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tests: Test client and session timeouts with multiple threads

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This commit is contained in:
Olivier Crête 2013-02-18 20:15:41 -05:00 committed by Wim Taymans
parent b9d111372e
commit 2e11184171
1 changed files with 190 additions and 1 deletions
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191
tests/check/gst/rtspserver.c
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@ -305,11 +305,20 @@ do_request (GstRTSPConnection * conn, GstRTSPMethod method,
}
if (session_out) {
gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &value, 0);
value = g_strdup (value);
/* Remove the timeout */
if (value) {
char *pos = strchr (value, ';');
if (pos)
*pos = 0;
}
if (session_in) {
/* check that we got the same session back */
fail_unless (!g_strcmp0 (value, session_in));
}
*session_out = g_strdup (value);
*session_out = value;
}
if (transport_out) {
gst_rtsp_message_get_header (response, GST_RTSP_HDR_TRANSPORT, &value, 0);
@ -830,6 +839,184 @@ GST_START_TEST (test_play_multithreaded_block_in_describe)
GST_END_TEST;
static void
new_session_timeout_one (GstRTSPClient * client,
GstRTSPSession * session, gpointer user_data)
{
gst_rtsp_session_set_timeout (session, 1);
g_signal_handlers_disconnect_by_func (client, new_session_timeout_one,
user_data);
}
static void
session_connected_new_session_cb (GstRTSPServer * server,
GstRTSPClient * client, gpointer user_data)
{
g_signal_connect (client, "new-session", user_data, NULL);
}
GST_START_TEST (test_play_multithreaded_timeout_client)
{
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;
GstRTSPMessage *request;
GstRTSPMessage *response;
gst_rtsp_server_set_max_threads (server, -1);
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);
sleep (7);
fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
/* send TEARDOWN request and check that we get 454 Session Not found */
request = create_request (conn, GST_RTSP_TEARDOWN, NULL);
gst_rtsp_message_add_header (request, GST_RTSP_HDR_SESSION, session);
fail_unless (send_request (conn, request));
gst_rtsp_message_free (request);
fail_unless (gst_rtsp_message_new (&response) == GST_RTSP_OK);
fail_unless (gst_rtsp_connection_receive (conn, response, NULL) ==
GST_RTSP_ESYS);
fail_unless (errno == ECONNRESET);
gst_rtsp_message_free (response);
/* 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);
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_END_TEST;
GST_START_TEST (test_play_multithreaded_timeout_session)
{
GstRTSPConnection *conn;
GstSDPMessage *sdp_message = NULL;
const GstSDPMedia *sdp_media;
const gchar *video_control;
const gchar *audio_control;
GstRTSPRange client_port;
gchar *session1 = NULL;
gchar *session2 = NULL;
GstRTSPTransport *video_transport = NULL;
GstRTSPTransport *audio_transport = NULL;
GstRTSPSessionPool *pool;
gst_rtsp_server_set_max_threads (server, -1);
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);
gst_rtsp_connection_set_remember_session_id (conn, FALSE);
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, &session1,
&video_transport) == GST_RTSP_STS_OK);
fail_unless (do_setup (conn, audio_control, &client_port, &session2,
&audio_transport) == GST_RTSP_STS_OK);
fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 2);
/* send PLAY request and check that we get 200 OK */
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
session1) == GST_RTSP_STS_OK);
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
session2) == GST_RTSP_STS_OK);
sleep (7);
fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
/* send TEARDOWN request and check that we get 454 Session Not found */
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
session1) == GST_RTSP_STS_SESSION_NOT_FOUND);
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
session2) == GST_RTSP_STS_OK);
/* clean up and iterate so the clean-up can finish */
g_object_unref (pool);
g_free (session1);
g_free (session2);
gst_rtsp_transport_free (video_transport);
gst_rtsp_transport_free (audio_transport);
gst_sdp_message_free (sdp_message);
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_END_TEST;
static Suite *
rtspserver_suite (void)
{
@ -849,6 +1036,8 @@ rtspserver_suite (void)
tcase_add_test (tc, test_bind_already_in_use);
tcase_add_test (tc, test_play_multithreaded);
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);
return s;
}