Commit 5e8e7c03 authored by Tommi Myöhänen's avatar Tommi Myöhänen Committed by Stefan Kost
Browse files

tests: new test for baseaudiosrc base_time comparison

This test reveals a bug in comparison operation between timestamp and
GstElement's base_time in GstBaseAudioSrc.
parent a4bc39ef
......@@ -119,7 +119,8 @@ check_PROGRAMS = \
libs/tag \
libs/video \
pipelines/simple-launch-lines \
pipelines/streamheader \
# elements/adder
/* 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
* 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 "config.h"
#include <gst/check/gstcheck.h>
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 gboolean
buffer_probe_cb (GstPad * pad, GstBuffer * buffer)
if (old_ts != GST_CLOCK_TIME_NONE) {
fail_unless (GST_BUFFER_TIMESTAMP (buffer) != old_ts,
"Two buffers had same timestamp");
old_ts = GST_BUFFER_TIMESTAMP (buffer);
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" */
bin = gst_bin_new ("audiobin");
asrc = gst_element_factory_make (DEFAULT_AUDIOSRC, NULL);
asink = gst_element_factory_make ("fakesink", NULL);
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_buffer_probe (pad, G_CALLBACK (buffer_probe_cb), 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);
#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);
tcase_add_test (tc_chain, test_basetime_calculation);
return s;
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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