Use generator macros for the process functions for the different sample types, add...

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.

diff --git a/gst/audiofx/audiowsincband.c b/gst/audiofx/audiowsincband.c
index 2304ac6886786feb8afcb7d1e710288e0ccffd74..79fa39d96033ae8c7cbe5c5a6cc4a9af692ed843 100644 (file)
--- 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 d790d0402b7c76d86b44769c98b46dda6eb59820..24a4d2da8fb7309ee2d2df5d29ed18c3ab0b14f3 100644 (file)
--- 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 3869553b8954f5bc103b6c2e928220ed5924e001..3cf14e5d2a5df56ef06dcc3e8dcd9387121bd31e 100644 (file)
--- 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 3e09bd521be379eb0a4dd923517c66546cff3465..f56f5a4df2bbd36bccfefa6df6571cd6824eb37a 100644 (file)
--- 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 */

diff --git a/tests/check/elements/audiowsinclimit.c b/tests/check/elements/audiowsinclimit.c
index 3d69bc95e96f8caebe008b8016f0d10c28671a95..d2aeda4772147a33411e8e219d5c075ab99f9605 100644 (file)
--- a/tests/check/elements/audiowsinclimit.c
+++ b/tests/check/elements/audiowsinclimit.c
@@ -108,7 +108,7 @@ GST_START_TEST (test_lp_0hz)
       "could not set to playing");
 
   /* cutoff = sampling rate / 4, data = 0 */
-  g_object_set (G_OBJECT (lpwsinc), "frequency", 44100 / 4.0, NULL);
+  g_object_set (G_OBJECT (lpwsinc), "cutoff", 44100 / 4.0, NULL);
   inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
   in = (gdouble *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 128; i++)
@@ -166,7 +166,7 @@ GST_START_TEST (test_lp_22050hz)
           GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
       "could not set to playing");
 
-  g_object_set (G_OBJECT (lpwsinc), "frequency", 44100 / 4.0, NULL);
+  g_object_set (G_OBJECT (lpwsinc), "cutoff", 44100 / 4.0, NULL);
   inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
   in = (gdouble *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 128; i += 2) {
@@ -226,7 +226,7 @@ GST_START_TEST (test_hp_0hz)
           GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
       "could not set to playing");
 
-  g_object_set (G_OBJECT (lpwsinc), "frequency", 44100 / 4.0, NULL);
+  g_object_set (G_OBJECT (lpwsinc), "cutoff", 44100 / 4.0, NULL);
   inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
   in = (gdouble *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 128; i++)
@@ -284,7 +284,7 @@ GST_START_TEST (test_hp_22050hz)
           GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
       "could not set to playing");
 
-  g_object_set (G_OBJECT (lpwsinc), "frequency", 44100 / 4.0, NULL);
+  g_object_set (G_OBJECT (lpwsinc), "cutoff", 44100 / 4.0, NULL);
   inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
   in = (gdouble *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 128; i += 2) {
@@ -344,7 +344,7 @@ GST_START_TEST (test_small_buffer)
           GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
       "could not set to playing");
 
-  g_object_set (G_OBJECT (lpwsinc), "frequency", 44100 / 4.0, NULL);
+  g_object_set (G_OBJECT (lpwsinc), "cutoff", 44100 / 4.0, NULL);
   inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gdouble));
   in = (gdouble *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 20; i++)