ffmpeg | branch: master | Rémi Denis-Courmont <remi@remlab.net> | Wed May 29 17:51:06 2024 +0300| [b1149520305839eaceeb20864f501c96d358731e] | committer: Rémi Denis-Courmont lavfi: get rid of bespoke double scalar products

http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=b1149520305839eace...

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)