From dcc92cbde12ad4c591ca8c67648bba82fb22b01f Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Mon, 18 Feb 2013 20:22:18 -0500 Subject: [PATCH] tests: Add test for client disconnection --- tests/check/gst/rtspserver.c | 66 ++++++++++++++++++++++++++++++++++++ 1 file changed, 66 insertions(+) diff --git a/tests/check/gst/rtspserver.c b/tests/check/gst/rtspserver.c index 3464f2f0c0..524ca7fa71 100644 --- a/tests/check/gst/rtspserver.c +++ b/tests/check/gst/rtspserver.c @@ -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; }