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4b84d7552f
On OS/X the DEFAULT_AUDIOSRC is not going to be available, because it isn't in gst-plugins-base. Just defer the test, instead of failing it.
153 lines
3.9 KiB
C
153 lines
3.9 KiB
C
/* GStreamer
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*
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* unit test for audiotestsrc basetime handling
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*
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* Copyright (C) 2009 Maemo Multimedia <multimedia at maemo 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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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/check/gstcheck.h>
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#ifndef GST_DISABLE_PARSE
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static GstClockTime old_ts = GST_CLOCK_TIME_NONE;
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static gboolean
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break_mainloop (gpointer data)
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{
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GMainLoop *loop;
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loop = (GMainLoop *) data;
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g_main_loop_quit (loop);
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return FALSE;
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}
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static gboolean
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buffer_probe_cb (GstPad * pad, GstBuffer * buffer)
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{
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if (old_ts != GST_CLOCK_TIME_NONE) {
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fail_unless (GST_BUFFER_TIMESTAMP (buffer) != old_ts,
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"Two buffers had same timestamp");
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}
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old_ts = GST_BUFFER_TIMESTAMP (buffer);
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return TRUE;
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}
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GST_START_TEST (test_basetime_calculation)
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{
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GstElement *p1, *bin;
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GstElement *asrc, *asink;
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GstPad *pad;
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GMainLoop *loop;
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loop = g_main_loop_new (NULL, FALSE);
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/* The "main" pipeline */
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p1 = gst_parse_launch ("fakesrc ! fakesink", NULL);
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fail_if (p1 == NULL);
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/* Create a sub-bin that is activated only in "certain situations" */
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asrc = gst_element_factory_make (DEFAULT_AUDIOSRC, NULL);
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if (asrc == NULL) {
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GST_WARNING ("Cannot run test. test audio source %s not available",
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DEFAULT_AUDIOSRC);
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gst_element_set_state (p1, GST_STATE_NULL);
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gst_object_unref (p1);
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return;
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}
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asink = gst_element_factory_make ("fakesink", NULL);
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bin = gst_bin_new ("audiobin");
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gst_bin_add_many (GST_BIN (bin), asrc, asink, NULL);
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gst_element_link (asrc, asink);
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gst_bin_add (GST_BIN (p1), bin);
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gst_element_set_state (p1, GST_STATE_READY);
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pad = gst_element_get_static_pad (asink, "sink");
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fail_unless (pad != NULL, "Could not get pad out of sink");
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gst_pad_add_buffer_probe (pad, G_CALLBACK (buffer_probe_cb), NULL);
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gst_element_set_locked_state (bin, TRUE);
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/* Run main pipeline first */
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gst_element_set_state (p1, GST_STATE_PLAYING);
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g_timeout_add (2 * 1000, break_mainloop, loop);
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g_main_loop_run (loop);
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/* Now activate the audio pipeline */
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gst_element_set_locked_state (bin, FALSE);
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gst_element_set_state (p1, GST_STATE_PAUSED);
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/* Normally our custom audiobin would send this message */
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gst_element_post_message (asrc,
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gst_message_new_clock_provide (GST_OBJECT (asrc), NULL, TRUE));
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/* At this point a new clock is selected */
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gst_element_set_state (p1, GST_STATE_PLAYING);
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g_timeout_add (2 * 1000, break_mainloop, loop);
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g_main_loop_run (loop);
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gst_object_unref (pad);
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gst_element_set_state (p1, GST_STATE_NULL);
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gst_object_unref (p1);
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}
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GST_END_TEST;
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#endif /* #ifndef GST_DISABLE_PARSE */
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static Suite *
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baseaudiosrc_suite (void)
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{
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Suite *s = suite_create ("baseaudiosrc");
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TCase *tc_chain = tcase_create ("general");
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/* timeout 6 sec */
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tcase_set_timeout (tc_chain, 6);
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suite_add_tcase (s, tc_chain);
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#ifndef GST_DISABLE_PARSE
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tcase_add_test (tc_chain, test_basetime_calculation);
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#endif
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return s;
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}
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int
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main (int argc, char **argv)
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{
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int nf;
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Suite *s = baseaudiosrc_suite ();
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SRunner *sr = srunner_create (s);
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gst_check_init (&argc, &argv);
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srunner_run_all (sr, CK_NORMAL);
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nf = srunner_ntests_failed (sr);
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srunner_free (sr);
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return nf;
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}
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