---
libavfilter/aap_template.c | 14 +-------------
libavfilter/anlms_template.c | 16 ++--------------
libavfilter/arls_template.c | 14 +-------------
3 files changed, 4 insertions(+), 40 deletions(-)
diff --git a/libavfilter/aap_template.c b/libavfilter/aap_template.c
index ea9c815a89..0e0580fb32 100644
--- a/libavfilter/aap_template.c
+++ b/libavfilter/aap_template.c
@@ -36,18 +36,6 @@
#define fn2(a,b) fn3(a,b)
#define fn(a) fn2(a, SAMPLE_FORMAT)
-#if DEPTH == 64
-static double scalarproduct_double(const double *v1, const double *v2, int len)
-{
- double p = 0.0;
-
- for (int i = 0; i < len; i++)
- p += v1[i] * v2[i];
-
- return p;
-}
-#endif
-
static ftype fn(fir_sample)(AudioAPContext *s, ftype sample, ftype *delay,
ftype *coeffs, ftype *tmp, int *offset)
{
@@ -60,7 +48,7 @@ static ftype fn(fir_sample)(AudioAPContext *s, ftype sample,
ftype *delay,
#if DEPTH == 32
output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size);
#else
- output = scalarproduct_double(delay, tmp, s->kernel_size);
+ output = s->fdsp->scalarproduct_double(delay, tmp, s->kernel_size);
#endif
if (--(*offset) < 0)
diff --git a/libavfilter/anlms_template.c b/libavfilter/anlms_template.c
index b25df4fa18..a8d1dbfe0f 100644
--- a/libavfilter/anlms_template.c
+++ b/libavfilter/anlms_template.c
@@ -33,18 +33,6 @@
#define fn2(a,b) fn3(a,b)
#define fn(a) fn2(a, SAMPLE_FORMAT)
-#if DEPTH == 64
-static double scalarproduct_double(const double *v1, const double *v2, int len)
-{
- double p = 0.0;
-
- for (int i = 0; i < len; i++)
- p += v1[i] * v2[i];
-
- return p;
-}
-#endif
-
static ftype fn(fir_sample)(AudioNLMSContext *s, ftype sample, ftype *delay,
ftype *coeffs, ftype *tmp, int *offset)
{
@@ -58,7 +46,7 @@ static ftype fn(fir_sample)(AudioNLMSContext *s, ftype
sample, ftype *delay,
#if DEPTH == 32
output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size);
#else
- output = scalarproduct_double(delay, tmp, s->kernel_size);
+ output = s->fdsp->scalarproduct_double(delay, tmp, s->kernel_size);
#endif
if (--(*offset) < 0)
@@ -85,7 +73,7 @@ static ftype fn(process_sample)(AudioNLMSContext *s, ftype
input, ftype desired,
#if DEPTH == 32
sum = s->fdsp->scalarproduct_float(delay, delay, s->kernel_size);
#else
- sum = scalarproduct_double(delay, delay, s->kernel_size);
+ sum = s->fdsp->scalarproduct_double(delay, delay, s->kernel_size);
#endif
norm = s->eps + sum;
b = mu * e / norm;
diff --git a/libavfilter/arls_template.c b/libavfilter/arls_template.c
index d8b19d89a5..c67b48cf6f 100644
--- a/libavfilter/arls_template.c
+++ b/libavfilter/arls_template.c
@@ -39,18 +39,6 @@
#define fn2(a,b) fn3(a,b)
#define fn(a) fn2(a, SAMPLE_FORMAT)
-#if DEPTH == 64
-static double scalarproduct_double(const double *v1, const double *v2, int len)
-{
- double p = 0.0;
-
- for (int i = 0; i < len; i++)
- p += v1[i] * v2[i];
-
- return p;
-}
-#endif
-
static ftype fn(fir_sample)(AudioRLSContext *s, ftype sample, ftype *delay,
ftype *coeffs, ftype *tmp, int *offset)
{
@@ -64,7 +52,7 @@ static ftype fn(fir_sample)(AudioRLSContext *s, ftype sample,
ftype *delay,
#if DEPTH == 32
output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size);
#else
- output = scalarproduct_double(delay, tmp, s->kernel_size);
+ output = s->fdsp->scalarproduct_double(delay, tmp, s->kernel_size);
#endif
if (--(*offset) < 0)
--
2.45.1
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