diff --git a/spa/plugins/audioconvert/channelmix-ops-c.c b/spa/plugins/audioconvert/channelmix-ops-c.c
index 6bf92763e..63947aa5f 100644
--- a/spa/plugins/audioconvert/channelmix-ops-c.c
+++ b/spa/plugins/audioconvert/channelmix-ops-c.c
@@ -78,10 +78,8 @@ channelmix_f32_n_m_c(struct channelmix *mix, void * SPA_RESTRICT dst[],
 				d[i][n] = sum;
 			}
 		}
-		for (i = 0; i < n_dst; i++) {
-			if (mix->lr4_info[i] > 0)
-				lr4_process(&mix->lr4[i], d[i], n_samples);
-		}
+		for (i = 0; i < n_dst; i++)
+			lr4_process(&mix->lr4[i], d[i], d[i], 1.0f, 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];
 			d[0][n] = s[0][n];
 			d[1][n] = s[1][n];
-			d[2][n] = c * v2;
-			d[3][n] = c * v3;
+			d[2][n] = c;
 		}
-		if (v3 > 0.0f)
-			lr4_process(&mix->lr4[3], d[3], n_samples);
+		lr4_process(&mix->lr4[3], d[3], d[2], v3, n_samples);
+		lr4_process(&mix->lr4[2], d[2], d[2], v2, n_samples);
 	}
 	else {
 		for (n = 0; n < n_samples; n++) {
 			float c = s[0][n] + s[1][n];
 			d[0][n] = s[0][n] * v0;
 			d[1][n] = s[1][n] * v1;
-			d[2][n] = c * v2;
-			d[3][n] = c * v3;
+			d[2][n] = c;
 		}
-		if (v3 > 0.0f)
-			lr4_process(&mix->lr4[3], d[3], n_samples);
+		lr4_process(&mix->lr4[3], d[3], d[2], v3, 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];
 			d[0][n] = d[4][n] = s[0][n];
 			d[1][n] = d[5][n] = s[1][n];
-			d[2][n] = c * v2;
-			d[3][n] = c * v3;
+			d[2][n] = c;
 		}
-		if (v3 > 0.0f)
-			lr4_process(&mix->lr4[3], d[3], n_samples);
+		lr4_process(&mix->lr4[3], d[3], d[2], v3, n_samples);
+		lr4_process(&mix->lr4[2], d[2], d[2], v2, n_samples);
 	}
 	else {
 		for (n = 0; n < n_samples; n++) {
 			float c = s[0][n] + s[1][n];
 			d[0][n] = s[0][n] * v0;
 			d[1][n] = s[1][n] * v1;
-			d[2][n] = c * v2;
-			d[3][n] = c * v3;
+			d[2][n] = c;
 			d[4][n] = s[0][n] * v4;
 			d[5][n] = s[1][n] * v5;
 		}
-		if (v3 > 0.0f)
-			lr4_process(&mix->lr4[3], d[3], n_samples);
+		lr4_process(&mix->lr4[3], d[3], d[2], v3, 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];
 			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[2][n] = c * v2;
-			d[3][n] = c * v3;
+			d[2][n] = c;
 		}
-		if (v3 > 0.0f)
-			lr4_process(&mix->lr4[3], d[3], n_samples);
+		lr4_process(&mix->lr4[3], d[3], d[2], v3, n_samples);
+		lr4_process(&mix->lr4[2], d[2], d[2], v2, n_samples);
 	}
 	else {
 		for (n = 0; n < n_samples; n++) {
 			float c = s[0][n] + s[1][n];
 			d[0][n] = s[0][n] * v0;
 			d[1][n] = s[1][n] * v1;
-			d[2][n] = c * v2;
-			d[3][n] = c * v3;
+			d[2][n] = c;
 			d[4][n] = s[0][n] * v4;
 			d[5][n] = s[1][n] * v5;
 			d[6][n] = s[0][n] * v6;
 			d[7][n] = s[1][n] * v7;
 		}
-		if (v3 > 0.0f)
-			lr4_process(&mix->lr4[3], d[3], n_samples);
+		lr4_process(&mix->lr4[3], d[3], d[2], v3, n_samples);
+		lr4_process(&mix->lr4[2], d[2], d[2], v2, n_samples);
 	}
 }
 
diff --git a/spa/plugins/audioconvert/channelmix-ops.c b/spa/plugins/audioconvert/channelmix-ops.c
index defcc0fce..199d14361 100644
--- a/spa/plugins/audioconvert/channelmix-ops.c
+++ b/spa/plugins/audioconvert/channelmix-ops.c
@@ -457,9 +457,8 @@ done:
 		if (i == _CH(LFE) && mix->lfe_cutoff > 0.0f) {
 			spa_log_debug(mix->log, "channel %d is LFE", ic);
 			lr4_set(&mix->lr4[ic], BQ_LOWPASS, mix->lfe_cutoff / mix->freq);
-			mix->lr4_info[ic] = 1;
 		} else {
-			mix->lr4_info[ic] = 0;
+			mix->lr4[ic].active = false;
 		}
 		ic++;
 	}
diff --git a/spa/plugins/audioconvert/channelmix-ops.h b/spa/plugins/audioconvert/channelmix-ops.h
index 48a26d4fb..29a7fec81 100644
--- a/spa/plugins/audioconvert/channelmix-ops.h
+++ b/spa/plugins/audioconvert/channelmix-ops.h
@@ -69,7 +69,6 @@ struct channelmix {
 
 	float freq;					/* sample frequency */
 	float lfe_cutoff;				/* in Hz, 0 is disabled */
-	uint32_t lr4_info[SPA_AUDIO_MAX_CHANNELS];
 	struct lr4 lr4[SPA_AUDIO_MAX_CHANNELS];
 
 	void (*process) (struct channelmix *mix, void * SPA_RESTRICT dst[],
diff --git a/spa/plugins/audioconvert/crossover.c b/spa/plugins/audioconvert/crossover.c
index 046c16a02..1ce1d221d 100644
--- a/spa/plugins/audioconvert/crossover.c
+++ b/spa/plugins/audioconvert/crossover.c
@@ -5,6 +5,9 @@
 
 #include "config.h"
 
+#include <float.h>
+#include <string.h>
+
 #include "crossover.h"
 
 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->z1 = 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 lx2 = lr4->x2;
@@ -31,11 +35,22 @@ void lr4_process(struct lr4 *lr4, float *data, int samples)
 	float lb2 = lr4->bq.b2;
 	float la1 = lr4->bq.a1;
 	float la2 = lr4->bq.a2;
-
 	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++) {
 		float x, y, z;
-		x = data[i];
+		x = src[i];
 		y = lb0*x + lb1*lx1 + lb2*lx2 - la1*ly1 - la2*ly2;
 		z = lb0*y + lb1*ly1 + lb2*ly2 - la1*lz1 - la2*lz2;
 		lx2 = lx1;
@@ -44,13 +59,14 @@ void lr4_process(struct lr4 *lr4, float *data, int samples)
 		ly1 = y;
 		lz2 = lz1;
 		lz1 = z;
-		data[i] = z;
+		dst[i] = z * vol;
 	}
-
-	lr4->x1 = lx1;
-	lr4->x2 = lx2;
-	lr4->y1 = ly1;
-	lr4->y2 = ly2;
-	lr4->z1 = lz1;
-	lr4->z2 = lz2;
+#define F(x) (-FLT_MIN < (x) && (x) < FLT_MIN ? 0.0f : (x))
+	lr4->x1 = F(lx1);
+	lr4->x2 = F(lx2);
+	lr4->y1 = F(ly1);
+	lr4->y2 = F(ly2);
+	lr4->z1 = F(lz1);
+	lr4->z2 = F(lz2);
+#undef F
 }
diff --git a/spa/plugins/audioconvert/crossover.h b/spa/plugins/audioconvert/crossover.h
index f18695ddd..b6f458baf 100644
--- a/spa/plugins/audioconvert/crossover.h
+++ b/spa/plugins/audioconvert/crossover.h
@@ -6,6 +6,8 @@
 #ifndef CROSSOVER_H_
 #define CROSSOVER_H_
 
+#include <stdbool.h>
+
 #include "biquad.h"
 /* An LR4 filter is two biquads with the same parameters connected in series:
  *
@@ -19,10 +21,11 @@ struct lr4 {
 	float x1, x2;
 	float y1, y2;
 	float z1, z2;
+	bool active;
 };
 
 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_ */