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Code Issues Releases Activity

avfilter/af_axcorrelate: add another algorithm for calculation

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Rewrite EOF logic while here.
This commit is contained in:
Paul B Mahol
2023-07-15 23:13:44 +02:00
parent f032234953
commit c183f840fa
2 changed files with 114 additions and 33 deletions
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4
doc/filters.texi
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@@ -3762,8 +3762,8 @@ Set size of segment over which cross-correlation is calculated.
Default is 256. Allowed range is from 2 to 131072. Default is 256. Allowed range is from 2 to 131072.
@item algo @item algo
Set algorithm for cross-correlation. Can be @code{slow} or @code{fast}. Set algorithm for cross-correlation. Can be @code{slow} or @code{fast} or @code{best}.
Default is @code{slow}. Fast algorithm assumes mean values over any given segment Default is @code{best}. Fast algorithm assumes mean values over any given segment
are always zero and thus need much less calculations to make. are always zero and thus need much less calculations to make.
This is generally not true, but is valid for typical audio streams. This is generally not true, but is valid for typical audio streams.
@end table @end table

129
libavfilter/af_axcorrelate.c
View File

@@ -110,7 +110,7 @@ static int xcorrelate_slow_##suffix(AVFilterContext *ctx, \
AVFrame *out, int available) \ AVFrame *out, int available) \
{ \ { \
AudioXCorrelateContext *s = ctx->priv; \ AudioXCorrelateContext *s = ctx->priv; \
const int size = FFMIN(available, s->size); \ const int size = s->size; \
int used; \ int used; \
\ \
for (int ch = 0; ch < out->ch_layout.nb_channels; ch++) { \ for (int ch = 0; ch < out->ch_layout.nb_channels; ch++) { \
@@ -128,10 +128,10 @@ static int xcorrelate_slow_##suffix(AVFilterContext *ctx, \
} \ } \
\ \
for (int n = 0; n < out->nb_samples; n++) { \ for (int n = 0; n < out->nb_samples; n++) { \
const int idx = available <= s->size ? out->nb_samples - n - 1 : n + size; \ const int idx = n + size; \
\ \
dst[n] = xcorrelate_##suffix(x + n, y + n, \ dst[n] = xcorrelate_##suffix(x + n, y + n, \
sumx[0], sumy[0], \ sumx[0], sumy[0],\
size); \ size); \
\ \
sumx[0] -= x[n]; \ sumx[0] -= x[n]; \
@@ -147,12 +147,15 @@ static int xcorrelate_slow_##suffix(AVFilterContext *ctx, \
XCORRELATE_SLOW(f, float) XCORRELATE_SLOW(f, float)
XCORRELATE_SLOW(d, double) XCORRELATE_SLOW(d, double)
#define XCORRELATE_FAST(suffix, type, zero, small, sqrtfun) \ #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, \ static int xcorrelate_fast_##suffix(AVFilterContext *ctx, AVFrame *out, \
int available) \ int available) \
{ \ { \
AudioXCorrelateContext *s = ctx->priv; \ AudioXCorrelateContext *s = ctx->priv; \
const int size = FFMIN(available, s->size); \ const int size = s->size; \
int used; \ int used; \
\ \
for (int ch = 0; ch < out->ch_layout.nb_channels; ch++) { \ for (int ch = 0; ch < out->ch_layout.nb_channels; ch++) { \
@@ -172,13 +175,13 @@ static int xcorrelate_fast_##suffix(AVFilterContext *ctx, AVFrame *out, \
} \ } \
\ \
for (int n = 0; n < out->nb_samples; n++) { \ for (int n = 0; n < out->nb_samples; n++) { \
const int idx = available <= s->size ? out->nb_samples - n - 1 : n + size; \ const int idx = n + size; \
type num, den; \ type num, den; \
\ \
num = num_sum[0] / size; \ num = num_sum[0] / size; \
den = sqrtfun((den_sumx[0] * den_sumy[0]) / size / size); \ den = sqrtfun((den_sumx[0] * den_sumy[0]) / size / size); \
\ \
dst[n] = den <= small ? zero : num / den; \ dst[n] = den <= small ? zero : CLIP(num / den); \
\ \
num_sum[0] -= x[n] * y[n]; \ num_sum[0] -= x[n] * y[n]; \
num_sum[0] += x[idx] * y[idx]; \ num_sum[0] += x[idx] * y[idx]; \
@@ -194,20 +197,82 @@ static int xcorrelate_fast_##suffix(AVFilterContext *ctx, AVFrame *out, \
return used; \ return used; \
} }
XCORRELATE_FAST(f, float, 0.f, 1e-6f, sqrtf) XCORRELATE_FAST(f, float, 0.f, 1e-6f, sqrtf, clipf)
XCORRELATE_FAST(d, double, 0.0, 1e-9, sqrt) 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) static int activate(AVFilterContext *ctx)
{ {
AudioXCorrelateContext *s = ctx->priv; AudioXCorrelateContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *frame = NULL; AVFrame *frame = NULL;
int ret, status; int ret, status;
int available; int available;
int64_t pts; int64_t pts;
FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx); FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
for (int i = 0; i < 2; i++) { for (int i = 0; i < 2 && !s->eof; i++) {
ret = ff_inlink_consume_frame(ctx->inputs[i], &frame); ret = ff_inlink_consume_frame(ctx->inputs[i], &frame);
if (ret > 0) { if (ret > 0) {
if (s->pts == AV_NOPTS_VALUE) if (s->pts == AV_NOPTS_VALUE)
@@ -221,20 +286,20 @@ static int activate(AVFilterContext *ctx)
} }
available = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1])); available = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1]));
if (available > s->size || (s->eof && available > 0)) { if (available > s->size) {
const int out_samples = s->eof ? available : available - s->size; const int out_samples = available - s->size;
AVFrame *out; AVFrame *out;
if (!s->cache[0] || s->cache[0]->nb_samples < available) { if (!s->cache[0] || s->cache[0]->nb_samples < available) {
av_frame_free(&s->cache[0]); av_frame_free(&s->cache[0]);
s->cache[0] = ff_get_audio_buffer(ctx->outputs[0], available); s->cache[0] = ff_get_audio_buffer(outlink, available);
if (!s->cache[0]) if (!s->cache[0])
return AVERROR(ENOMEM); return AVERROR(ENOMEM);
} }
if (!s->cache[1] || s->cache[1]->nb_samples < available) { if (!s->cache[1] || s->cache[1]->nb_samples < available) {
av_frame_free(&s->cache[1]); av_frame_free(&s->cache[1]);
s->cache[1] = ff_get_audio_buffer(ctx->outputs[0], available); s->cache[1] = ff_get_audio_buffer(outlink, available);
if (!s->cache[1]) if (!s->cache[1])
return AVERROR(ENOMEM); return AVERROR(ENOMEM);
} }
@@ -247,7 +312,7 @@ static int activate(AVFilterContext *ctx)
if (ret < 0) if (ret < 0)
return ret; return ret;
out = ff_get_audio_buffer(ctx->outputs[0], out_samples); out = ff_get_audio_buffer(outlink, out_samples);
if (!out) if (!out)
return AVERROR(ENOMEM); return AVERROR(ENOMEM);
@@ -259,18 +324,31 @@ static int activate(AVFilterContext *ctx)
av_audio_fifo_drain(s->fifo[0], out_samples); av_audio_fifo_drain(s->fifo[0], out_samples);
av_audio_fifo_drain(s->fifo[1], out_samples); av_audio_fifo_drain(s->fifo[1], out_samples);
return ff_filter_frame(ctx->outputs[0], out); return ff_filter_frame(outlink, out);
} }
for (int i = 0; i < 2 && !s->eof; i++) { for (int i = 0; i < 2 && !s->eof; i++) {
if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
AVFrame *silence = ff_get_audio_buffer(outlink, s->size);
s->eof = 1; 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 && if (s->eof &&
(av_audio_fifo_size(s->fifo[0]) <= 0 || (av_audio_fifo_size(s->fifo[0]) <= s->size ||
av_audio_fifo_size(s->fifo[1]) <= 0)) { av_audio_fifo_size(s->fifo[1]) <= s->size)) {
ff_outlink_set_status(ctx->outputs[0], AVERROR_EOF, s->pts); ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
return 0; return 0;
} }
@@ -280,7 +358,7 @@ static int activate(AVFilterContext *ctx)
return 0; return 0;
} }
if (ff_outlink_frame_wanted(ctx->outputs[0]) && !s->eof) { if (ff_outlink_frame_wanted(outlink) && !s->eof) {
for (int i = 0; i < 2; i++) { for (int i = 0; i < 2; i++) {
if (av_audio_fifo_size(s->fifo[i]) > s->size) if (av_audio_fifo_size(s->fifo[i]) > s->size)
continue; continue;
@@ -316,12 +394,14 @@ static int config_output(AVFilterLink *outlink)
switch (s->algo) { switch (s->algo) {
case 0: s->xcorrelate = xcorrelate_slow_f; break; case 0: s->xcorrelate = xcorrelate_slow_f; break;
case 1: s->xcorrelate = xcorrelate_fast_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) { if (outlink->format == AV_SAMPLE_FMT_DBLP) {
switch (s->algo) { switch (s->algo) {
case 0: s->xcorrelate = xcorrelate_slow_d; break; case 0: s->xcorrelate = xcorrelate_slow_d; break;
case 1: s->xcorrelate = xcorrelate_fast_d; break; case 1: s->xcorrelate = xcorrelate_fast_d; break;
case 2: s->xcorrelate = xcorrelate_best_d; break;
} }
} }
@@ -366,10 +446,11 @@ static const AVFilterPad outputs[] = {
#define OFFSET(x) offsetof(AudioXCorrelateContext, x) #define OFFSET(x) offsetof(AudioXCorrelateContext, x)
static const AVOption axcorrelate_options[] = { static const AVOption axcorrelate_options[] = {
{ "size", "set segment size", OFFSET(size), AV_OPT_TYPE_INT, {.i64=256}, 2, 131072, AF }, { "size", "set the segment size", OFFSET(size), AV_OPT_TYPE_INT, {.i64=256}, 2, 131072, AF },
{ "algo", "set algorithm", OFFSET(algo), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, "algo" }, { "algo", "set the algorithm", OFFSET(algo), AV_OPT_TYPE_INT, {.i64=2}, 0, 2, AF, "algo" },
{ "slow", "slow algorithm", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, 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" }, { "fast", "fast algorithm", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "algo" },
{ "best", "best algorithm", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "algo" },
{ NULL } { NULL }
}; };