[FFmpeg-cvslog] avfilter/af_acrossover: add even more orders
Paul B Mahol
git at videolan.org
Thu Nov 26 14:18:32 EET 2020
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Thu Nov 26 11:19:51 2020 +0100| [cc74700a83a918e4fc805528d9e158de156078fc] | committer: Paul B Mahol
avfilter/af_acrossover: add even more orders
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=cc74700a83a918e4fc805528d9e158de156078fc
---
doc/filters.texi | 16 +++++++++++++++-
libavfilter/af_acrossover.c | 42 ++++++++++++++++++++++++++----------------
2 files changed, 41 insertions(+), 17 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index 229cf42f8f..109cacc23d 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -523,7 +523,21 @@ The filter accepts the following options:
Set split frequencies. Those must be positive and increasing.
@item order
-Set filter order, can be @var{2nd}, @var{4th} or @var{8th}.
+Set filter order. Availabe values are:
+
+ at table @samp
+ at item 2nd
+ at item 4th
+ at item 6th
+ at item 8th
+ at item 10th
+ at item 12th
+ at item 14th
+ at item 16th
+ at item 18th
+ at item 20th
+ at end table
+
Default is @var{4th}.
@end table
diff --git a/libavfilter/af_acrossover.c b/libavfilter/af_acrossover.c
index f8c4d1f25a..3f32f403f0 100644
--- a/libavfilter/af_acrossover.c
+++ b/libavfilter/af_acrossover.c
@@ -45,8 +45,8 @@ typedef struct BiquadContext {
} BiquadContext;
typedef struct CrossoverChannel {
- BiquadContext lp[MAX_BANDS][16];
- BiquadContext hp[MAX_BANDS][16];
+ BiquadContext lp[MAX_BANDS][20];
+ BiquadContext hp[MAX_BANDS][20];
} CrossoverChannel;
typedef struct AudioCrossoverContext {
@@ -71,12 +71,17 @@ typedef struct AudioCrossoverContext {
static const AVOption acrossover_options[] = {
{ "split", "set split frequencies", OFFSET(splits_str), AV_OPT_TYPE_STRING, {.str="500"}, 0, 0, AF },
- { "order", "set order", OFFSET(order_opt), AV_OPT_TYPE_INT, {.i64=1}, 0, 4, AF, "m" },
+ { "order", "set order", OFFSET(order_opt), AV_OPT_TYPE_INT, {.i64=1}, 0, 9, AF, "m" },
{ "2nd", "2nd order", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "m" },
{ "4th", "4th order", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "m" },
- { "8th", "8th order", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "m" },
- { "12th", "12th order", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "m" },
- { "16th", "16th order", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, "m" },
+ { "6th", "6th order", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "m" },
+ { "8th", "8th order", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "m" },
+ { "10th", "10th order", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, "m" },
+ { "12th", "12th order", 0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, AF, "m" },
+ { "14th", "14th order", 0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, AF, "m" },
+ { "16th", "16th order", 0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, AF, "m" },
+ { "18th", "18th order", 0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, AF, "m" },
+ { "20th", "20th order", 0, AV_OPT_TYPE_CONST, {.i64=9}, 0, 0, AF, "m" },
{ NULL }
};
@@ -169,12 +174,10 @@ static void set_hp(BiquadContext *b, double fc, double q, double sr)
static void calc_q_factors(int order, double *q)
{
- int num = 1, den = 4 * order;
+ double n = order / 2.;
- for (int i = 0; i < order; i++) {
- q[i] = fabs(1. / (2. * cos(num * M_PI / den)));
- num += 2;
- }
+ for (int i = 0; i < n / 2; i++)
+ q[i] = 1. / (-2. * cos(M_PI * (2. * (i + 1) + n - 1.) / (2. * n)));
}
static int config_input(AVFilterLink *inlink)
@@ -182,20 +185,27 @@ static int config_input(AVFilterLink *inlink)
AVFilterContext *ctx = inlink->dst;
AudioCrossoverContext *s = ctx->priv;
int sample_rate = inlink->sample_rate;
- double q[16] = { 0.5 };
+ int first_order;
+ double q[16];
s->xover = av_calloc(inlink->channels, sizeof(*s->xover));
if (!s->xover)
return AVERROR(ENOMEM);
- s->order = FFMAX(2, s->order_opt * 4);
+ s->order = (s->order_opt + 1) * 2;
s->filter_count = s->order / 2;
- calc_q_factors(s->filter_count / 2, q + (s->order == 2));
+ first_order = s->filter_count & 1;
+ calc_q_factors(s->order, q);
for (int ch = 0; ch < inlink->channels; ch++) {
for (int band = 0; band <= s->nb_splits; band++) {
- for (int n = 0; n < s->filter_count; n++) {
- const int idx = (n + (s->order == 2)) / 2;
+ if (first_order) {
+ set_lp(&s->xover[ch].lp[band][0], s->splits[band], 0.5, sample_rate);
+ set_hp(&s->xover[ch].hp[band][0], s->splits[band], 0.5, sample_rate);
+ }
+
+ for (int n = first_order; n < s->filter_count; n++) {
+ const int idx = s->filter_count / 2 - ((n + first_order) / 2 - first_order) - 1;
set_lp(&s->xover[ch].lp[band][n], s->splits[band], q[idx], sample_rate);
set_hp(&s->xover[ch].hp[band][n], s->splits[band], q[idx], sample_rate);
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