From 6871d561db6dc655c467d81838e90cc48648e77e Mon Sep 17 00:00:00 2001 From: Sebastian Dröge Date: Sat, 11 Aug 2007 15:58:30 +0000 Subject: gst/filter/: Fix processing with buffer sizes that are larger than the filter kernel size. Original commit message from CVS: * gst/filter/gstbpwsinc.c: (process_32), (process_64): * gst/filter/gstlpwsinc.c: (process_32), (process_64): Fix processing with buffer sizes that are larger than the filter kernel size. --- gst/audiofx/audiowsincband.c | 28 ++++++++++++++++++++++++---- gst/audiofx/audiowsinclimit.c | 28 ++++++++++++++++++++++++---- 2 files changed, 48 insertions(+), 8 deletions(-) (limited to 'gst/audiofx') diff --git a/gst/audiofx/audiowsincband.c b/gst/audiofx/audiowsincband.c index d6bb3d12..9d550ba3 100644 --- a/gst/audiofx/audiowsincband.c +++ b/gst/audiofx/audiowsincband.c @@ -245,6 +245,7 @@ process_32 (GstBPWSinc * self, gfloat * src, gfloat * dst, guint input_samples) gint kernel_length = self->kernel_length; gint i, j, k, l; gint channels = GST_AUDIO_FILTER (self)->format.channels; + gint res_start; /* convolution */ for (i = 0; i < input_samples; i++) { @@ -260,8 +261,17 @@ process_32 (GstBPWSinc * self, gfloat * src, gfloat * dst, guint input_samples) dst[i] += src[(l - j) * channels + k] * self->kernel[j]; } - /* copy the tail of the current input buffer to the residue */ - for (i = 0; i < kernel_length * channels; i++) + /* copy the tail of the current input buffer to the residue, while + * keeping parts of the residue if the input buffer is smaller than + * the kernel length */ + if (input_samples < kernel_length * channels) + res_start = kernel_length * channels - input_samples; + else + res_start = 0; + + for (i = 0; i < res_start; i++) + self->residue[i] = self->residue[i + input_samples]; + for (i = res_start; i < kernel_length * channels; i++) self->residue[i] = src[input_samples - kernel_length * channels + i]; } @@ -272,6 +282,7 @@ process_64 (GstBPWSinc * self, gdouble * src, gdouble * dst, gint kernel_length = self->kernel_length; gint i, j, k, l; gint channels = GST_AUDIO_FILTER (self)->format.channels; + gint res_start; /* convolution */ for (i = 0; i < input_samples; i++) { @@ -287,8 +298,17 @@ process_64 (GstBPWSinc * self, gdouble * src, gdouble * dst, dst[i] += src[(l - j) * channels + k] * self->kernel[j]; } - /* copy the tail of the current input buffer to the residue */ - for (i = 0; i < kernel_length * channels; i++) + /* copy the tail of the current input buffer to the residue, while + * keeping parts of the residue if the input buffer is smaller than + * the kernel length */ + if (input_samples < kernel_length * channels) + res_start = kernel_length * channels - input_samples; + else + res_start = 0; + + for (i = 0; i < res_start; i++) + self->residue[i] = self->residue[i + input_samples]; + for (i = res_start; i < kernel_length * channels; i++) self->residue[i] = src[input_samples - kernel_length * channels + i]; } diff --git a/gst/audiofx/audiowsinclimit.c b/gst/audiofx/audiowsinclimit.c index f6393169..54ebc2a3 100644 --- a/gst/audiofx/audiowsinclimit.c +++ b/gst/audiofx/audiowsinclimit.c @@ -240,6 +240,7 @@ process_32 (GstLPWSinc * self, gfloat * src, gfloat * dst, guint input_samples) gint kernel_length = self->kernel_length; gint i, j, k, l; gint channels = GST_AUDIO_FILTER (self)->format.channels; + gint res_start; /* convolution */ for (i = 0; i < input_samples; i++) { @@ -255,8 +256,17 @@ process_32 (GstLPWSinc * self, gfloat * src, gfloat * dst, guint input_samples) dst[i] += src[(l - j) * channels + k] * self->kernel[j]; } - /* copy the tail of the current input buffer to the residue */ - for (i = 0; i < kernel_length * channels; i++) + /* copy the tail of the current input buffer to the residue, while + * keeping parts of the residue if the input buffer is smaller than + * the kernel length */ + if (input_samples < kernel_length * channels) + res_start = kernel_length * channels - input_samples; + else + res_start = 0; + + for (i = 0; i < res_start; i++) + self->residue[i] = self->residue[i + input_samples]; + for (i = res_start; i < kernel_length * channels; i++) self->residue[i] = src[input_samples - kernel_length * channels + i]; } @@ -267,6 +277,7 @@ process_64 (GstLPWSinc * self, gdouble * src, gdouble * dst, gint kernel_length = self->kernel_length; gint i, j, k, l; gint channels = GST_AUDIO_FILTER (self)->format.channels; + gint res_start; /* convolution */ for (i = 0; i < input_samples; i++) { @@ -282,8 +293,17 @@ process_64 (GstLPWSinc * self, gdouble * src, gdouble * dst, dst[i] += src[(l - j) * channels + k] * self->kernel[j]; } - /* copy the tail of the current input buffer to the residue */ - for (i = 0; i < kernel_length * channels; i++) + /* copy the tail of the current input buffer to the residue, while + * keeping parts of the residue if the input buffer is smaller than + * the kernel length */ + if (input_samples < kernel_length * channels) + res_start = kernel_length * channels - input_samples; + else + res_start = 0; + + for (i = 0; i < res_start; i++) + self->residue[i] = self->residue[i + input_samples]; + for (i = res_start; i < kernel_length * channels; i++) self->residue[i] = src[input_samples - kernel_length * channels + i]; } -- cgit