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tests: Add test for client disconnection

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This commit is contained in:
Olivier Crête 2013-02-18 20:22:18 -05:00 committed by Wim Taymans
parent 2e11184171
commit dcc92cbde1
1 changed files with 66 additions and 0 deletions
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66
tests/check/gst/rtspserver.c
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@ -1017,6 +1017,71 @@ GST_END_TEST;
GST_START_TEST (test_play_disconnect)
{
GstRTSPConnection *conn;
GstSDPMessage *sdp_message = NULL;
const GstSDPMedia *sdp_media;
const gchar *video_control;
const gchar *audio_control;
GstRTSPRange client_port;
gchar *session = NULL;
GstRTSPTransport *video_transport = NULL;
GstRTSPTransport *audio_transport = NULL;
GstRTSPSessionPool *pool;
pool = gst_rtsp_server_get_session_pool (server);
g_signal_connect (server, "client-connected",
G_CALLBACK (session_connected_new_session_cb), new_session_timeout_one);
start_server ();
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
/* get control strings from DESCRIBE response */
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
sdp_media = gst_sdp_message_get_media (sdp_message, 1);
audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
get_client_ports (&client_port);
/* do SETUP for video and audio */
fail_unless (do_setup (conn, video_control, &client_port, &session,
&video_transport) == GST_RTSP_STS_OK);
fail_unless (do_setup (conn, audio_control, &client_port, &session,
&audio_transport) == GST_RTSP_STS_OK);
fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
/* send PLAY request and check that we get 200 OK */
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
session) == GST_RTSP_STS_OK);
gst_rtsp_connection_free (conn);
sleep (7);
fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
/* clean up and iterate so the clean-up can finish */
g_object_unref (pool);
g_free (session);
gst_rtsp_transport_free (video_transport);
gst_rtsp_transport_free (audio_transport);
gst_sdp_message_free (sdp_message);
stop_server ();
iterate ();
}
GST_END_TEST;
static Suite *
rtspserver_suite (void)
{
@ -1038,6 +1103,7 @@ rtspserver_suite (void)
tcase_add_test (tc, test_play_multithreaded_block_in_describe);
tcase_add_test (tc, test_play_multithreaded_timeout_client);
tcase_add_test (tc, test_play_multithreaded_timeout_session);
tcase_add_test (tc, test_play_disconnect);
return s;
}