[FFmpeg-cvslog] avfilter: add axcorrelate filter
Paul B Mahol
git at videolan.org
Sat Nov 23 13:12:04 EET 2019
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Thu Nov 14 21:16:18 2019 +0100| [93414ce831864ec3589294bf27481f6bdb8007fc] | committer: Paul B Mahol
avfilter: add axcorrelate filter
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=93414ce831864ec3589294bf27481f6bdb8007fc
---
Changelog | 1 +
doc/filters.texi | 33 ++++
libavfilter/Makefile | 1 +
libavfilter/af_axcorrelate.c | 378 +++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
6 files changed, 415 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index a18dcfbd42..1e9476b24d 100644
--- a/Changelog
+++ b/Changelog
@@ -24,6 +24,7 @@ version <next>:
- AV1 encoding support via librav1e
- AV1 frame merge bitstream filter
- AV1 Annex B demuxer
+- axcorrelate filter
version 4.2:
diff --git a/doc/filters.texi b/doc/filters.texi
index 39570d893b..16bf2df6c2 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2531,6 +2531,39 @@ ffmpeg -i INPUT -af atrim=end_sample=1000
@end itemize
+ at section axcorrelate
+Calculate normalized cross-correlation between two input audio streams.
+
+Resulted samples are always between -1 and 1 inclusive.
+If result is 1 it means two input samples are highly correlated in that selected segment.
+Result 0 means they are not correlated at all.
+If result is -1 it means two input samples are out of phase, which means they cancel each
+other.
+
+The filter accepts the following options:
+
+ at table @option
+ at item size
+Set size of segment over which cross-correlation is calculated.
+Default is 256. Allowed range is from 2 to 131072.
+
+ at item algo
+Set algorithm for cross-correlation. Can be @code{slow} or @code{fast}.
+Default is @code{slow}. Fast algorithm assumes mean values over any given segment
+are always zero and thus need much less calculations to make.
+This is generally not true, but is valid for typical audio streams.
+ at end table
+
+ at subsection Examples
+
+ at itemize
+ at item
+Calculate correlation between channels in stereo audio stream:
+ at example
+ffmpeg -i stereo.wav -af channelsplit,axcorrelate=size=1024:algo=fast correlation.wav
+ at end example
+ at end itemize
+
@section bandpass
Apply a two-pole Butterworth band-pass filter with central
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 8434983b7d..46e3eecf9a 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -88,6 +88,7 @@ OBJS-$(CONFIG_ASTATS_FILTER) += af_astats.o
OBJS-$(CONFIG_ASTREAMSELECT_FILTER) += f_streamselect.o framesync.o
OBJS-$(CONFIG_ATEMPO_FILTER) += af_atempo.o
OBJS-$(CONFIG_ATRIM_FILTER) += trim.o
+OBJS-$(CONFIG_AXCORRELATE_FILTER) += af_axcorrelate.o
OBJS-$(CONFIG_AZMQ_FILTER) += f_zmq.o
OBJS-$(CONFIG_BANDPASS_FILTER) += af_biquads.o
OBJS-$(CONFIG_BANDREJECT_FILTER) += af_biquads.o
diff --git a/libavfilter/af_axcorrelate.c b/libavfilter/af_axcorrelate.c
new file mode 100644
index 0000000000..861903b0f1
--- /dev/null
+++ b/libavfilter/af_axcorrelate.c
@@ -0,0 +1,378 @@
+/*
+ * Copyright (c) 2019 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 "libavutil/avassert.h"
+#include "libavutil/audio_fifo.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/common.h"
+#include "libavutil/opt.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "formats.h"
+#include "filters.h"
+#include "internal.h"
+
+typedef struct AudioXCorrelateContext {
+ const AVClass *class;
+
+ int size;
+ int algo;
+ int64_t pts;
+
+ AVAudioFifo *fifo[2];
+ AVFrame *cache[2];
+ AVFrame *mean_sum[2];
+ AVFrame *num_sum;
+ AVFrame *den_sum[2];
+ int used;
+
+ int (*xcorrelate)(AVFilterContext *ctx, AVFrame *out);
+} AudioXCorrelateContext;
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats;
+ AVFilterChannelLayouts *layouts;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE
+ };
+ int ret;
+
+ layouts = ff_all_channel_counts();
+ if (!layouts)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_channel_layouts(ctx, layouts);
+ if (ret < 0)
+ return ret;
+
+ formats = ff_make_format_list(sample_fmts);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_formats(ctx, formats);
+ if (ret < 0)
+ return ret;
+
+ formats = ff_all_samplerates();
+ if (!formats)
+ return AVERROR(ENOMEM);
+ return ff_set_common_samplerates(ctx, formats);
+}
+
+static float mean_sum(const float *in, int size)
+{
+ float mean_sum = 0.f;
+
+ for (int i = 0; i < size; i++)
+ mean_sum += in[i];
+
+ return mean_sum;
+}
+
+static float square_sum(const float *x, const float *y, int size)
+{
+ float square_sum = 0.f;
+
+ for (int i = 0; i < size; i++)
+ square_sum += x[i] * y[i];
+
+ return square_sum;
+}
+
+static float xcorrelate(const float *x, const float *y, float sumx, float sumy, int size)
+{
+ const float xm = sumx / size, ym = sumy / size;
+ float num = 0.f, den, den0 = 0.f, den1 = 0.f;
+
+ for (int i = 0; i < size; i++) {
+ float xd = x[i] - xm;
+ float yd = y[i] - ym;
+
+ num += xd * yd;
+ den0 += xd * xd;
+ den1 += yd * yd;
+ }
+
+ num /= size;
+ den = sqrtf((den0 * den1) / (size * size));
+
+ return den <= 1e-6f ? 0.f : num / den;
+}
+
+static int xcorrelate_slow(AVFilterContext *ctx, AVFrame *out)
+{
+ AudioXCorrelateContext *s = ctx->priv;
+ const int size = s->size;
+ int used;
+
+ for (int ch = 0; ch < out->channels; ch++) {
+ const float *x = (const float *)s->cache[0]->extended_data[ch];
+ const float *y = (const float *)s->cache[1]->extended_data[ch];
+ float *sumx = (float *)s->mean_sum[0]->extended_data[ch];
+ float *sumy = (float *)s->mean_sum[1]->extended_data[ch];
+ float *dst = (float *)out->extended_data[ch];
+
+ used = s->used;
+ if (!used) {
+ sumx[0] = mean_sum(x, size);
+ sumy[0] = mean_sum(y, size);
+ used = 1;
+ }
+
+ for (int n = 0; n < out->nb_samples; n++) {
+ dst[n] = xcorrelate(x + n, y + n, sumx[0], sumy[0], size);
+
+ sumx[0] -= x[n];
+ sumx[0] += x[n + size];
+ sumy[0] -= y[n];
+ sumy[0] += y[n + size];
+ }
+ }
+
+ return used;
+}
+
+static int xcorrelate_fast(AVFilterContext *ctx, AVFrame *out)
+{
+ AudioXCorrelateContext *s = ctx->priv;
+ const int size = s->size;
+ int used;
+
+ for (int ch = 0; ch < out->channels; ch++) {
+ const float *x = (const float *)s->cache[0]->extended_data[ch];
+ const float *y = (const float *)s->cache[1]->extended_data[ch];
+ float *num_sum = (float *)s->num_sum->extended_data[ch];
+ float *den_sumx = (float *)s->den_sum[0]->extended_data[ch];
+ float *den_sumy = (float *)s->den_sum[1]->extended_data[ch];
+ float *dst = (float *)out->extended_data[ch];
+
+ used = s->used;
+ if (!used) {
+ num_sum[0] = square_sum(x, y, size);
+ den_sumx[0] = square_sum(x, x, size);
+ den_sumy[0] = square_sum(y, y, size);
+ used = 1;
+ }
+
+ for (int n = 0; n < out->nb_samples; n++) {
+ float num, den;
+
+ num = num_sum[0] / size;
+ den = sqrtf((den_sumx[0] * den_sumy[0]) / (size * size));
+
+ dst[n] = den <= 1e-6f ? 0.f : num / den;
+
+ num_sum[0] -= x[n] * y[n];
+ num_sum[0] += x[n + size] * y[n + size];
+ den_sumx[0] -= x[n] * x[n];
+ den_sumx[0] = FFMAX(den_sumx[0], 0.f);
+ den_sumx[0] += x[n + size] * x[n + size];
+ den_sumy[0] -= y[n] * y[n];
+ den_sumy[0] = FFMAX(den_sumy[0], 0.f);
+ den_sumy[0] += y[n + size] * y[n + size];
+ }
+ }
+
+ return used;
+}
+
+static int activate(AVFilterContext *ctx)
+{
+ AudioXCorrelateContext *s = ctx->priv;
+ AVFrame *frame = NULL;
+ int ret, status;
+ int available;
+ int64_t pts;
+
+ FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
+
+ for (int i = 0; i < 2; i++) {
+ ret = ff_inlink_consume_frame(ctx->inputs[i], &frame);
+ if (ret > 0) {
+ if (s->pts == AV_NOPTS_VALUE)
+ s->pts = frame->pts;
+ ret = av_audio_fifo_write(s->fifo[i], (void **)frame->extended_data,
+ frame->nb_samples);
+ av_frame_free(&frame);
+ if (ret < 0)
+ return ret;
+ }
+ }
+
+ available = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1]));
+ if (available > s->size) {
+ const int out_samples = available - s->size;
+ AVFrame *out;
+
+ if (!s->cache[0] || s->cache[0]->nb_samples < available) {
+ av_frame_free(&s->cache[0]);
+ s->cache[0] = ff_get_audio_buffer(ctx->outputs[0], available);
+ if (!s->cache[0])
+ return AVERROR(ENOMEM);
+ }
+
+ if (!s->cache[1] || s->cache[1]->nb_samples < available) {
+ av_frame_free(&s->cache[1]);
+ s->cache[1] = ff_get_audio_buffer(ctx->outputs[0], available);
+ if (!s->cache[1])
+ return AVERROR(ENOMEM);
+ }
+
+ ret = av_audio_fifo_peek(s->fifo[0], (void **)s->cache[0]->extended_data, available);
+ if (ret < 0)
+ return ret;;
+
+ ret = av_audio_fifo_peek(s->fifo[1], (void **)s->cache[1]->extended_data, available);
+ if (ret < 0)
+ return ret;;
+
+ out = ff_get_audio_buffer(ctx->outputs[0], out_samples);
+ if (!out)
+ return AVERROR(ENOMEM);
+
+ s->used = s->xcorrelate(ctx, out);
+
+ out->pts = s->pts;
+ s->pts += out_samples;
+
+ av_audio_fifo_drain(s->fifo[0], out_samples);
+ av_audio_fifo_drain(s->fifo[1], out_samples);
+
+ return ff_filter_frame(ctx->outputs[0], out);
+ }
+
+ if (av_audio_fifo_size(s->fifo[0]) > s->size &&
+ av_audio_fifo_size(s->fifo[1]) > s->size) {
+ ff_filter_set_ready(ctx, 10);
+ return 0;
+ }
+
+ for (int i = 0; i < 2; i++) {
+ if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
+ ff_outlink_set_status(ctx->outputs[0], status, pts);
+ return 0;
+ }
+ }
+
+ if (ff_outlink_frame_wanted(ctx->outputs[0])) {
+ for (int i = 0; i < 2; i++) {
+ if (av_audio_fifo_size(s->fifo[i]) > s->size)
+ continue;
+ ff_inlink_request_frame(ctx->inputs[i]);
+ return 0;
+ }
+ }
+
+ return FFERROR_NOT_READY;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AVFilterLink *inlink = ctx->inputs[0];
+ AudioXCorrelateContext *s = ctx->priv;
+
+ s->pts = AV_NOPTS_VALUE;
+
+ outlink->format = inlink->format;
+ outlink->channels = inlink->channels;
+ s->fifo[0] = av_audio_fifo_alloc(outlink->format, outlink->channels, s->size);
+ s->fifo[1] = av_audio_fifo_alloc(outlink->format, outlink->channels, s->size);
+ if (!s->fifo[0] || !s->fifo[1])
+ return AVERROR(ENOMEM);
+
+ s->mean_sum[0] = ff_get_audio_buffer(outlink, 1);
+ s->mean_sum[1] = ff_get_audio_buffer(outlink, 1);
+ s->num_sum = ff_get_audio_buffer(outlink, 1);
+ s->den_sum[0] = ff_get_audio_buffer(outlink, 1);
+ s->den_sum[1] = ff_get_audio_buffer(outlink, 1);
+ if (!s->mean_sum[0] || !s->mean_sum[1] || !s->num_sum ||
+ !s->den_sum[0] || !s->den_sum[1])
+ return AVERROR(ENOMEM);
+
+ switch (s->algo) {
+ case 0: s->xcorrelate = xcorrelate_slow; break;
+ case 1: s->xcorrelate = xcorrelate_fast; break;
+ }
+
+ return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ AudioXCorrelateContext *s = ctx->priv;
+
+ av_audio_fifo_free(s->fifo[0]);
+ av_audio_fifo_free(s->fifo[1]);
+ av_frame_free(&s->cache[0]);
+ av_frame_free(&s->cache[1]);
+ av_frame_free(&s->mean_sum[0]);
+ av_frame_free(&s->mean_sum[1]);
+ av_frame_free(&s->num_sum);
+ av_frame_free(&s->den_sum[0]);
+ av_frame_free(&s->den_sum[1]);
+}
+
+static const AVFilterPad inputs[] = {
+ {
+ .name = "axcorrelate0",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ {
+ .name = "axcorrelate1",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
+ },
+ { NULL }
+};
+
+#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+#define OFFSET(x) offsetof(AudioXCorrelateContext, x)
+
+static const AVOption axcorrelate_options[] = {
+ { "size", "set segment size", OFFSET(size), AV_OPT_TYPE_INT, {.i64=256}, 2, 131072, AF },
+ { "algo", "set alghorithm", OFFSET(algo), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, "algo" },
+ { "slow", "slow algorithm", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "algo" },
+ { "fast", "fast algorithm", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "algo" },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(axcorrelate);
+
+AVFilter ff_af_axcorrelate = {
+ .name = "axcorrelate",
+ .description = NULL_IF_CONFIG_SMALL("Cross-correlate two audio streams."),
+ .priv_size = sizeof(AudioXCorrelateContext),
+ .priv_class = &axcorrelate_class,
+ .query_formats = query_formats,
+ .activate = activate,
+ .uninit = uninit,
+ .inputs = inputs,
+ .outputs = outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 7c1e19e1da..2a69227476 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -81,6 +81,7 @@ extern AVFilter ff_af_astats;
extern AVFilter ff_af_astreamselect;
extern AVFilter ff_af_atempo;
extern AVFilter ff_af_atrim;
+extern AVFilter ff_af_axcorrelate;
extern AVFilter ff_af_azmq;
extern AVFilter ff_af_bandpass;
extern AVFilter ff_af_bandreject;
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 2e3ff53b20..7e8d849e0c 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 7
-#define LIBAVFILTER_VERSION_MINOR 66
+#define LIBAVFILTER_VERSION_MINOR 67
#define LIBAVFILTER_VERSION_MICRO 100
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