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/* GStreamer
*
* unit test for audiotestsrc basetime handling
*
* Copyright (C) 2009 Maemo Multimedia <multimedia at maemo 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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/check/gstcheck.h>
#ifndef GST_DISABLE_PARSE
static GstClockTime old_ts = GST_CLOCK_TIME_NONE;
static gboolean
break_mainloop (gpointer data)
{
GMainLoop *loop;
loop = (GMainLoop *) data;
g_main_loop_quit (loop);
return FALSE;
}
static GstPadProbeReturn
buffer_probe_cb (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
{
GstBuffer *buffer = GST_PAD_PROBE_INFO_BUFFER (info);
GstClockTime new_ts = GST_BUFFER_TIMESTAMP (buffer);
GST_LOG ("ts = %" GST_TIME_FORMAT, GST_TIME_ARGS (new_ts));
if (old_ts != GST_CLOCK_TIME_NONE) {
fail_unless (new_ts != old_ts,
"Two buffers had same timestamp: %" GST_TIME_FORMAT,
GST_TIME_ARGS (old_ts));
}
old_ts = new_ts;
return TRUE;
}
GST_START_TEST (test_basetime_calculation)
{
GstElement *p1, *bin;
GstElement *asrc, *asink;
GstPad *pad;
GMainLoop *loop;
loop = g_main_loop_new (NULL, FALSE);
/* The "main" pipeline */
p1 = gst_parse_launch ("fakesrc ! fakesink", NULL);
fail_if (p1 == NULL);
/* Create a sub-bin that is activated only in "certain situations" */
asrc = gst_element_factory_make ("audiotestsrc", NULL);
if (asrc == NULL) {
GST_WARNING ("Cannot run test. 'audiotestsrc' not available");
gst_element_set_state (p1, GST_STATE_NULL);
gst_object_unref (p1);
return;
}
asink = gst_element_factory_make ("fakesink", NULL);
bin = gst_bin_new ("audiobin");
gst_bin_add_many (GST_BIN (bin), asrc, asink, NULL);
gst_element_link (asrc, asink);
gst_bin_add (GST_BIN (p1), bin);
gst_element_set_state (p1, GST_STATE_READY);
pad = gst_element_get_static_pad (asink, "sink");
fail_unless (pad != NULL, "Could not get pad out of sink");
gst_pad_add_probe (pad, GST_PAD_PROBE_TYPE_BUFFER, buffer_probe_cb, NULL,
NULL);
gst_element_set_locked_state (bin, TRUE);
/* Run main pipeline first */
gst_element_set_state (p1, GST_STATE_PLAYING);
g_timeout_add (2 * 1000, break_mainloop, loop);
g_main_loop_run (loop);
/* Now activate the audio pipeline */
gst_element_set_locked_state (bin, FALSE);
gst_element_set_state (p1, GST_STATE_PAUSED);
/* Normally our custom audiobin would send this message */
gst_element_post_message (asrc,
gst_message_new_clock_provide (GST_OBJECT (asrc), NULL, TRUE));
/* At this point a new clock is selected */
gst_element_set_state (p1, GST_STATE_PLAYING);
g_timeout_add (2 * 1000, break_mainloop, loop);
g_main_loop_run (loop);
gst_object_unref (pad);
gst_element_set_state (p1, GST_STATE_NULL);
gst_object_unref (p1);
g_main_loop_unref (loop);
}
GST_END_TEST;
#endif /* #ifndef GST_DISABLE_PARSE */
static Suite *
baseaudiosrc_suite (void)
{
Suite *s = suite_create ("baseaudiosrc");
TCase *tc_chain = tcase_create ("general");
/* timeout 6 sec */
tcase_set_timeout (tc_chain, 6);
suite_add_tcase (s, tc_chain);
#ifndef GST_DISABLE_PARSE
tcase_add_test (tc_chain, test_basetime_calculation);
#endif
return s;
}
int
main (int argc, char **argv)
{
int nf;
Suite *s = baseaudiosrc_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;
}
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