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channelmix: improve corssover processing

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Apply volume after crossover, optimize when disabled or silent.
This commit is contained in:
Wim Taymans 2022-02-27 20:41:44 +01:00
parent 168511c157
commit d62d2764e8
5 changed files with 52 additions and 43 deletions
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48
spa/plugins/audioconvert/channelmix-ops-c.c
View file

@ -78,10 +78,8 @@ channelmix_f32_n_m_c(struct channelmix *mix, void * SPA_RESTRICT dst[],
d[i][n] = sum; d[i][n] = sum;
} }
} }
for (i = 0; i < n_dst; i++) { for (i = 0; i < n_dst; i++)
if (mix->lr4_info[i] > 0) lr4_process(&mix->lr4[i], d[i], d[i], 1.0f, n_samples);
lr4_process(&mix->lr4[i], d[i], n_samples);
}
} }
} }
@ -253,22 +251,20 @@ channelmix_f32_2_3p1_c(struct channelmix *mix, void * SPA_RESTRICT dst[],
float c = s[0][n] + s[1][n]; float c = s[0][n] + s[1][n];
d[0][n] = s[0][n]; d[0][n] = s[0][n];
d[1][n] = s[1][n]; d[1][n] = s[1][n];
d[2][n] = c * v2; d[2][n] = c;
d[3][n] = c * v3;
} }
if (v3 > 0.0f) lr4_process(&mix->lr4[3], d[3], d[2], v3, n_samples);
lr4_process(&mix->lr4[3], d[3], n_samples); lr4_process(&mix->lr4[2], d[2], d[2], v2, n_samples);
} }
else { else {
for (n = 0; n < n_samples; n++) { for (n = 0; n < n_samples; n++) {
float c = s[0][n] + s[1][n]; float c = s[0][n] + s[1][n];
d[0][n] = s[0][n] * v0; d[0][n] = s[0][n] * v0;
d[1][n] = s[1][n] * v1; d[1][n] = s[1][n] * v1;
d[2][n] = c * v2; d[2][n] = c;
d[3][n] = c * v3;
} }
if (v3 > 0.0f) lr4_process(&mix->lr4[3], d[3], d[2], v3, n_samples);
lr4_process(&mix->lr4[3], d[3], n_samples); lr4_process(&mix->lr4[2], d[2], d[2], v2, n_samples);
} }
} }
@ -296,24 +292,22 @@ channelmix_f32_2_5p1_c(struct channelmix *mix, void * SPA_RESTRICT dst[],
float c = s[0][n] + s[1][n]; float c = s[0][n] + s[1][n];
d[0][n] = d[4][n] = s[0][n]; d[0][n] = d[4][n] = s[0][n];
d[1][n] = d[5][n] = s[1][n]; d[1][n] = d[5][n] = s[1][n];
d[2][n] = c * v2; d[2][n] = c;
d[3][n] = c * v3;
} }
if (v3 > 0.0f) lr4_process(&mix->lr4[3], d[3], d[2], v3, n_samples);
lr4_process(&mix->lr4[3], d[3], n_samples); lr4_process(&mix->lr4[2], d[2], d[2], v2, n_samples);
} }
else { else {
for (n = 0; n < n_samples; n++) { for (n = 0; n < n_samples; n++) {
float c = s[0][n] + s[1][n]; float c = s[0][n] + s[1][n];
d[0][n] = s[0][n] * v0; d[0][n] = s[0][n] * v0;
d[1][n] = s[1][n] * v1; d[1][n] = s[1][n] * v1;
d[2][n] = c * v2; d[2][n] = c;
d[3][n] = c * v3;
d[4][n] = s[0][n] * v4; d[4][n] = s[0][n] * v4;
d[5][n] = s[1][n] * v5; d[5][n] = s[1][n] * v5;
} }
if (v3 > 0.0f) lr4_process(&mix->lr4[3], d[3], d[2], v3, n_samples);
lr4_process(&mix->lr4[3], d[3], n_samples); lr4_process(&mix->lr4[2], d[2], d[2], v2, n_samples);
} }
} }
@ -343,26 +337,24 @@ channelmix_f32_2_7p1_c(struct channelmix *mix, void * SPA_RESTRICT dst[],
float c = s[0][n] + s[1][n]; float c = s[0][n] + s[1][n];
d[0][n] = d[4][n] = d[6][n] = s[0][n]; d[0][n] = d[4][n] = d[6][n] = s[0][n];
d[1][n] = d[5][n] = d[7][n] = s[1][n]; d[1][n] = d[5][n] = d[7][n] = s[1][n];
d[2][n] = c * v2; d[2][n] = c;
d[3][n] = c * v3;
} }
if (v3 > 0.0f) lr4_process(&mix->lr4[3], d[3], d[2], v3, n_samples);
lr4_process(&mix->lr4[3], d[3], n_samples); lr4_process(&mix->lr4[2], d[2], d[2], v2, n_samples);
} }
else { else {
for (n = 0; n < n_samples; n++) { for (n = 0; n < n_samples; n++) {
float c = s[0][n] + s[1][n]; float c = s[0][n] + s[1][n];
d[0][n] = s[0][n] * v0; d[0][n] = s[0][n] * v0;
d[1][n] = s[1][n] * v1; d[1][n] = s[1][n] * v1;
d[2][n] = c * v2; d[2][n] = c;
d[3][n] = c * v3;
d[4][n] = s[0][n] * v4; d[4][n] = s[0][n] * v4;
d[5][n] = s[1][n] * v5; d[5][n] = s[1][n] * v5;
d[6][n] = s[0][n] * v6; d[6][n] = s[0][n] * v6;
d[7][n] = s[1][n] * v7; d[7][n] = s[1][n] * v7;
} }
if (v3 > 0.0f) lr4_process(&mix->lr4[3], d[3], d[2], v3, n_samples);
lr4_process(&mix->lr4[3], d[3], n_samples); lr4_process(&mix->lr4[2], d[2], d[2], v2, n_samples);
} }
} }

3
spa/plugins/audioconvert/channelmix-ops.c
View file

@ -457,9 +457,8 @@ done:
if (i == _CH(LFE) && mix->lfe_cutoff > 0.0f) { if (i == _CH(LFE) && mix->lfe_cutoff > 0.0f) {
spa_log_debug(mix->log, "channel %d is LFE", ic); spa_log_debug(mix->log, "channel %d is LFE", ic);
lr4_set(&mix->lr4[ic], BQ_LOWPASS, mix->lfe_cutoff / mix->freq); lr4_set(&mix->lr4[ic], BQ_LOWPASS, mix->lfe_cutoff / mix->freq);
mix->lr4_info[ic] = 1;
} else { } else {
mix->lr4_info[ic] = 0; mix->lr4[ic].active = false;
} }
ic++; ic++;
} }

1
spa/plugins/audioconvert/channelmix-ops.h
View file

@ -69,7 +69,6 @@ struct channelmix {
float freq; /* sample frequency */ float freq; /* sample frequency */
float lfe_cutoff; /* in Hz, 0 is disabled */ float lfe_cutoff; /* in Hz, 0 is disabled */
uint32_t lr4_info[SPA_AUDIO_MAX_CHANNELS];
struct lr4 lr4[SPA_AUDIO_MAX_CHANNELS]; struct lr4 lr4[SPA_AUDIO_MAX_CHANNELS];
void (*process) (struct channelmix *mix, void * SPA_RESTRICT dst[], void (*process) (struct channelmix *mix, void * SPA_RESTRICT dst[],

38
spa/plugins/audioconvert/crossover.c
View file

@ -5,6 +5,9 @@
#include "config.h" #include "config.h"
#include <float.h>
#include <string.h>
#include "crossover.h" #include "crossover.h"
void lr4_set(struct lr4 *lr4, enum biquad_type type, float freq) void lr4_set(struct lr4 *lr4, enum biquad_type type, float freq)
@ -16,9 +19,10 @@ void lr4_set(struct lr4 *lr4, enum biquad_type type, float freq)
lr4->y2 = 0; lr4->y2 = 0;
lr4->z1 = 0; lr4->z1 = 0;
lr4->z2 = 0; lr4->z2 = 0;
lr4->active = true;
} }
void lr4_process(struct lr4 *lr4, float *data, int samples) void lr4_process(struct lr4 *lr4, float *dst, const float *src, const float vol, int samples)
{ {
float lx1 = lr4->x1; float lx1 = lr4->x1;
float lx2 = lr4->x2; float lx2 = lr4->x2;
@ -31,11 +35,22 @@ void lr4_process(struct lr4 *lr4, float *data, int samples)
float lb2 = lr4->bq.b2; float lb2 = lr4->bq.b2;
float la1 = lr4->bq.a1; float la1 = lr4->bq.a1;
float la2 = lr4->bq.a2; float la2 = lr4->bq.a2;
int i; int i;
if (vol == 0.0f) {
memset(dst, 0, samples * sizeof(float));
return;
} else if (!lr4->active) {
if (src != dst || vol != 1.0f) {
for (i = 0; i < samples; i++)
dst[i] = src[i] * vol;
}
return;
}
for (i = 0; i < samples; i++) { for (i = 0; i < samples; i++) {
float x, y, z; float x, y, z;
x = data[i]; x = src[i];
y = lb0*x + lb1*lx1 + lb2*lx2 - la1*ly1 - la2*ly2; y = lb0*x + lb1*lx1 + lb2*lx2 - la1*ly1 - la2*ly2;
z = lb0*y + lb1*ly1 + lb2*ly2 - la1*lz1 - la2*lz2; z = lb0*y + lb1*ly1 + lb2*ly2 - la1*lz1 - la2*lz2;
lx2 = lx1; lx2 = lx1;
@ -44,13 +59,14 @@ void lr4_process(struct lr4 *lr4, float *data, int samples)
ly1 = y; ly1 = y;
lz2 = lz1; lz2 = lz1;
lz1 = z; lz1 = z;
data[i] = z; dst[i] = z * vol;
} }
#define F(x) (-FLT_MIN < (x) && (x) < FLT_MIN ? 0.0f : (x))
lr4->x1 = lx1; lr4->x1 = F(lx1);
lr4->x2 = lx2; lr4->x2 = F(lx2);
lr4->y1 = ly1; lr4->y1 = F(ly1);
lr4->y2 = ly2; lr4->y2 = F(ly2);
lr4->z1 = lz1; lr4->z1 = F(lz1);
lr4->z2 = lz2; lr4->z2 = F(lz2);
#undef F
} }

5
spa/plugins/audioconvert/crossover.h
View file

@ -6,6 +6,8 @@
#ifndef CROSSOVER_H_ #ifndef CROSSOVER_H_
#define CROSSOVER_H_ #define CROSSOVER_H_
#include <stdbool.h>
#include "biquad.h" #include "biquad.h"
/* An LR4 filter is two biquads with the same parameters connected in series: /* An LR4 filter is two biquads with the same parameters connected in series:
* *
@ -19,10 +21,11 @@ struct lr4 {
float x1, x2; float x1, x2;
float y1, y2; float y1, y2;
float z1, z2; float z1, z2;
bool active;
}; };
void lr4_set(struct lr4 *lr4, enum biquad_type type, float freq); void lr4_set(struct lr4 *lr4, enum biquad_type type, float freq);
void lr4_process(struct lr4 *lr4, float *data, int samples); void lr4_process(struct lr4 *lr4, float *dst, const float *src, const float vol, int samples);
#endif /* CROSSOVER_H_ */ #endif /* CROSSOVER_H_ */