral, int size, int max_freq)
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{
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float num = 0.f, den = 0.f;
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for (int n = 1; n < size; n++) {
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num += (spectral[n] - spectral[0]) / n;
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den += spectral[n];
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}
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if (den <= FLT_EPSILON)
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return 0.f;
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return num / den;
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}
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static float spectral_rolloff(const float *const spectral, int size, int max_freq)
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{
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const float scale = max_freq / (float)size;
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float norm = 0.f, sum = 0.f;
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int idx = 0.f;
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for (int n = 0; n < size; n++)
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norm += spectral[n];
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norm *= 0.85f;
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for (int n = 0; n < size; n++) {
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sum += spectral[n];
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if (sum >= norm) {
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idx = n;
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break;
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}
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}
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return idx * scale;
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}
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static int filter_channel(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
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{
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AudioSpectralStatsContext *s = ctx->priv;
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AVFrame *in = arg;
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const int channels = s->nb_channels;
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const int samples = in->nb_samples;
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const int start = (channels * jobnr) / nb_jobs;
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const int end = (channels * (jobnr+1)) / nb_jobs;
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for (int ch = start; ch < end; ch++) {
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const float *const src = (const float *const)in->extended_data[ch];
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ChannelSpectralStats *stats = &s->stats[ch];
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AVComplexFloat *fft_out = s->fft_out[ch];
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AVComplexFloat *fft_in = s->fft_in[ch];
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float *magnitude = s->magnitude[ch];
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float *prev_magnitude = s->prev_magnitude[ch];
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const float scale = 1.f / s->win_size;
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for (int n = 0; n < samples; n++) {
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fft_in[n].re = src[n] * s->window_func_lut[n];
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fft_in[n].im = 0;
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}
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for (int n = in->nb_samples; n < s->win_size; n++) {
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fft_in[n].re = 0;
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fft_in[n].im = 0;
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}
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s->tx_fn(s->fft[ch], fft_out, fft_in, sizeof(float));
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for (int n = 0; n < s->win_size / 2; n++) {
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fft_out[n].re *= scale;
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fft_out[n].im *= scale;
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}
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for (int n = 0; n < s->win_size / 2; n++)
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magnitude[n] = hypotf(fft_out[n].re, fft_out[n].im);
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stats->mean = spectral_mean(magnitude, s->win_size / 2, in->sample_rate / 2);
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stats->variance = spectral_variance(magnitude, s->win_size / 2, in->sample_rate / 2, stats->mean);
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stats->centroid = spectral_centroid(magnitude, s->win_size / 2, in->sample_rate / 2);
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stats->spread = spectral_spread(magnitude, s->win_size / 2, in->sample_rate / 2, stats->centroid);
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stats->skewness = spectral_skewness(magnitude, s->win_size / 2, in->sample_rate / 2, stats->centroid, stats->spread);
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stats->kurtosis = spectral_kurtosis(magnitude, s->win_size / 2, in->sample_rate / 2, stats->centroid, stats->spread);
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stats->entropy = spectral_entropy(magnitude, s->win_size / 2, in->sample_rate / 2);
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stats->flatness = spectral_flatness(magnitude, s->win_size / 2, in->sample_rate / 2);
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stats->crest = spectral_crest(magnitude, s->win_size / 2, in->sample_rate / 2);
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stats->flux = spectral_flux(magnitude, prev_magnitude, s->win_size / 2, in->sample_rate / 2);
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stats->slope = spectral_slope(magnitude, s->win_size / 2, in->sample_rate / 2);
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stats->decrease = spectral_decrease(magnitude, s->win_size / 2, in->sample_rate / 2);
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stats->rolloff = spectral_rolloff(magnitude, s->win_size / 2, in->sample_rate / 2);
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memcpy(prev_magnitude, magnitude, s->win_size * sizeof(float));
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}
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return 0;
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}
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static int filter_frame(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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AVFilterLink *outlink = ctx->outputs[0];
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AudioSpectralStatsContext *s = ctx->priv;
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AVDictionary **metadata;
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AVFrame *out, *in = NULL;
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int ret = 0;
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out = ff_get_audio_buffer(outlink, s->hop_size);
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if (!out) {
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ret = AVERROR(ENOMEM);
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goto fail;
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}
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if (!in) {
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in = ff_get_audio_buffer(outlink, s->win_size);
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if (!in)
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return AVERROR(ENOMEM);
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}
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ret = av_audio_fifo_peek(s->fifo, (void **)in->extended_data, s->win_size);
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if (ret < 0)
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goto fail;
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metadata = &out->metadata;
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ff_filter_execute(ctx, filter_channel, in, NULL,
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FFMIN(inlink->channels, ff_filter_get_nb_threads(ctx)));
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set_metadata(s, metadata);
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out->pts = s->pts;
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s->pts += av_rescale_q(s->hop_size, (AVRational){1, outlink->sample_rate}, outlink->time_base);
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av_audio_fifo_read(s->fifo, (void **)out->extended_data, s->hop_size);
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av_frame_free(&in);
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return ff_filter_frame(outlink, out);
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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,
|
|
};
|