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.

tests/check/Makefile.am

@@ -119,7 +119,8 @@ check_PROGRAMS = \
 	libs/tag \
 	libs/video \
 	pipelines/simple-launch-lines \
-	pipelines/streamheader
+	pipelines/streamheader \
+	pipelines/basetime
 
 # TORTURE_TO_FIX = \
 #       elements/adder

tests/check/pipelines/basetime.c

@@ -0,0 +1,145 @@
+/* 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, 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 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);
+}
+
+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;
+}