ffmpeg/libavfilter/af_axcorrelate.c

469 lines
21 KiB
C

/*
* 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/audio_fifo.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.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 eof;
int (*xcorrelate)(AVFilterContext *ctx, AVFrame *out, int available);
} AudioXCorrelateContext;
#define MEAN_SUM(suffix, type, zero) \
static type mean_sum_##suffix(const type *in, \
int size) \
{ \
type mean_sum = zero; \
\
for (int i = 0; i < size; i++) \
mean_sum += in[i]; \
\
return mean_sum; \
}
MEAN_SUM(f, float, 0.f)
MEAN_SUM(d, double, 0.0)
#define SQUARE_SUM(suffix, type, zero) \
static type square_sum_##suffix(const type *x, \
const type *y, \
int size) \
{ \
type square_sum = zero; \
\
for (int i = 0; i < size; i++) \
square_sum += x[i] * y[i]; \
\
return square_sum; \
}
SQUARE_SUM(f, float, 0.f)
SQUARE_SUM(d, double, 0.0)
#define XCORRELATE(suffix, type, zero, small, sqrtfun)\
static type xcorrelate_##suffix(const type *x, \
const type *y, \
type sumx, \
type sumy, int size) \
{ \
const type xm = sumx / size, ym = sumy / size; \
type num = zero, den, den0 = zero, den1 = zero; \
\
for (int i = 0; i < size; i++) { \
type xd = x[i] - xm; \
type yd = y[i] - ym; \
\
num += xd * yd; \
den0 += xd * xd; \
den1 += yd * yd; \
} \
\
num /= size; \
den = sqrtfun((den0 * den1) / size / size); \
\
return den <= small ? zero : num / den; \
}
XCORRELATE(f, float, 0.f, 1e-6f, sqrtf)
XCORRELATE(d, double, 0.0, 1e-9, sqrt)
#define XCORRELATE_SLOW(suffix, type) \
static int xcorrelate_slow_##suffix(AVFilterContext *ctx, \
AVFrame *out, int available) \
{ \
AudioXCorrelateContext *s = ctx->priv; \
const int size = s->size; \
int used; \
\
for (int ch = 0; ch < out->ch_layout.nb_channels; ch++) { \
const type *x = (const type *)s->cache[0]->extended_data[ch]; \
const type *y = (const type *)s->cache[1]->extended_data[ch]; \
type *sumx = (type *)s->mean_sum[0]->extended_data[ch]; \
type *sumy = (type *)s->mean_sum[1]->extended_data[ch]; \
type *dst = (type *)out->extended_data[ch]; \
\
used = s->used; \
if (!used) { \
sumx[0] = mean_sum_##suffix(x, size); \
sumy[0] = mean_sum_##suffix(y, size); \
used = 1; \
} \
\
for (int n = 0; n < out->nb_samples; n++) { \
const int idx = n + size; \
\
dst[n] = xcorrelate_##suffix(x + n, y + n, \
sumx[0], sumy[0],\
size); \
\
sumx[0] -= x[n]; \
sumx[0] += x[idx]; \
sumy[0] -= y[n]; \
sumy[0] += y[idx]; \
} \
} \
\
return used; \
}
XCORRELATE_SLOW(f, float)
XCORRELATE_SLOW(d, double)
#define clipf(x) (av_clipf(x, -1.f, 1.f))
#define clipd(x) (av_clipd(x, -1.0, 1.0))
#define XCORRELATE_FAST(suffix, type, zero, small, sqrtfun, CLIP) \
static int xcorrelate_fast_##suffix(AVFilterContext *ctx, AVFrame *out, \
int available) \
{ \
AudioXCorrelateContext *s = ctx->priv; \
const int size = s->size; \
int used; \
\
for (int ch = 0; ch < out->ch_layout.nb_channels; ch++) { \
const type *x = (const type *)s->cache[0]->extended_data[ch]; \
const type *y = (const type *)s->cache[1]->extended_data[ch]; \
type *num_sum = (type *)s->num_sum->extended_data[ch]; \
type *den_sumx = (type *)s->den_sum[0]->extended_data[ch]; \
type *den_sumy = (type *)s->den_sum[1]->extended_data[ch]; \
type *dst = (type *)out->extended_data[ch]; \
\
used = s->used; \
if (!used) { \
num_sum[0] = square_sum_##suffix(x, y, size); \
den_sumx[0] = square_sum_##suffix(x, x, size); \
den_sumy[0] = square_sum_##suffix(y, y, size); \
used = 1; \
} \
\
for (int n = 0; n < out->nb_samples; n++) { \
const int idx = n + size; \
type num, den; \
\
num = num_sum[0] / size; \
den = sqrtfun((den_sumx[0] * den_sumy[0]) / size / size); \
\
dst[n] = den <= small ? zero : CLIP(num / den); \
\
num_sum[0] -= x[n] * y[n]; \
num_sum[0] += x[idx] * y[idx]; \
den_sumx[0] -= x[n] * x[n]; \
den_sumx[0] += x[idx] * x[idx]; \
den_sumx[0] = FFMAX(den_sumx[0], zero); \
den_sumy[0] -= y[n] * y[n]; \
den_sumy[0] += y[idx] * y[idx]; \
den_sumy[0] = FFMAX(den_sumy[0], zero); \
} \
} \
\
return used; \
}
XCORRELATE_FAST(f, float, 0.f, 1e-6f, sqrtf, clipf)
XCORRELATE_FAST(d, double, 0.0, 1e-9, sqrt, clipd)
#define XCORRELATE_BEST(suffix, type, zero, small, sqrtfun, FMAX, CLIP) \
static int xcorrelate_best_##suffix(AVFilterContext *ctx, AVFrame *out, \
int available) \
{ \
AudioXCorrelateContext *s = ctx->priv; \
const int size = s->size; \
int used; \
\
for (int ch = 0; ch < out->ch_layout.nb_channels; ch++) { \
const type *x = (const type *)s->cache[0]->extended_data[ch]; \
const type *y = (const type *)s->cache[1]->extended_data[ch]; \
type *mean_sumx = (type *)s->mean_sum[0]->extended_data[ch]; \
type *mean_sumy = (type *)s->mean_sum[1]->extended_data[ch]; \
type *num_sum = (type *)s->num_sum->extended_data[ch]; \
type *den_sumx = (type *)s->den_sum[0]->extended_data[ch]; \
type *den_sumy = (type *)s->den_sum[1]->extended_data[ch]; \
type *dst = (type *)out->extended_data[ch]; \
\
used = s->used; \
if (!used) { \
num_sum[0] = square_sum_##suffix(x, y, size); \
den_sumx[0] = square_sum_##suffix(x, x, size); \
den_sumy[0] = square_sum_##suffix(y, y, size); \
mean_sumx[0] = mean_sum_##suffix(x, size); \
mean_sumy[0] = mean_sum_##suffix(y, size); \
used = 1; \
} \
\
for (int n = 0; n < out->nb_samples; n++) { \
const int idx = n + size; \
type num, den, xm, ym; \
\
xm = mean_sumx[0] / size; \
ym = mean_sumy[0] / size; \
num = num_sum[0] - size * xm * ym; \
den = sqrtfun(FMAX(den_sumx[0] - size * xm * xm, zero)) * \
sqrtfun(FMAX(den_sumy[0] - size * ym * ym, zero)); \
\
dst[n] = den <= small ? zero : CLIP(num / den); \
\
mean_sumx[0]-= x[n]; \
mean_sumx[0]+= x[idx]; \
mean_sumy[0]-= y[n]; \
mean_sumy[0]+= y[idx]; \
num_sum[0] -= x[n] * y[n]; \
num_sum[0] += x[idx] * y[idx]; \
den_sumx[0] -= x[n] * x[n]; \
den_sumx[0] += x[idx] * x[idx]; \
den_sumx[0] = FMAX(den_sumx[0], zero); \
den_sumy[0] -= y[n] * y[n]; \
den_sumy[0] += y[idx] * y[idx]; \
den_sumy[0] = FMAX(den_sumy[0], zero); \
} \
} \
\
return used; \
}
XCORRELATE_BEST(f, float, 0.f, 1e-6f, sqrtf, fmaxf, clipf)
XCORRELATE_BEST(d, double, 0.0, 1e-9, sqrt, fmax, clipd)
static int activate(AVFilterContext *ctx)
{
AudioXCorrelateContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *frame = NULL;
int ret, status;
int available;
int64_t pts;
FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
for (int i = 0; i < 2 && !s->eof; 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(outlink, 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(outlink, 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(outlink, out_samples);
if (!out)
return AVERROR(ENOMEM);
s->used = s->xcorrelate(ctx, out, available);
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(outlink, out);
}
for (int i = 0; i < 2 && !s->eof; i++) {
if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
AVFrame *silence = ff_get_audio_buffer(outlink, s->size);
s->eof = 1;
if (!silence)
return AVERROR(ENOMEM);
av_audio_fifo_write(s->fifo[0], (void **)silence->extended_data,
silence->nb_samples);
av_audio_fifo_write(s->fifo[1], (void **)silence->extended_data,
silence->nb_samples);
av_frame_free(&silence);
}
}
if (s->eof &&
(av_audio_fifo_size(s->fifo[0]) <= s->size ||
av_audio_fifo_size(s->fifo[1]) <= s->size)) {
ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
return 0;
}
if ((av_audio_fifo_size(s->fifo[0]) > s->size &&
av_audio_fifo_size(s->fifo[1]) > s->size) || s->eof) {
ff_filter_set_ready(ctx, 10);
return 0;
}
if (ff_outlink_frame_wanted(outlink) && !s->eof) {
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;
AudioXCorrelateContext *s = ctx->priv;
s->pts = AV_NOPTS_VALUE;
s->fifo[0] = av_audio_fifo_alloc(outlink->format, outlink->ch_layout.nb_channels, s->size);
s->fifo[1] = av_audio_fifo_alloc(outlink->format, outlink->ch_layout.nb_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_f; break;
case 1: s->xcorrelate = xcorrelate_fast_f; break;
case 2: s->xcorrelate = xcorrelate_best_f; break;
}
if (outlink->format == AV_SAMPLE_FMT_DBLP) {
switch (s->algo) {
case 0: s->xcorrelate = xcorrelate_slow_d; break;
case 1: s->xcorrelate = xcorrelate_fast_d; break;
case 2: s->xcorrelate = xcorrelate_best_d; 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,
},
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
#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 the segment size", OFFSET(size), AV_OPT_TYPE_INT, {.i64=256}, 2, 131072, AF },
{ "algo", "set the algorithm", OFFSET(algo), AV_OPT_TYPE_INT, {.i64=2}, 0, 2, AF, .unit = "algo" },
{ "slow", "slow algorithm", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "algo" },
{ "fast", "fast algorithm", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "algo" },
{ "best", "best algorithm", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "algo" },
{ NULL }
};
AVFILTER_DEFINE_CLASS(axcorrelate);
const AVFilter ff_af_axcorrelate = {
.name = "axcorrelate",
.description = NULL_IF_CONFIG_SMALL("Cross-correlate two audio streams."),
.priv_size = sizeof(AudioXCorrelateContext),
.priv_class = &axcorrelate_class,
.activate = activate,
.uninit = uninit,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP),
};