Rémi Denis-Courmont:
> ---
> 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)
1. lavf is libavformat; libavfilter is lavfi.
2. Do you intend to use this outside of lavfi?
- Andreas
_______________________________________________
ffmpeg-devel mailing list
ffmpeg-devel@ffmpeg.org
https://ffmpeg.org/mailman/listinfo/ffmpeg-devel
To unsubscribe, visit link above, or email
ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".
