mirror of
https://github.com/FFmpeg/FFmpeg.git
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avfilter: add axcorrelate filter
This commit is contained in:
Paul B Mahol
parent
aaac48fb50
commit
93414ce831
@ -24,6 +24,7 @@ version <next>:
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- AV1 encoding support via librav1e
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- AV1 frame merge bitstream filter
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- AV1 Annex B demuxer
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- axcorrelate filter
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version 4.2:
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@ -2531,6 +2531,39 @@ ffmpeg -i INPUT -af atrim=end_sample=1000
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@end itemize
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@section axcorrelate
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Calculate normalized cross-correlation between two input audio streams.
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Resulted samples are always between -1 and 1 inclusive.
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If result is 1 it means two input samples are highly correlated in that selected segment.
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Result 0 means they are not correlated at all.
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If result is -1 it means two input samples are out of phase, which means they cancel each
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other.
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The filter accepts the following options:
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@table @option
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@item size
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Set size of segment over which cross-correlation is calculated.
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Default is 256. Allowed range is from 2 to 131072.
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@item algo
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Set algorithm for cross-correlation. Can be @code{slow} or @code{fast}.
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Default is @code{slow}. Fast algorithm assumes mean values over any given segment
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are always zero and thus need much less calculations to make.
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This is generally not true, but is valid for typical audio streams.
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@end table
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@subsection Examples
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@itemize
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@item
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Calculate correlation between channels in stereo audio stream:
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@example
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ffmpeg -i stereo.wav -af channelsplit,axcorrelate=size=1024:algo=fast correlation.wav
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@end example
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@end itemize
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@section bandpass
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Apply a two-pole Butterworth band-pass filter with central
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@ -88,6 +88,7 @@ OBJS-$(CONFIG_ASTATS_FILTER) += af_astats.o
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OBJS-$(CONFIG_ASTREAMSELECT_FILTER) += f_streamselect.o framesync.o
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OBJS-$(CONFIG_ATEMPO_FILTER) += af_atempo.o
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OBJS-$(CONFIG_ATRIM_FILTER) += trim.o
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OBJS-$(CONFIG_AXCORRELATE_FILTER) += af_axcorrelate.o
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OBJS-$(CONFIG_AZMQ_FILTER) += f_zmq.o
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OBJS-$(CONFIG_BANDPASS_FILTER) += af_biquads.o
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OBJS-$(CONFIG_BANDREJECT_FILTER) += af_biquads.o
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378
libavfilter/af_axcorrelate.c
Normal file
378
libavfilter/af_axcorrelate.c
Normal file
@ -0,0 +1,378 @@
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/*
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* Copyright (c) 2019 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 "libavutil/avassert.h"
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#include "libavutil/audio_fifo.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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#include "libavutil/opt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "formats.h"
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#include "filters.h"
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#include "internal.h"
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typedef struct AudioXCorrelateContext {
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const AVClass *class;
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int size;
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int algo;
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int64_t pts;
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AVAudioFifo *fifo[2];
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AVFrame *cache[2];
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AVFrame *mean_sum[2];
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AVFrame *num_sum;
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AVFrame *den_sum[2];
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int used;
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int (*xcorrelate)(AVFilterContext *ctx, AVFrame *out);
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} AudioXCorrelateContext;
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats;
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AVFilterChannelLayouts *layouts;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_NONE
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};
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int ret;
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layouts = ff_all_channel_counts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ret = ff_set_common_channel_layouts(ctx, layouts);
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if (ret < 0)
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return ret;
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formats = ff_make_format_list(sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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ret = ff_set_common_formats(ctx, formats);
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if (ret < 0)
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return ret;
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formats = ff_all_samplerates();
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if (!formats)
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return AVERROR(ENOMEM);
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return ff_set_common_samplerates(ctx, formats);
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}
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static float mean_sum(const float *in, int size)
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{
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float mean_sum = 0.f;
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for (int i = 0; i < size; i++)
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mean_sum += in[i];
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return mean_sum;
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}
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static float square_sum(const float *x, const float *y, int size)
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{
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float square_sum = 0.f;
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for (int i = 0; i < size; i++)
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square_sum += x[i] * y[i];
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return square_sum;
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}
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static float xcorrelate(const float *x, const float *y, float sumx, float sumy, int size)
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{
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const float xm = sumx / size, ym = sumy / size;
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float num = 0.f, den, den0 = 0.f, den1 = 0.f;
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for (int i = 0; i < size; i++) {
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float xd = x[i] - xm;
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float yd = y[i] - ym;
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num += xd * yd;
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den0 += xd * xd;
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den1 += yd * yd;
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}
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num /= size;
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den = sqrtf((den0 * den1) / (size * size));
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return den <= 1e-6f ? 0.f : num / den;
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}
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static int xcorrelate_slow(AVFilterContext *ctx, AVFrame *out)
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{
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AudioXCorrelateContext *s = ctx->priv;
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const int size = s->size;
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int used;
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for (int ch = 0; ch < out->channels; ch++) {
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const float *x = (const float *)s->cache[0]->extended_data[ch];
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const float *y = (const float *)s->cache[1]->extended_data[ch];
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float *sumx = (float *)s->mean_sum[0]->extended_data[ch];
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float *sumy = (float *)s->mean_sum[1]->extended_data[ch];
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float *dst = (float *)out->extended_data[ch];
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used = s->used;
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if (!used) {
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sumx[0] = mean_sum(x, size);
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sumy[0] = mean_sum(y, size);
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used = 1;
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}
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for (int n = 0; n < out->nb_samples; n++) {
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dst[n] = xcorrelate(x + n, y + n, sumx[0], sumy[0], size);
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sumx[0] -= x[n];
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sumx[0] += x[n + size];
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sumy[0] -= y[n];
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sumy[0] += y[n + size];
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}
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}
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return used;
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}
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static int xcorrelate_fast(AVFilterContext *ctx, AVFrame *out)
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{
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AudioXCorrelateContext *s = ctx->priv;
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const int size = s->size;
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int used;
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for (int ch = 0; ch < out->channels; ch++) {
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const float *x = (const float *)s->cache[0]->extended_data[ch];
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const float *y = (const float *)s->cache[1]->extended_data[ch];
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float *num_sum = (float *)s->num_sum->extended_data[ch];
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float *den_sumx = (float *)s->den_sum[0]->extended_data[ch];
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float *den_sumy = (float *)s->den_sum[1]->extended_data[ch];
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float *dst = (float *)out->extended_data[ch];
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used = s->used;
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if (!used) {
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num_sum[0] = square_sum(x, y, size);
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den_sumx[0] = square_sum(x, x, size);
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den_sumy[0] = square_sum(y, y, size);
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used = 1;
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}
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for (int n = 0; n < out->nb_samples; n++) {
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float num, den;
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num = num_sum[0] / size;
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den = sqrtf((den_sumx[0] * den_sumy[0]) / (size * size));
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dst[n] = den <= 1e-6f ? 0.f : num / den;
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num_sum[0] -= x[n] * y[n];
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num_sum[0] += x[n + size] * y[n + size];
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den_sumx[0] -= x[n] * x[n];
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den_sumx[0] = FFMAX(den_sumx[0], 0.f);
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den_sumx[0] += x[n + size] * x[n + size];
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den_sumy[0] -= y[n] * y[n];
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den_sumy[0] = FFMAX(den_sumy[0], 0.f);
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den_sumy[0] += y[n + size] * y[n + size];
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}
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}
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return used;
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}
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static int activate(AVFilterContext *ctx)
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{
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AudioXCorrelateContext *s = ctx->priv;
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AVFrame *frame = NULL;
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int ret, status;
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int available;
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int64_t pts;
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FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
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for (int i = 0; i < 2; i++) {
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ret = ff_inlink_consume_frame(ctx->inputs[i], &frame);
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if (ret > 0) {
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if (s->pts == AV_NOPTS_VALUE)
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s->pts = frame->pts;
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ret = av_audio_fifo_write(s->fifo[i], (void **)frame->extended_data,
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frame->nb_samples);
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av_frame_free(&frame);
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if (ret < 0)
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return ret;
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}
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}
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available = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1]));
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if (available > s->size) {
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const int out_samples = available - s->size;
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AVFrame *out;
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if (!s->cache[0] || s->cache[0]->nb_samples < available) {
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av_frame_free(&s->cache[0]);
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s->cache[0] = ff_get_audio_buffer(ctx->outputs[0], available);
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if (!s->cache[0])
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return AVERROR(ENOMEM);
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}
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if (!s->cache[1] || s->cache[1]->nb_samples < available) {
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av_frame_free(&s->cache[1]);
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s->cache[1] = ff_get_audio_buffer(ctx->outputs[0], available);
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if (!s->cache[1])
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return AVERROR(ENOMEM);
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}
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ret = av_audio_fifo_peek(s->fifo[0], (void **)s->cache[0]->extended_data, available);
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if (ret < 0)
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return ret;;
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ret = av_audio_fifo_peek(s->fifo[1], (void **)s->cache[1]->extended_data, available);
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if (ret < 0)
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return ret;;
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out = ff_get_audio_buffer(ctx->outputs[0], out_samples);
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if (!out)
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return AVERROR(ENOMEM);
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s->used = s->xcorrelate(ctx, out);
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out->pts = s->pts;
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s->pts += out_samples;
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av_audio_fifo_drain(s->fifo[0], out_samples);
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av_audio_fifo_drain(s->fifo[1], out_samples);
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return ff_filter_frame(ctx->outputs[0], out);
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}
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if (av_audio_fifo_size(s->fifo[0]) > s->size &&
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av_audio_fifo_size(s->fifo[1]) > s->size) {
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ff_filter_set_ready(ctx, 10);
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return 0;
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}
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for (int i = 0; i < 2; i++) {
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if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
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ff_outlink_set_status(ctx->outputs[0], status, pts);
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return 0;
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}
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}
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if (ff_outlink_frame_wanted(ctx->outputs[0])) {
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for (int i = 0; i < 2; i++) {
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if (av_audio_fifo_size(s->fifo[i]) > s->size)
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continue;
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ff_inlink_request_frame(ctx->inputs[i]);
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return 0;
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}
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}
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return FFERROR_NOT_READY;
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}
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static int config_output(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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AVFilterLink *inlink = ctx->inputs[0];
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AudioXCorrelateContext *s = ctx->priv;
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s->pts = AV_NOPTS_VALUE;
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outlink->format = inlink->format;
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outlink->channels = inlink->channels;
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s->fifo[0] = av_audio_fifo_alloc(outlink->format, outlink->channels, s->size);
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s->fifo[1] = av_audio_fifo_alloc(outlink->format, outlink->channels, s->size);
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if (!s->fifo[0] || !s->fifo[1])
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return AVERROR(ENOMEM);
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s->mean_sum[0] = ff_get_audio_buffer(outlink, 1);
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s->mean_sum[1] = ff_get_audio_buffer(outlink, 1);
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s->num_sum = ff_get_audio_buffer(outlink, 1);
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s->den_sum[0] = ff_get_audio_buffer(outlink, 1);
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s->den_sum[1] = ff_get_audio_buffer(outlink, 1);
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if (!s->mean_sum[0] || !s->mean_sum[1] || !s->num_sum ||
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!s->den_sum[0] || !s->den_sum[1])
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return AVERROR(ENOMEM);
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switch (s->algo) {
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case 0: s->xcorrelate = xcorrelate_slow; break;
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case 1: s->xcorrelate = xcorrelate_fast; break;
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}
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return 0;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AudioXCorrelateContext *s = ctx->priv;
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av_audio_fifo_free(s->fifo[0]);
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av_audio_fifo_free(s->fifo[1]);
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av_frame_free(&s->cache[0]);
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av_frame_free(&s->cache[1]);
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av_frame_free(&s->mean_sum[0]);
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av_frame_free(&s->mean_sum[1]);
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av_frame_free(&s->num_sum);
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av_frame_free(&s->den_sum[0]);
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av_frame_free(&s->den_sum[1]);
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}
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static const AVFilterPad inputs[] = {
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{
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.name = "axcorrelate0",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{
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.name = "axcorrelate1",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{ NULL }
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};
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static const AVFilterPad outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_output,
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},
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{ NULL }
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};
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#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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#define OFFSET(x) offsetof(AudioXCorrelateContext, x)
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static const AVOption axcorrelate_options[] = {
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{ "size", "set segment size", OFFSET(size), AV_OPT_TYPE_INT, {.i64=256}, 2, 131072, AF },
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{ "algo", "set alghorithm", OFFSET(algo), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, "algo" },
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{ "slow", "slow algorithm", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "algo" },
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{ "fast", "fast algorithm", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "algo" },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(axcorrelate);
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AVFilter ff_af_axcorrelate = {
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.name = "axcorrelate",
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.description = NULL_IF_CONFIG_SMALL("Cross-correlate two audio streams."),
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.priv_size = sizeof(AudioXCorrelateContext),
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.priv_class = &axcorrelate_class,
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.query_formats = query_formats,
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.activate = activate,
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.uninit = uninit,
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.inputs = inputs,
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.outputs = outputs,
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};
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@ -81,6 +81,7 @@ extern AVFilter ff_af_astats;
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extern AVFilter ff_af_astreamselect;
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extern AVFilter ff_af_atempo;
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extern AVFilter ff_af_atrim;
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extern AVFilter ff_af_axcorrelate;
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extern AVFilter ff_af_azmq;
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extern AVFilter ff_af_bandpass;
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extern AVFilter ff_af_bandreject;
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@ -30,7 +30,7 @@
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#include "libavutil/version.h"
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#define LIBAVFILTER_VERSION_MAJOR 7
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#define LIBAVFILTER_VERSION_MINOR 66
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#define LIBAVFILTER_VERSION_MINOR 67
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#define LIBAVFILTER_VERSION_MICRO 100
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Reference in New Issue
Block a user