Use generator macros for the process functions for the different sample types, add lower upper boundaries for the GOb...

Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(bpwsinc_set_property), (bpwsinc_get_property):
* gst/filter/gstbpwsinc.h:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (lpwsinc_build_kernel), (lpwsinc_set_property),
(lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
* tests/check/elements/lpwsinc.c: (GST_START_TEST):
Use generator macros for the process functions for the different
sample types, add lower upper boundaries for the GObject properties
so automatically generated UIs can use sliders and change frequency
properties to floats to save a bit of memory, even ints would in
theory be enough. Also rename frequency to cutoff for consistency
reasons.
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
Regenerated for the above changes.

4 files changed, 110 insertions, 178 deletions

diff --git a/gst/audiofx/audiowsincband.c b/gst/audiofx/audiowsincband.c
index 2304ac68..79fa39d9 100644
--- a/gst/audiofx/audiowsincband.c
+++ b/gst/audiofx/audiowsincband.c
@@ -230,18 +230,18 @@ gst_bpwsinc_class_init (GstBPWSincClass * klass)
   gobject_class->get_property = bpwsinc_get_property;
   gobject_class->dispose = gst_bpwsinc_dispose;
 
+  /* FIXME: Don't use the complete possible range but restrict the upper boundary
+   * so automatically generated UIs can use a slider */
   g_object_class_install_property (gobject_class, PROP_LOWER_FREQUENCY,
-      g_param_spec_double ("lower-frequency", "Lower Frequency",
-          "Cut-off lower frequency (Hz)",
-          0.0, G_MAXDOUBLE, 0, G_PARAM_READWRITE));
+      g_param_spec_float ("lower-frequency", "Lower Frequency",
+          "Cut-off lower frequency (Hz)", 0.0, 100000.0, 0, G_PARAM_READWRITE));
   g_object_class_install_property (gobject_class, PROP_UPPER_FREQUENCY,
-      g_param_spec_double ("upper-frequency", "Upper Frequency",
-          "Cut-off upper frequency (Hz)",
-          0.0, G_MAXDOUBLE, 0, G_PARAM_READWRITE));
+      g_param_spec_float ("upper-frequency", "Upper Frequency",
+          "Cut-off upper frequency (Hz)", 0.0, 100000.0, 0, G_PARAM_READWRITE));
   g_object_class_install_property (gobject_class, PROP_LENGTH,
       g_param_spec_int ("length", "Length",
           "Filter kernel length, will be rounded to the next odd number",
-          3, G_MAXINT, 101, G_PARAM_READWRITE));
+          3, 50000, 101, G_PARAM_READWRITE));
 
   g_object_class_install_property (gobject_class, PROP_MODE,
       g_param_spec_enum ("mode", "Mode",
@@ -282,86 +282,51 @@ gst_bpwsinc_init (GstBPWSinc * self, GstBPWSincClass * g_class)
       bpwsinc_query_type);
 }
 
-static void
-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++) {
-    dst[i] = 0.0;
-    k = i % channels;
-    l = i / channels;
-    for (j = 0; j < kernel_length; j++)
-      if (l < j)
-        dst[i] +=
-            self->residue[(kernel_length + l - j) * channels +
-            k] * self->kernel[j];
-      else
-        dst[i] += src[(l - j) * channels + k] * self->kernel[j];
-  }
-
-  /* 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];
-
-  self->residue_length += kernel_length * channels - res_start;
-  if (self->residue_length > kernel_length * channels)
-    self->residue_length = kernel_length * channels;
+#define DEFINE_PROCESS_FUNC(width,ctype) \
+static void \
+process_##width (GstBPWSinc * self, g##ctype * src, g##ctype * 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++) { \
+    dst[i] = 0.0; \
+    k = i % channels; \
+    l = i / channels; \
+    for (j = 0; j < kernel_length; j++) \
+      if (l < j) \
+        dst[i] += \
+            self->residue[(kernel_length + l - j) * channels + \
+            k] * self->kernel[j]; \
+      else \
+        dst[i] += src[(l - j) * channels + k] * self->kernel[j]; \
+  } \
+  \
+  /* 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]; \
+  \
+  self->residue_length += kernel_length * channels - res_start; \
+  if (self->residue_length > kernel_length * channels) \
+    self->residue_length = kernel_length * channels; \
 }
 
-static void
-process_64 (GstBPWSinc * self, gdouble * src, gdouble * 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++) {
-    dst[i] = 0.0;
-    k = i % channels;
-    l = i / channels;
-    for (j = 0; j < kernel_length; j++)
-      if (l < j)
-        dst[i] +=
-            self->residue[(kernel_length + l - j) * channels +
-            k] * self->kernel[j];
-      else
-        dst[i] += src[(l - j) * channels + k] * self->kernel[j];
-  }
+DEFINE_PROCESS_FUNC (32, float);
+DEFINE_PROCESS_FUNC (64, double);
 
-  /* 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];
-
-  self->residue_length += kernel_length * channels - res_start;
-  if (self->residue_length > kernel_length * channels)
-    self->residue_length = kernel_length * channels;
-}
+#undef DEFINE_PROCESS_FUNC
 
 static void
 bpwsinc_build_kernel (GstBPWSinc * self)
@@ -860,13 +825,13 @@ bpwsinc_set_property (GObject * object, guint prop_id, const GValue * value,
     }
     case PROP_LOWER_FREQUENCY:
       GST_BASE_TRANSFORM_LOCK (self);
-      self->lower_frequency = g_value_get_double (value);
+      self->lower_frequency = g_value_get_float (value);
       bpwsinc_build_kernel (self);
       GST_BASE_TRANSFORM_UNLOCK (self);
       break;
     case PROP_UPPER_FREQUENCY:
       GST_BASE_TRANSFORM_LOCK (self);
-      self->upper_frequency = g_value_get_double (value);
+      self->upper_frequency = g_value_get_float (value);
       bpwsinc_build_kernel (self);
       GST_BASE_TRANSFORM_UNLOCK (self);
       break;
@@ -899,10 +864,10 @@ bpwsinc_get_property (GObject * object, guint prop_id, GValue * value,
       g_value_set_int (value, self->kernel_length);
       break;
     case PROP_LOWER_FREQUENCY:
-      g_value_set_double (value, self->lower_frequency);
+      g_value_set_float (value, self->lower_frequency);
       break;
     case PROP_UPPER_FREQUENCY:
-      g_value_set_double (value, self->upper_frequency);
+      g_value_set_float (value, self->upper_frequency);
       break;
     case PROP_MODE:
       g_value_set_enum (value, self->mode);
diff --git a/gst/audiofx/audiowsincband.h b/gst/audiofx/audiowsincband.h
index d790d040..24a4d2da 100644
--- a/gst/audiofx/audiowsincband.h
+++ b/gst/audiofx/audiowsincband.h
@@ -65,7 +65,7 @@ struct _GstBPWSinc {
 
   gint mode;
   gint window;
-  gdouble lower_frequency, upper_frequency;
+  gfloat lower_frequency, upper_frequency;
   gint kernel_length;           /* length of the filter kernel */
 
   gdouble *residue;             /* buffer for left-over samples from previous buffer */
diff --git a/gst/audiofx/audiowsinclimit.c b/gst/audiofx/audiowsinclimit.c
index 3869553b..3cf14e5d 100644
--- a/gst/audiofx/audiowsinclimit.c
+++ b/gst/audiofx/audiowsinclimit.c
@@ -227,15 +227,17 @@ gst_lpwsinc_class_init (GstLPWSincClass * klass)
   gobject_class->get_property = lpwsinc_get_property;
   gobject_class->dispose = gst_lpwsinc_dispose;
 
+
+  /* FIXME: Don't use the complete possible range but restrict the upper boundary
+   * so automatically generated UIs can use a slider */
   g_object_class_install_property (gobject_class, PROP_FREQUENCY,
-      g_param_spec_double ("frequency", "Frequency",
-          "Cut-off Frequency (Hz)", 0.0, G_MAXDOUBLE, 0.0,
+      g_param_spec_float ("cutoff", "Cutoff",
+          "Cut-off Frequency (Hz)", 0.0, 100000.0, 0.0,
           G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
-
   g_object_class_install_property (gobject_class, PROP_LENGTH,
       g_param_spec_int ("length", "Length",
           "Filter kernel length, will be rounded to the next odd number",
-          3, G_MAXINT, 101, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+          3, 50000, 101, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
 
   g_object_class_install_property (gobject_class, PROP_MODE,
       g_param_spec_enum ("mode", "Mode",
@@ -261,7 +263,7 @@ gst_lpwsinc_init (GstLPWSinc * self, GstLPWSincClass * g_class)
   self->window = WINDOW_HAMMING;
   self->kernel_length = 101;
   self->latency = 50;
-  self->frequency = 0.0;
+  self->cutoff = 0.0;
   self->kernel = NULL;
   self->residue = NULL;
 
@@ -275,86 +277,51 @@ gst_lpwsinc_init (GstLPWSinc * self, GstLPWSincClass * g_class)
       lpwsinc_query_type);
 }
 
-static void
-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++) {
-    dst[i] = 0.0;
-    k = i % channels;
-    l = i / channels;
-    for (j = 0; j < kernel_length; j++)
-      if (l < j)
-        dst[i] +=
-            self->residue[(kernel_length + l - j) * channels +
-            k] * self->kernel[j];
-      else
-        dst[i] += src[(l - j) * channels + k] * self->kernel[j];
-  }
-
-  /* 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];
-
-  self->residue_length += kernel_length * channels - res_start;
-  if (self->residue_length > kernel_length * channels)
-    self->residue_length = kernel_length * channels;
+#define DEFINE_PROCESS_FUNC(width,ctype) \
+static void \
+process_##width (GstLPWSinc * self, g##ctype * src, g##ctype * 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++) { \
+    dst[i] = 0.0; \
+    k = i % channels; \
+    l = i / channels; \
+    for (j = 0; j < kernel_length; j++) \
+      if (l < j) \
+        dst[i] += \
+            self->residue[(kernel_length + l - j) * channels + \
+            k] * self->kernel[j]; \
+      else \
+        dst[i] += src[(l - j) * channels + k] * self->kernel[j]; \
+  } \
+  \
+  /* 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]; \
+  \
+  self->residue_length += kernel_length * channels - res_start; \
+  if (self->residue_length > kernel_length * channels) \
+    self->residue_length = kernel_length * channels; \
 }
 
-static void
-process_64 (GstLPWSinc * self, gdouble * src, gdouble * 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++) {
-    dst[i] = 0.0;
-    k = i % channels;
-    l = i / channels;
-    for (j = 0; j < kernel_length; j++)
-      if (l < j)
-        dst[i] +=
-            self->residue[(kernel_length + l - j) * channels +
-            k] * self->kernel[j];
-      else
-        dst[i] += src[(l - j) * channels + k] * self->kernel[j];
-  }
+DEFINE_PROCESS_FUNC (32, float);
+DEFINE_PROCESS_FUNC (64, double);
 
-  /* 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];
-
-  self->residue_length += kernel_length * channels - res_start;
-  if (self->residue_length > kernel_length * channels)
-    self->residue_length = kernel_length * channels;
-}
+#undef DEFINE_PROCESS_FUNC
 
 static void
 lpwsinc_build_kernel (GstLPWSinc * self)
@@ -377,17 +344,17 @@ lpwsinc_build_kernel (GstLPWSinc * self)
   }
 
   /* Clamp cutoff frequency between 0 and the nyquist frequency */
-  self->frequency =
-      CLAMP (self->frequency, 0.0, GST_AUDIO_FILTER (self)->format.rate / 2);
+  self->cutoff =
+      CLAMP (self->cutoff, 0.0, GST_AUDIO_FILTER (self)->format.rate / 2);
 
   GST_DEBUG ("lpwsinc: initializing filter kernel of length %d "
       "with cutoff %.2lf Hz "
       "for mode %s",
-      len, self->frequency,
+      len, self->cutoff,
       (self->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass");
 
   /* fill the kernel */
-  w = 2 * M_PI * (self->frequency / GST_AUDIO_FILTER (self)->format.rate);
+  w = 2 * M_PI * (self->cutoff / GST_AUDIO_FILTER (self)->format.rate);
 
   if (self->kernel)
     g_free (self->kernel);
@@ -800,7 +767,7 @@ lpwsinc_set_property (GObject * object, guint prop_id, const GValue * value,
     }
     case PROP_FREQUENCY:
       GST_BASE_TRANSFORM_LOCK (self);
-      self->frequency = g_value_get_double (value);
+      self->cutoff = g_value_get_float (value);
       lpwsinc_build_kernel (self);
       GST_BASE_TRANSFORM_UNLOCK (self);
       break;
@@ -833,7 +800,7 @@ lpwsinc_get_property (GObject * object, guint prop_id, GValue * value,
       g_value_set_int (value, self->kernel_length);
       break;
     case PROP_FREQUENCY:
-      g_value_set_double (value, self->frequency);
+      g_value_set_float (value, self->cutoff);
       break;
     case PROP_MODE:
       g_value_set_enum (value, self->mode);
diff --git a/gst/audiofx/audiowsinclimit.h b/gst/audiofx/audiowsinclimit.h
index 3e09bd52..f56f5a4d 100644
--- a/gst/audiofx/audiowsinclimit.h
+++ b/gst/audiofx/audiowsinclimit.h
@@ -65,7 +65,7 @@ struct _GstLPWSinc {
 
   gint mode;
   gint window;
-  gdouble frequency;
+  gfloat cutoff;
   gint kernel_length;           /* length of the filter kernel */
 
   gdouble *residue;             /* buffer for left-over samples from previous buffer */