+2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * check/Makefile.am:
+ * check/elements/audiotestsrc.c: (setup_audiotestsrc),
+ (cleanup_audiotestsrc), (GST_START_TEST), (audiotestsrc_suite),
+ (main):
+ add a test for audiotestsrc, testing all waves. Even seems
+ leak-free at first glance, nice job Stefan
+
2005-11-23 Jan Schmidt <thaytan@mad.scientist.com>
* po/af.po:
check_PROGRAMS = \
elements/audioconvert \
elements/audioresample \
+ elements/audiotestsrc \
elements/volume \
generic/states \
pipelines/simple_launch_lines \
clocks/selection \
$(check_vorbis)
-# tests to fix leaks in
VALGRIND_TO_FIX = \
elements/audioresample \
generic/states \
--- /dev/null
+/* 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#include <unistd.h>
+
+#include <gst/check/gstcheck.h>
+
+GList *buffers = NULL;
+gboolean have_eos = FALSE;
+
+/* 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 *mysinkpad;
+
+
+#define CAPS_TEMPLATE_STRING \
+ "audio/x-raw-int, " \
+ "channels = (int) 1, " \
+ "rate = (int) [ 1, MAX ], " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 16, " \
+ "depth = (int) 16, " \
+ "signed = (bool) TRUE"
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (CAPS_TEMPLATE_STRING)
+ );
+
+GstElement *
+setup_audiotestsrc ()
+{
+ GstElement *audiotestsrc;
+
+ GST_DEBUG ("setup_audiotestsrc");
+ audiotestsrc = gst_check_setup_element ("audiotestsrc");
+ mysinkpad = gst_check_setup_sink_pad (audiotestsrc, &sinktemplate, NULL);
+ gst_pad_set_active (mysinkpad, TRUE);
+
+ return audiotestsrc;
+}
+
+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_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);
+
+ fail_unless (gst_element_set_state (audiotestsrc,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ while (g_list_length (buffers) < 10);
+
+ 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;
+
+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;
+}
+
+int
+main (int argc, char **argv)
+{
+ int nf;
+
+ Suite *s = audiotestsrc_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;
+}
check_PROGRAMS = \
elements/audioconvert \
elements/audioresample \
+ elements/audiotestsrc \
elements/volume \
generic/states \
pipelines/simple_launch_lines \
clocks/selection \
$(check_vorbis)
-# tests to fix leaks in
VALGRIND_TO_FIX = \
elements/audioresample \
generic/states \
--- /dev/null
+/* 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#include <unistd.h>
+
+#include <gst/check/gstcheck.h>
+
+GList *buffers = NULL;
+gboolean have_eos = FALSE;
+
+/* 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 *mysinkpad;
+
+
+#define CAPS_TEMPLATE_STRING \
+ "audio/x-raw-int, " \
+ "channels = (int) 1, " \
+ "rate = (int) [ 1, MAX ], " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 16, " \
+ "depth = (int) 16, " \
+ "signed = (bool) TRUE"
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (CAPS_TEMPLATE_STRING)
+ );
+
+GstElement *
+setup_audiotestsrc ()
+{
+ GstElement *audiotestsrc;
+
+ GST_DEBUG ("setup_audiotestsrc");
+ audiotestsrc = gst_check_setup_element ("audiotestsrc");
+ mysinkpad = gst_check_setup_sink_pad (audiotestsrc, &sinktemplate, NULL);
+ gst_pad_set_active (mysinkpad, TRUE);
+
+ return audiotestsrc;
+}
+
+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_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);
+
+ fail_unless (gst_element_set_state (audiotestsrc,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ while (g_list_length (buffers) < 10);
+
+ 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;
+
+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;
+}
+
+int
+main (int argc, char **argv)
+{
+ int nf;
+
+ Suite *s = audiotestsrc_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;
+}
projects / platform / upstream / gstreamer.git / commitdiff