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gstreamer/tests/check/elements/audiowsinclimit.c
Sebastian Dröge 0016658ace gst/audiofx/: Implement a base class for generic audio FIR filters. 
Original commit message from CVS:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofxbasefirfilter.c:
(gst_audio_fx_base_fir_filter_dispose),
(gst_audio_fx_base_fir_filter_base_init),
(gst_audio_fx_base_fir_filter_class_init),
(gst_audio_fx_base_fir_filter_init),
(gst_audio_fx_base_fir_filter_push_residue),
(gst_audio_fx_base_fir_filter_setup),
(gst_audio_fx_base_fir_filter_transform),
(gst_audio_fx_base_fir_filter_start),
(gst_audio_fx_base_fir_filter_stop),
(gst_audio_fx_base_fir_filter_query),
(gst_audio_fx_base_fir_filter_query_type),
(gst_audio_fx_base_fir_filter_event),
(gst_audio_fx_base_fir_filter_set_kernel):
* gst/audiofx/audiofxbasefirfilter.h:
* gst/audiofx/audiofxbaseiirfilter.c:
Implement a base class for generic audio FIR filters.
* gst/audiofx/audiowsincband.c:
(gst_gst_audio_wsincband_mode_get_type),
(gst_gst_audio_wsincband_window_get_type),
(gst_audio_wsincband_base_init), (gst_audio_wsincband_class_init),
(gst_audio_wsincband_init), (gst_audio_wsincband_build_kernel),
(gst_audio_wsincband_setup), (gst_audio_wsincband_set_property),
(gst_audio_wsincband_get_property):
* gst/audiofx/audiowsincband.h:
* gst/audiofx/audiowsinclimit.c:
(gst_audio_wsinclimit_mode_get_type),
(gst_audio_wsinclimit_window_get_type),
(gst_audio_wsinclimit_base_init),
(gst_audio_wsinclimit_class_init), (gst_audio_wsinclimit_init),
(gst_audio_wsinclimit_build_kernel), (gst_audio_wsinclimit_setup),
(gst_audio_wsinclimit_set_property),
(gst_audio_wsinclimit_get_property):
* gst/audiofx/audiowsinclimit.h:
* tests/check/elements/audiowsincband.c: (GST_START_TEST):
* tests/check/elements/audiowsinclimit.c: (GST_START_TEST):
Use this new base class for audiowsincband and audiowsinclimit.
Also cleanup both elements.
2009-01-11 19:03:38 +00:00

704 lines
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/* GStreamer
*
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* audiowsinclimit.c: Unit test for the audiowsinclimit element
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 of
* the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/check/gstcheck.h>
#include <math.h>
/* For ease of programming we use globals to keep refs for our floating
* src and sink pads we create; otherwise we always have to do get_pad,
* get_peer, and then remove references in every test function */
GstPad *mysrcpad, *mysinkpad;
#define AUDIO_WSINC_LIMIT_CAPS_STRING_32 \
"audio/x-raw-float, " \
"channels = (int) 1, " \
"rate = (int) 44100, " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 32" \
#define AUDIO_WSINC_LIMIT_CAPS_STRING_64 \
"audio/x-raw-float, " \
"channels = (int) 1, " \
"rate = (int) 44100, " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 64" \
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"channels = (int) 1, "
"rate = (int) 44100, "
"endianness = (int) BYTE_ORDER, " "width = (int) { 32, 64 } ")
);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"channels = (int) 1, "
"rate = (int) 44100, "
"endianness = (int) BYTE_ORDER, " "width = (int) { 32, 64 } ")
);
GstElement *
setup_audiowsinclimit ()
{
GstElement *audiowsinclimit;
GST_DEBUG ("setup_audiowsinclimit");
audiowsinclimit = gst_check_setup_element ("audiowsinclimit");
mysrcpad = gst_check_setup_src_pad (audiowsinclimit, &srctemplate, NULL);
mysinkpad = gst_check_setup_sink_pad (audiowsinclimit, &sinktemplate, NULL);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
return audiowsinclimit;
}
void
cleanup_audiowsinclimit (GstElement * audiowsinclimit)
{
GST_DEBUG ("cleanup_audiowsinclimit");
g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
g_list_free (buffers);
buffers = NULL;
gst_pad_set_active (mysrcpad, FALSE);
gst_pad_set_active (mysinkpad, FALSE);
gst_check_teardown_src_pad (audiowsinclimit);
gst_check_teardown_sink_pad (audiowsinclimit);
gst_check_teardown_element (audiowsinclimit);
}
/* Test if data containing only one frequency component
* at 0 is preserved with lowpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_32_lp_0hz)
{
GstElement *audiowsinclimit;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gfloat *in, *res, rms;
gint i;
GList *node;
audiowsinclimit = setup_audiowsinclimit ();
/* Set to lowpass */
g_object_set (G_OBJECT (audiowsinclimit), "mode", 0, NULL);
g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL);
fail_unless (gst_element_set_state (audiowsinclimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
/* cutoff = sampling rate / 4, data = 0 */
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++)
in[i] = 1.0;
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gfloat *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gfloat);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms >= 0.9);
}
/* cleanup */
cleanup_audiowsinclimit (audiowsinclimit);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is erased with lowpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_32_lp_22050hz)
{
GstElement *audiowsinclimit;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gfloat *in, *res, rms;
gint i;
GList *node;
audiowsinclimit = setup_audiowsinclimit ();
/* Set to lowpass */
g_object_set (G_OBJECT (audiowsinclimit), "mode", 0, NULL);
g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL);
fail_unless (gst_element_set_state (audiowsinclimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gfloat *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gfloat);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms <= 0.1);
}
/* cleanup */
cleanup_audiowsinclimit (audiowsinclimit);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at 0 is erased with highpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_32_hp_0hz)
{
GstElement *audiowsinclimit;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gfloat *in, *res, rms;
gint i;
GList *node;
audiowsinclimit = setup_audiowsinclimit ();
/* Set to highpass */
g_object_set (G_OBJECT (audiowsinclimit), "mode", 1, NULL);
g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL);
fail_unless (gst_element_set_state (audiowsinclimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++)
in[i] = 1.0;
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gfloat *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gfloat);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms <= 0.1);
}
/* cleanup */
cleanup_audiowsinclimit (audiowsinclimit);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is preserved with highpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_32_hp_22050hz)
{
GstElement *audiowsinclimit;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gfloat *in, *res, rms;
gint i;
GList *node;
audiowsinclimit = setup_audiowsinclimit ();
/* Set to highpass */
g_object_set (G_OBJECT (audiowsinclimit), "mode", 1, NULL);
g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL);
fail_unless (gst_element_set_state (audiowsinclimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gfloat *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gfloat);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms >= 0.9);
}
/* cleanup */
cleanup_audiowsinclimit (audiowsinclimit);
}
GST_END_TEST;
/* Test if buffers smaller than the kernel size are handled
* correctly without accessing wrong memory areas */
GST_START_TEST (test_32_small_buffer)
{
GstElement *audiowsinclimit;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gfloat *in;
gint i;
audiowsinclimit = setup_audiowsinclimit ();
/* Set to lowpass */
g_object_set (G_OBJECT (audiowsinclimit), "mode", 0, NULL);
g_object_set (G_OBJECT (audiowsinclimit), "length", 101, NULL);
fail_unless (gst_element_set_state (audiowsinclimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gfloat));
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 20; i++)
in[i] = 1.0;
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
/* cleanup */
cleanup_audiowsinclimit (audiowsinclimit);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at 0 is preserved with lowpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_64_lp_0hz)
{
GstElement *audiowsinclimit;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
GList *node;
audiowsinclimit = setup_audiowsinclimit ();
/* Set to lowpass */
g_object_set (G_OBJECT (audiowsinclimit), "mode", 0, NULL);
g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL);
fail_unless (gst_element_set_state (audiowsinclimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
/* cutoff = sampling rate / 4, data = 0 */
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++)
in[i] = 1.0;
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_64);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gdouble);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms >= 0.9);
}
/* cleanup */
cleanup_audiowsinclimit (audiowsinclimit);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is erased with lowpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_64_lp_22050hz)
{
GstElement *audiowsinclimit;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
GList *node;
audiowsinclimit = setup_audiowsinclimit ();
/* Set to lowpass */
g_object_set (G_OBJECT (audiowsinclimit), "mode", 0, NULL);
g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL);
fail_unless (gst_element_set_state (audiowsinclimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_64);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gdouble);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms <= 0.1);
}
/* cleanup */
cleanup_audiowsinclimit (audiowsinclimit);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at 0 is erased with highpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_64_hp_0hz)
{
GstElement *audiowsinclimit;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
GList *node;
audiowsinclimit = setup_audiowsinclimit ();
/* Set to highpass */
g_object_set (G_OBJECT (audiowsinclimit), "mode", 1, NULL);
g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL);
fail_unless (gst_element_set_state (audiowsinclimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++)
in[i] = 1.0;
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_64);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gdouble);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms <= 0.1);
}
/* cleanup */
cleanup_audiowsinclimit (audiowsinclimit);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is preserved with highpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_64_hp_22050hz)
{
GstElement *audiowsinclimit;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
GList *node;
audiowsinclimit = setup_audiowsinclimit ();
/* Set to highpass */
g_object_set (G_OBJECT (audiowsinclimit), "mode", 1, NULL);
g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL);
fail_unless (gst_element_set_state (audiowsinclimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_64);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gdouble);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms >= 0.9);
}
/* cleanup */
cleanup_audiowsinclimit (audiowsinclimit);
}
GST_END_TEST;
/* Test if buffers smaller than the kernel size are handled
* correctly without accessing wrong memory areas */
GST_START_TEST (test_64_small_buffer)
{
GstElement *audiowsinclimit;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in;
gint i;
audiowsinclimit = setup_audiowsinclimit ();
/* Set to lowpass */
g_object_set (G_OBJECT (audiowsinclimit), "mode", 0, NULL);
g_object_set (G_OBJECT (audiowsinclimit), "length", 101, NULL);
fail_unless (gst_element_set_state (audiowsinclimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gdouble));
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 20; i++)
in[i] = 1.0;
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_64);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
/* cleanup */
cleanup_audiowsinclimit (audiowsinclimit);
}
GST_END_TEST;
Suite *
audiowsinclimit_suite (void)
{
Suite *s = suite_create ("audiowsinclimit");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_32_lp_0hz);
tcase_add_test (tc_chain, test_32_lp_22050hz);
tcase_add_test (tc_chain, test_32_hp_0hz);
tcase_add_test (tc_chain, test_32_hp_22050hz);
tcase_add_test (tc_chain, test_32_small_buffer);
tcase_add_test (tc_chain, test_64_lp_0hz);
tcase_add_test (tc_chain, test_64_lp_22050hz);
tcase_add_test (tc_chain, test_64_hp_0hz);
tcase_add_test (tc_chain, test_64_hp_22050hz);
tcase_add_test (tc_chain, test_64_small_buffer);
return s;
}
int
main (int argc, char **argv)
{
int nf;
Suite *s = audiowsinclimit_suite ();
SRunner *sr = srunner_create (s);
gst_check_init (&argc, &argv);
srunner_run_all (sr, CK_NORMAL);
nf = srunner_ntests_failed (sr);
srunner_free (sr);
return nf;
}
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