/* GStreamer * * unit test for audiotestsrc basetime handling * * Copyright (C) 2009 Maemo Multimedia * * 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. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #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 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; /* Don't run with osxaudiosrc . This is because libcheck runs the actual * test in a forked process and causes havoc with osx's API. */ if (G_UNLIKELY (!g_ascii_strcasecmp (DEFAULT_AUDIOSRC, "osxaudiosrc"))) return; 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 (DEFAULT_AUDIOSRC, NULL); if (asrc == NULL) { GST_WARNING ("Cannot run test. test audio source %s not available", DEFAULT_AUDIOSRC); 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_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); } 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; }