[FFmpeg-cvslog] avfilter/af_acrossover: add higher orders
Paul B Mahol
git at videolan.org
Thu Nov 26 14:18:25 EET 2020
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Wed Nov 25 20:56:17 2020 +0100| [aba17bf6a5ee0e9a584513301597d229647d3765] | committer: Paul B Mahol
avfilter/af_acrossover: add higher orders
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=aba17bf6a5ee0e9a584513301597d229647d3765
---
libavfilter/af_acrossover.c | 104 ++++++++++++++++++++------------------------
1 file changed, 48 insertions(+), 56 deletions(-)
diff --git a/libavfilter/af_acrossover.c b/libavfilter/af_acrossover.c
index 002f378c3d..4c4bbbc76e 100644
--- a/libavfilter/af_acrossover.c
+++ b/libavfilter/af_acrossover.c
@@ -46,16 +46,17 @@ typedef struct BiquadContext {
} BiquadContext;
typedef struct CrossoverChannel {
- BiquadContext lp[MAX_BANDS][4];
- BiquadContext hp[MAX_BANDS][4];
+ BiquadContext lp[MAX_BANDS][16];
+ BiquadContext hp[MAX_BANDS][16];
} CrossoverChannel;
typedef struct AudioCrossoverContext {
const AVClass *class;
char *splits_str;
- int order;
+ int order_opt;
+ int order;
int filter_count;
int nb_splits;
float *splits;
@@ -71,10 +72,12 @@ 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), AV_OPT_TYPE_INT, {.i64=1}, 0, 2, AF, "m" },
+ { "order", "set order", OFFSET(order_opt), AV_OPT_TYPE_INT, {.i64=1}, 0, 4, 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" },
{ NULL }
};
@@ -139,75 +142,64 @@ static av_cold int init(AVFilterContext *ctx)
static void set_lp(BiquadContext *b, double fc, double q, double sr)
{
- double omega = 2.0 * M_PI * fc / sr;
- double sn = sin(omega);
- double cs = cos(omega);
- double alpha = sn / (2. * q);
- double inv = 1.0 / (1.0 + alpha);
-
- b->a0 = (1. - cs) * 0.5 * inv;
- b->a1 = (1. - cs) * inv;
- b->a2 = b->a0;
- b->b1 = -2. * cs * inv;
- b->b2 = (1. - alpha) * inv;
+ double thetac = 2.0 * M_PI * fc / sr;
+ double d = 1.0 / q;
+ double beta = 0.5 * (1.0 - (d / 2.0) * sin(thetac)) / (1.0 + (d / 2.0) * sin(thetac));
+ double gamma = (0.5 + beta) * cos(thetac);
+
+ b->a0 = (0.5 + beta - gamma) / 2.0;
+ b->a1 = 0.5 + beta - gamma;
+ b->a2 = b->a1 / 2.0;
+ b->b1 = -2.0 * gamma;
+ b->b2 = 2.0 * beta;
}
static void set_hp(BiquadContext *b, double fc, double q, double sr)
{
- double omega = 2 * M_PI * fc / sr;
- double sn = sin(omega);
- double cs = cos(omega);
- double alpha = sn / (2 * q);
- double inv = 1.0 / (1.0 + alpha);
-
- b->a0 = inv * (1. + cs) / 2.;
- b->a1 = -2. * b->a0;
+ double thetac = 2.0 * M_PI * fc / sr;
+ double d = 1.0 / q;
+ double beta = 0.5 * (1.0 - (d / 2.0) * sin(thetac)) / (1.0 + (d / 2.0) * sin(thetac));
+ double gamma = (0.5 + beta) * cos(thetac);
+
+ b->a0 = (0.5 + beta + gamma) / 2.0;
+ b->a1 = -(0.5 + beta + gamma);
b->a2 = b->a0;
- b->b1 = -2. * cs * inv;
- b->b2 = (1. - alpha) * inv;
+ b->b1 = -2.0 * gamma;
+ b->b2 = 2.0 * beta;
+}
+
+static void calc_q_factors(int order, double *q)
+{
+ int num = 1, den = 4 * order;
+
+ for (int i = 0; i < order; i++) {
+ q[i] = fabs(1. / (2. * cos(num * M_PI / den)));
+ num += 2;
+ }
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioCrossoverContext *s = ctx->priv;
- int ch, band, sample_rate = inlink->sample_rate;
- double q;
+ int sample_rate = inlink->sample_rate;
+ double q[16] = { 0.5 };
s->xover = av_calloc(inlink->channels, sizeof(*s->xover));
if (!s->xover)
return AVERROR(ENOMEM);
- switch (s->order) {
- case 0:
- q = 0.5;
- s->filter_count = 1;
- break;
- case 1:
- q = M_SQRT1_2;
- s->filter_count = 2;
- break;
- case 2:
- q = 0.54;
- s->filter_count = 4;
- break;
- }
+ s->order = FFMAX(2, s->order_opt * 4);
+ s->filter_count = s->order / 2;
+ calc_q_factors(s->filter_count / 2, q + (s->order == 2));
+
+ 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;
- for (ch = 0; ch < inlink->channels; ch++) {
- for (band = 0; band <= s->nb_splits; band++) {
- set_lp(&s->xover[ch].lp[band][0], s->splits[band], q, sample_rate);
- set_hp(&s->xover[ch].hp[band][0], s->splits[band], q, sample_rate);
-
- if (s->order > 1) {
- set_lp(&s->xover[ch].lp[band][1], s->splits[band], 1.34, sample_rate);
- set_hp(&s->xover[ch].hp[band][1], s->splits[band], 1.34, sample_rate);
- set_lp(&s->xover[ch].lp[band][2], s->splits[band], q, sample_rate);
- set_hp(&s->xover[ch].hp[band][2], s->splits[band], q, sample_rate);
- set_lp(&s->xover[ch].lp[band][3], s->splits[band], 1.34, sample_rate);
- set_hp(&s->xover[ch].hp[band][3], s->splits[band], 1.34, sample_rate);
- } else {
- set_lp(&s->xover[ch].lp[band][1], s->splits[band], q, sample_rate);
- set_hp(&s->xover[ch].hp[band][1], s->splits[band], q, sample_rate);
+ 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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