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127 lines
3.6 KiB
C
127 lines
3.6 KiB
C
/* GStreamer
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*
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* unit test for audiotestsrc
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*
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* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <unistd.h>
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#include <gst/check/gstcheck.h>
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#include <gst/audio/audio.h>
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/* For ease of programming we use globals to keep refs for our floating
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* src and sink pads we create; otherwise we always have to do get_pad,
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* get_peer, and then remove references in every test function */
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static GstPad *mysinkpad;
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#define CAPS_TEMPLATE_STRING \
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"audio/x-raw, " \
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"format = (string) "GST_AUDIO_NE(S16)", " \
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"channels = (int) 1, " \
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"rate = (int) [ 1, MAX ]"
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static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (CAPS_TEMPLATE_STRING)
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);
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static GstElement *
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setup_audiotestsrc (void)
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{
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GstElement *audiotestsrc;
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GST_DEBUG ("setup_audiotestsrc");
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audiotestsrc = gst_check_setup_element ("audiotestsrc");
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mysinkpad = gst_check_setup_sink_pad (audiotestsrc, &sinktemplate);
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gst_pad_set_active (mysinkpad, TRUE);
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return audiotestsrc;
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}
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static void
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cleanup_audiotestsrc (GstElement * audiotestsrc)
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{
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GST_DEBUG ("cleanup_audiotestsrc");
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g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
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g_list_free (buffers);
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buffers = NULL;
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gst_pad_set_active (mysinkpad, FALSE);
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gst_check_teardown_sink_pad (audiotestsrc);
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gst_check_teardown_element (audiotestsrc);
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}
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GST_START_TEST (test_all_waves)
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{
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GstElement *audiotestsrc;
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GObjectClass *oclass;
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GParamSpec *property;
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GEnumValue *values;
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guint j = 0;
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audiotestsrc = setup_audiotestsrc ();
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oclass = G_OBJECT_GET_CLASS (audiotestsrc);
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property = g_object_class_find_property (oclass, "wave");
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fail_unless (G_IS_PARAM_SPEC_ENUM (property));
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values = G_ENUM_CLASS (g_type_class_ref (property->value_type))->values;
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while (values[j].value_name) {
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GST_DEBUG_OBJECT (audiotestsrc, "testing wave %s", values[j].value_name);
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g_object_set (audiotestsrc, "wave", values[j].value, NULL);
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fail_unless (gst_element_set_state (audiotestsrc,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
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"could not set to playing");
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g_mutex_lock (&check_mutex);
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while (g_list_length (buffers) < 10)
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g_cond_wait (&check_cond, &check_mutex);
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g_mutex_unlock (&check_mutex);
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gst_element_set_state (audiotestsrc, GST_STATE_READY);
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g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
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g_list_free (buffers);
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buffers = NULL;
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++j;
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}
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/* cleanup */
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cleanup_audiotestsrc (audiotestsrc);
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}
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GST_END_TEST;
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static Suite *
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audiotestsrc_suite (void)
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{
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Suite *s = suite_create ("audiotestsrc");
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TCase *tc_chain = tcase_create ("general");
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suite_add_tcase (s, tc_chain);
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tcase_add_test (tc_chain, test_all_waves);
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return s;
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}
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GST_CHECK_MAIN (audiotestsrc);
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