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| author | Jan Schmidt <thaytan@noraisin.net> | 2009-07-14 15:06:41 +0100 |
|---|---|---|
| committer | Jan Schmidt <thaytan@noraisin.net> | 2009-07-14 15:06:41 +0100 |
| commit | 7753d463501a656f29c2ef016eac1c66336c7cc1 (patch) | |
| tree | 8b4c5f47f08e817c6fc11ff5bb44b789a9824590 | |
| parent | b26eae25d0c7c311745b8ede150a9b3774c6c8f0 (diff) | |
| download | gstreamer-plugins-base-7753d463501a656f29c2ef016eac1c66336c7cc1.tar.gz | |
adder: Perform get_state() in the unit test
Wait for the alsasrc to return to NULL after setting it to PAUSED for
testing, otherwise it leads to segfaults later on.
| -rw-r--r-- | tests/check/elements/adder.c | 2 |
1 files changed, 2 insertions, 0 deletions
diff --git a/tests/check/elements/adder.c b/tests/check/elements/adder.c index 0b4bb9689..53ff3496a 100644 --- a/tests/check/elements/adder.c +++ b/tests/check/elements/adder.c @@ -443,6 +443,7 @@ GST_START_TEST (test_live_seeking) /* Test that the audio source can get to paused, else skip */ res = gst_element_set_state (src1, GST_STATE_PAUSED); (void) gst_element_set_state (src1, GST_STATE_NULL); + (void) gst_element_get_state (src1, NULL, NULL, GST_CLOCK_TIME_NONE); if (res == GST_STATE_CHANGE_FAILURE) { gst_object_unref (src1); goto cleanup; @@ -521,6 +522,7 @@ GST_START_TEST (test_live_seeking) /* cleanup */ cleanup: + GST_INFO ("cleaning up"); if (main_loop) g_main_loop_unref (main_loop); if (play_seek_event) |