libavfilter/af_axcorrelate.c - manifest_repos/ffmpeg - Git at Google

 /*
  * 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,
 };
	/*
	* 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,
	};