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/* GStreamer
*
* unit test for audiotestsrc
*
* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#include <unistd.h>
#include <gst/check/gstcheck.h>
#include <gst/audio/audio.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 */
static GstPad *mysinkpad;
#define CAPS_TEMPLATE_STRING \
"audio/x-raw, " \
"format = (string) "GST_AUDIO_NE(S16)", " \
"channels = (int) 1, " \
"rate = (int) [ 1, MAX ]"
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (CAPS_TEMPLATE_STRING)
);
static GstElement *
setup_audiotestsrc (void)
{
GstElement *audiotestsrc;
GST_DEBUG ("setup_audiotestsrc");
audiotestsrc = gst_check_setup_element ("audiotestsrc");
mysinkpad = gst_check_setup_sink_pad (audiotestsrc, &sinktemplate);
gst_pad_set_active (mysinkpad, TRUE);
return audiotestsrc;
}
static void
cleanup_audiotestsrc (GstElement * audiotestsrc)
{
GST_DEBUG ("cleanup_audiotestsrc");
g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
g_list_free (buffers);
buffers = NULL;
gst_pad_set_active (mysinkpad, FALSE);
gst_check_teardown_sink_pad (audiotestsrc);
gst_check_teardown_element (audiotestsrc);
}
GST_START_TEST (test_all_waves)
{
GstElement *audiotestsrc;
GObjectClass *oclass;
GParamSpec *property;
GEnumValue *values;
guint j = 0;
audiotestsrc = setup_audiotestsrc ();
oclass = G_OBJECT_GET_CLASS (audiotestsrc);
property = g_object_class_find_property (oclass, "wave");
fail_unless (G_IS_PARAM_SPEC_ENUM (property));
values = G_ENUM_CLASS (g_type_class_ref (property->value_type))->values;
while (values[j].value_name) {
GST_DEBUG_OBJECT (audiotestsrc, "testing wave %s", values[j].value_name);
g_object_set (audiotestsrc, "wave", values[j].value, NULL);
fail_unless (gst_element_set_state (audiotestsrc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_mutex_lock (&check_mutex);
while (g_list_length (buffers) < 10)
g_cond_wait (&check_cond, &check_mutex);
g_mutex_unlock (&check_mutex);
gst_element_set_state (audiotestsrc, GST_STATE_READY);
g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
g_list_free (buffers);
buffers = NULL;
++j;
}
/* cleanup */
cleanup_audiotestsrc (audiotestsrc);
}
GST_END_TEST;
static Suite *
audiotestsrc_suite (void)
{
Suite *s = suite_create ("audiotestsrc");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_all_waves);
return s;
}
GST_CHECK_MAIN (audiotestsrc);