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rtspclientsink: add unit test for potential shutdown deadlock

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
This commit is contained in:
Tim-Philipp Müller 2021-02-15 12:07:45 +00:00
parent abacfb3792
commit 015e4dc810
1 changed files with 47 additions and 0 deletions
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47
tests/check/gst/rtspclientsink.c
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@ -238,6 +238,52 @@ GST_START_TEST (test_record)
GST_END_TEST;
/* Make sure we can shut down rtspclientsink while it's still waiting for
* the initial preroll data */
GST_START_TEST (test_record_no_data)
{
start_record_server ("( rtppcmadepay name=depay0 ! fakesink )");
/* Create an rtspclientsink and send some data */
{
gchar *uri = get_server_uri (test_port, TEST_MOUNT_POINT);
gchar *pipe_str;
GstMessage *msg;
GstElement *pipeline;
GstBus *bus;
pipe_str = g_strdup_printf ("appsrc caps=audio/x-alaw,rate=8000,channels=1"
" ! rtspclientsink name=sink location=%s", uri);
g_free (uri);
pipeline = gst_parse_launch (pipe_str, NULL);
g_free (pipe_str);
fail_unless (pipeline != NULL);
bus = gst_element_get_bus (pipeline);
fail_if (bus == NULL);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
/* wait for a bit */
msg = gst_bus_poll (bus, GST_MESSAGE_EOS | GST_MESSAGE_ERROR,
500 * GST_MSECOND);
fail_unless (msg == NULL);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
gst_object_unref (bus);
}
/* clean up and iterate so the clean-up can finish */
stop_server ();
iterate ();
}
GST_END_TEST;
static Suite *
rtspclientsink_suite (void)
{
@ -248,6 +294,7 @@ rtspclientsink_suite (void)
tcase_add_checked_fixture (tc, setup, teardown);
tcase_set_timeout (tc, 120);
tcase_add_test (tc, test_record);
tcase_add_test (tc, test_record_no_data);
return s;
}