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

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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.
This commit is contained in:
Sebastian Dröge 2007-08-17 15:05:17 +00:00
parent f86bfaf5f9
commit 1301d15e4f
5 changed files with 116 additions and 184 deletions
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141
gst/audiofx/audiowsincband.c
View file

@ -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;
DEFINE_PROCESS_FUNC (32, float);
DEFINE_PROCESS_FUNC (64, double);
/* 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;
}
#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);

2
gst/audiofx/audiowsincband.h
View file

@ -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 */

145
gst/audiofx/audiowsinclimit.c
View file

@ -227,15 +227,17 @@ gst_lpwsinc_class_init (GstLPWSincClass * klass)
gobject_class->get_property = lpwsinc_get_property;
gobject_class->dispose = gst_lpwsinc_dispose;
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_READWRITE | GST_PARAM_CONTROLLABLE));
/* 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_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;
DEFINE_PROCESS_FUNC (32, float);
DEFINE_PROCESS_FUNC (64, double);
/* 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;
}
#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);

2
gst/audiofx/audiowsinclimit.h
View file

@ -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 */

10
tests/check/elements/audiowsinclimit.c
View file

@ -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++)