mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-10 03:19:40 +00:00
d524439b35
Makes source code smaller, and ensures we go through common initialization path (like the one that sets up XML unit test output ...)
788 lines
25 KiB
C
788 lines
25 KiB
C
/* 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/audio/audio.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, " \
|
|
"format = (string) " GST_AUDIO_NE (F32) ", " \
|
|
"layout = (string) interleaved, " \
|
|
"channels = (int) 1, " \
|
|
"rate = (int) 44100"
|
|
|
|
#define AUDIO_WSINC_LIMIT_CAPS_STRING_64 \
|
|
"audio/x-raw, " \
|
|
"format = (string) " GST_AUDIO_NE (F64) ", " \
|
|
"layout = (string) interleaved, " \
|
|
"channels = (int) 1, " \
|
|
"rate = (int) 44100"
|
|
|
|
#define FORMATS "{ "GST_AUDIO_NE (F32)","GST_AUDIO_NE (F64)" }"
|
|
|
|
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/x-raw, "
|
|
"format = (string) " FORMATS ", "
|
|
"layout = (string) interleaved, "
|
|
"channels = (int) 1, " "rate = (int) 44100")
|
|
);
|
|
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
|
|
GST_PAD_SRC,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/x-raw, "
|
|
"format = (string) " FORMATS ", "
|
|
"layout = (string) interleaved, "
|
|
"channels = (int) 1, " "rate = (int) 44100")
|
|
);
|
|
|
|
static GstElement *
|
|
setup_audiowsinclimit (void)
|
|
{
|
|
GstElement *audiowsinclimit;
|
|
|
|
GST_DEBUG ("setup_audiowsinclimit");
|
|
audiowsinclimit = gst_check_setup_element ("audiowsinclimit");
|
|
mysrcpad = gst_check_setup_src_pad (audiowsinclimit, &srctemplate);
|
|
mysinkpad = gst_check_setup_sink_pad (audiowsinclimit, &sinktemplate);
|
|
gst_pad_set_active (mysrcpad, TRUE);
|
|
gst_pad_set_active (mysinkpad, TRUE);
|
|
|
|
return audiowsinclimit;
|
|
}
|
|
|
|
static 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;
|
|
GstMapInfo map;
|
|
GList *node;
|
|
GstSegment segment;
|
|
|
|
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;
|
|
gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
|
|
in = (gfloat *) map.data;
|
|
for (i = 0; i < 128; i++)
|
|
in[i] = 1.0;
|
|
gst_buffer_unmap (inbuffer, &map);
|
|
|
|
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32);
|
|
gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME);
|
|
gst_caps_unref (caps);
|
|
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
|
|
|
|
/* ensure segment (format) properly setup */
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
|
|
|
|
/* 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);
|
|
|
|
gst_buffer_map (outbuffer, &map, GST_MAP_READ);
|
|
res = (gfloat *) map.data;
|
|
buffer_length = map.size / sizeof (gfloat);
|
|
rms = 0.0;
|
|
for (i = 0; i < buffer_length; i++)
|
|
rms += res[i] * res[i];
|
|
rms = sqrt (rms / buffer_length);
|
|
gst_buffer_unmap (outbuffer, &map);
|
|
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;
|
|
GstMapInfo map;
|
|
GList *node;
|
|
GstSegment segment;
|
|
|
|
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;
|
|
gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
|
|
in = (gfloat *) map.data;
|
|
for (i = 0; i < 128; i += 2) {
|
|
in[i] = 1.0;
|
|
in[i + 1] = -1.0;
|
|
}
|
|
gst_buffer_unmap (inbuffer, &map);
|
|
|
|
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32);
|
|
gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME);
|
|
gst_caps_unref (caps);
|
|
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
|
|
|
|
/* ensure segment (format) properly setup */
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
|
|
|
|
/* 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);
|
|
|
|
gst_buffer_map (outbuffer, &map, GST_MAP_READ);
|
|
res = (gfloat *) map.data;
|
|
buffer_length = map.size / sizeof (gfloat);
|
|
rms = 0.0;
|
|
for (i = 0; i < buffer_length; i++)
|
|
rms += res[i] * res[i];
|
|
rms = sqrt (rms / buffer_length);
|
|
gst_buffer_unmap (outbuffer, &map);
|
|
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;
|
|
GstMapInfo map;
|
|
GList *node;
|
|
GstSegment segment;
|
|
|
|
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;
|
|
gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
|
|
in = (gfloat *) map.data;
|
|
for (i = 0; i < 128; i++)
|
|
in[i] = 1.0;
|
|
gst_buffer_unmap (inbuffer, &map);
|
|
|
|
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32);
|
|
gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME);
|
|
gst_caps_unref (caps);
|
|
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
|
|
|
|
/* ensure segment (format) properly setup */
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
|
|
|
|
/* 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);
|
|
|
|
gst_buffer_map (outbuffer, &map, GST_MAP_READ);
|
|
res = (gfloat *) map.data;
|
|
buffer_length = map.size / sizeof (gfloat);
|
|
rms = 0.0;
|
|
for (i = 0; i < buffer_length; i++)
|
|
rms += res[i] * res[i];
|
|
rms = sqrt (rms / buffer_length);
|
|
gst_buffer_unmap (outbuffer, &map);
|
|
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;
|
|
GstMapInfo map;
|
|
GList *node;
|
|
GstSegment segment;
|
|
|
|
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;
|
|
gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
|
|
in = (gfloat *) map.data;
|
|
for (i = 0; i < 128; i += 2) {
|
|
in[i] = 1.0;
|
|
in[i + 1] = -1.0;
|
|
}
|
|
gst_buffer_unmap (inbuffer, &map);
|
|
|
|
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32);
|
|
gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME);
|
|
gst_caps_unref (caps);
|
|
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
|
|
|
|
/* ensure segment (format) properly setup */
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
|
|
|
|
/* 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);
|
|
|
|
gst_buffer_map (outbuffer, &map, GST_MAP_READ);
|
|
res = (gfloat *) map.data;
|
|
buffer_length = map.size / sizeof (gfloat);
|
|
rms = 0.0;
|
|
for (i = 0; i < buffer_length; i++)
|
|
rms += res[i] * res[i];
|
|
rms = sqrt (rms / buffer_length);
|
|
gst_buffer_unmap (outbuffer, &map);
|
|
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;
|
|
GstMapInfo map;
|
|
GstSegment segment;
|
|
|
|
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;
|
|
gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
|
|
in = (gfloat *) map.data;
|
|
for (i = 0; i < 20; i++)
|
|
in[i] = 1.0;
|
|
gst_buffer_unmap (inbuffer, &map);
|
|
|
|
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32);
|
|
gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME);
|
|
gst_caps_unref (caps);
|
|
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
|
|
|
|
/* ensure segment (format) properly setup */
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
|
|
|
|
/* 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;
|
|
GstMapInfo map;
|
|
GList *node;
|
|
GstSegment segment;
|
|
|
|
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;
|
|
gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
|
|
in = (gdouble *) map.data;
|
|
for (i = 0; i < 128; i++)
|
|
in[i] = 1.0;
|
|
gst_buffer_unmap (inbuffer, &map);
|
|
|
|
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_64);
|
|
gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME);
|
|
gst_caps_unref (caps);
|
|
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
|
|
|
|
/* ensure segment (format) properly setup */
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
|
|
|
|
/* 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);
|
|
|
|
gst_buffer_map (outbuffer, &map, GST_MAP_READ);
|
|
res = (gdouble *) map.data;
|
|
buffer_length = map.size / sizeof (gdouble);
|
|
rms = 0.0;
|
|
for (i = 0; i < buffer_length; i++)
|
|
rms += res[i] * res[i];
|
|
rms = sqrt (rms / buffer_length);
|
|
gst_buffer_unmap (outbuffer, &map);
|
|
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;
|
|
GstMapInfo map;
|
|
GList *node;
|
|
GstSegment segment;
|
|
|
|
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;
|
|
gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
|
|
in = (gdouble *) map.data;
|
|
for (i = 0; i < 128; i += 2) {
|
|
in[i] = 1.0;
|
|
in[i + 1] = -1.0;
|
|
}
|
|
gst_buffer_unmap (inbuffer, &map);
|
|
|
|
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_64);
|
|
gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME);
|
|
gst_caps_unref (caps);
|
|
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
|
|
|
|
/* ensure segment (format) properly setup */
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
|
|
|
|
/* 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);
|
|
|
|
gst_buffer_map (outbuffer, &map, GST_MAP_READ);
|
|
res = (gdouble *) map.data;
|
|
buffer_length = map.size / sizeof (gdouble);
|
|
rms = 0.0;
|
|
for (i = 0; i < buffer_length; i++)
|
|
rms += res[i] * res[i];
|
|
rms = sqrt (rms / buffer_length);
|
|
gst_buffer_unmap (outbuffer, &map);
|
|
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;
|
|
GstMapInfo map;
|
|
GList *node;
|
|
GstSegment segment;
|
|
|
|
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;
|
|
gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
|
|
in = (gdouble *) map.data;
|
|
for (i = 0; i < 128; i++)
|
|
in[i] = 1.0;
|
|
gst_buffer_unmap (inbuffer, &map);
|
|
|
|
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_64);
|
|
gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME);
|
|
gst_caps_unref (caps);
|
|
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
|
|
|
|
/* ensure segment (format) properly setup */
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
|
|
|
|
/* 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);
|
|
|
|
gst_buffer_map (outbuffer, &map, GST_MAP_READ);
|
|
res = (gdouble *) map.data;
|
|
buffer_length = map.size / sizeof (gdouble);
|
|
rms = 0.0;
|
|
for (i = 0; i < buffer_length; i++)
|
|
rms += res[i] * res[i];
|
|
rms = sqrt (rms / buffer_length);
|
|
gst_buffer_unmap (outbuffer, &map);
|
|
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;
|
|
GstMapInfo map;
|
|
GList *node;
|
|
GstSegment segment;
|
|
|
|
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;
|
|
gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
|
|
in = (gdouble *) map.data;
|
|
for (i = 0; i < 128; i += 2) {
|
|
in[i] = 1.0;
|
|
in[i + 1] = -1.0;
|
|
}
|
|
gst_buffer_unmap (inbuffer, &map);
|
|
|
|
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_64);
|
|
gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME);
|
|
gst_caps_unref (caps);
|
|
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
|
|
|
|
/* ensure segment (format) properly setup */
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
|
|
|
|
/* 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);
|
|
|
|
gst_buffer_map (outbuffer, &map, GST_MAP_READ);
|
|
res = (gdouble *) map.data;
|
|
buffer_length = map.size / sizeof (gdouble);
|
|
rms = 0.0;
|
|
for (i = 0; i < buffer_length; i++)
|
|
rms += res[i] * res[i];
|
|
rms = sqrt (rms / buffer_length);
|
|
gst_buffer_unmap (outbuffer, &map);
|
|
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;
|
|
GstMapInfo map;
|
|
GstSegment segment;
|
|
|
|
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;
|
|
gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
|
|
in = (gdouble *) map.data;
|
|
for (i = 0; i < 20; i++)
|
|
in[i] = 1.0;
|
|
gst_buffer_unmap (inbuffer, &map);
|
|
|
|
caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_64);
|
|
gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME);
|
|
gst_caps_unref (caps);
|
|
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
|
|
|
|
/* ensure segment (format) properly setup */
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
|
|
|
|
/* 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;
|
|
|
|
static 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;
|
|
}
|
|
|
|
GST_CHECK_MAIN (audiowsinclimit);
|