mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-30 05:31:15 +00:00
951 lines
31 KiB
C
951 lines
31 KiB
C
/* GStreamer unit tests for the GstRTSPConnection API (RTSP support
|
|
* library)
|
|
*
|
|
* Copyright (C) 2014 Ognyan Tonchev <ognyan axis com>
|
|
*
|
|
* This library is free software; you can redistribute it and/or
|
|
* modify it under the terms of the GNU Library General Public
|
|
* License as published by the Free Software Foundation; either
|
|
* version 2 of the License, or (at your option) any later version.
|
|
*
|
|
* This library is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
* Library General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU Library General Public
|
|
* License along with this library; if not, write to the
|
|
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
|
|
* Boston, MA 02110-1301, USA.
|
|
*/
|
|
|
|
#ifdef HAVE_CONFIG_H
|
|
#include "config.h"
|
|
#endif
|
|
|
|
#include <gst/check/gstcheck.h>
|
|
|
|
#include <gst/rtsp/gstrtspconnection.h>
|
|
#include <string.h>
|
|
|
|
|
|
static const gchar *get_msg =
|
|
"GET /example/url HTTP/1.0\r\n"
|
|
"Host: 127.0.0.1\r\n" "x-sessioncookie: 805849328\r\n\r\n";
|
|
static const gchar *post_msg =
|
|
"POST /example/url HTTP/1.0\r\n"
|
|
"Host: 127.0.0.1\r\n"
|
|
"x-sessioncookie: 805849328\r\n"
|
|
"Content-Length: 0\r\n"
|
|
"Content-Type: application/x-rtsp-tunnelled\r\n\r\n";
|
|
|
|
static guint tunnel_get_count;
|
|
static guint tunnel_post_count;
|
|
static guint tunnel_lost_count;
|
|
static guint closed_count;
|
|
static guint message_sent_count;
|
|
|
|
typedef struct
|
|
{
|
|
GMainLoop *loop;
|
|
guint16 port;
|
|
GSocketConnection *conn;
|
|
GMutex mutex;
|
|
GCond cond;
|
|
gboolean started;
|
|
} ServiceData;
|
|
|
|
static gboolean
|
|
incoming_callback (GSocketService * service, GSocketConnection * connection,
|
|
GObject * source_object, gpointer user_data)
|
|
{
|
|
ServiceData *data = user_data;
|
|
|
|
GST_DEBUG ("new incoming connection");
|
|
data->conn = g_object_ref (connection);
|
|
g_main_loop_quit (data->loop);
|
|
return FALSE;
|
|
}
|
|
|
|
static gpointer
|
|
service_thread_func (gpointer user_data)
|
|
{
|
|
ServiceData *data = user_data;
|
|
GMainContext *service_context;
|
|
GSocketService *service;
|
|
|
|
service_context = g_main_context_new ();
|
|
g_main_context_push_thread_default (service_context);
|
|
|
|
data->loop = g_main_loop_new (service_context, FALSE);
|
|
|
|
/* find available port and start service */
|
|
service = g_socket_service_new ();
|
|
data->port = g_socket_listener_add_any_inet_port ((GSocketListener *) service,
|
|
NULL, NULL);
|
|
fail_unless (data->port != 0);
|
|
|
|
/* get notified upon new connection */
|
|
g_signal_connect (service, "incoming", G_CALLBACK (incoming_callback), data);
|
|
|
|
g_socket_service_start (service);
|
|
|
|
/* service is started */
|
|
g_mutex_lock (&data->mutex);
|
|
data->started = TRUE;
|
|
g_cond_signal (&data->cond);
|
|
g_mutex_unlock (&data->mutex);
|
|
|
|
/* our service will run in the main context of this main loop */
|
|
g_main_loop_run (data->loop);
|
|
|
|
g_main_context_pop_thread_default (service_context);
|
|
|
|
g_main_loop_unref (data->loop);
|
|
data->loop = NULL;
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static void
|
|
create_connection (GSocketConnection ** client_conn,
|
|
GSocketConnection ** server_conn)
|
|
{
|
|
ServiceData *data;
|
|
GThread *service_thread;
|
|
GSocketClient *client = g_socket_client_new ();
|
|
|
|
data = g_new0 (ServiceData, 1);
|
|
g_mutex_init (&data->mutex);
|
|
g_cond_init (&data->cond);
|
|
|
|
service_thread = g_thread_new ("service thread", service_thread_func, data);
|
|
fail_unless (service_thread != NULL);
|
|
|
|
/* wait for the service to start */
|
|
g_mutex_lock (&data->mutex);
|
|
while (!data->started) {
|
|
g_cond_wait (&data->cond, &data->mutex);
|
|
}
|
|
g_mutex_unlock (&data->mutex);
|
|
|
|
/* create the tcp link */
|
|
*client_conn = g_socket_client_connect_to_host (client, (gchar *) "localhost",
|
|
data->port, NULL, NULL);
|
|
fail_unless (*client_conn != NULL);
|
|
fail_unless (g_socket_connection_is_connected (*client_conn));
|
|
|
|
g_thread_join (service_thread);
|
|
*server_conn = data->conn;
|
|
data->conn = NULL;
|
|
fail_unless (g_socket_connection_is_connected (*server_conn));
|
|
|
|
g_mutex_clear (&data->mutex);
|
|
g_cond_clear (&data->cond);
|
|
g_free (data);
|
|
g_object_unref (client);
|
|
}
|
|
|
|
static GstRTSPStatusCode
|
|
tunnel_get (GstRTSPWatch * watch, gpointer user_data)
|
|
{
|
|
tunnel_get_count++;
|
|
return GST_RTSP_STS_OK;
|
|
}
|
|
|
|
static GstRTSPResult
|
|
tunnel_post (GstRTSPWatch * watch, gpointer user_data)
|
|
{
|
|
tunnel_post_count++;
|
|
return GST_RTSP_OK;
|
|
}
|
|
|
|
static GstRTSPResult
|
|
tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
|
|
{
|
|
tunnel_lost_count++;
|
|
return GST_RTSP_OK;
|
|
}
|
|
|
|
static GstRTSPResult
|
|
closed (GstRTSPWatch * watch, gpointer user_data)
|
|
{
|
|
closed_count++;
|
|
return GST_RTSP_OK;
|
|
}
|
|
|
|
static GstRTSPResult
|
|
message_sent (GstRTSPWatch * watch, guint id, gpointer user_data)
|
|
{
|
|
message_sent_count++;
|
|
return GST_RTSP_OK;
|
|
}
|
|
|
|
static GstRTSPWatchFuncs watch_funcs = {
|
|
NULL,
|
|
message_sent,
|
|
closed,
|
|
NULL,
|
|
tunnel_get,
|
|
tunnel_post,
|
|
NULL,
|
|
tunnel_lost
|
|
};
|
|
|
|
/* sets up a new tunnel, then disconnects the read connection and creates it
|
|
* again */
|
|
GST_START_TEST (test_rtspconnection_tunnel_setup)
|
|
{
|
|
GstRTSPConnection *rtsp_conn1 = NULL;
|
|
GstRTSPConnection *rtsp_conn2 = NULL;
|
|
GstRTSPWatch *watch1;
|
|
GstRTSPWatch *watch2;
|
|
GstRTSPResult res;
|
|
GSocketConnection *client_get = NULL;
|
|
GSocketConnection *server_get = NULL;
|
|
GSocketConnection *client_post = NULL;
|
|
GSocketConnection *server_post = NULL;
|
|
GSocket *server_sock;
|
|
GOutputStream *ostream_get;
|
|
GInputStream *istream_get;
|
|
GOutputStream *ostream_post;
|
|
gsize size = 0;
|
|
gchar buffer[1024];
|
|
|
|
/* create GET connection */
|
|
create_connection (&client_get, &server_get);
|
|
server_sock = g_socket_connection_get_socket (server_get);
|
|
fail_unless (server_sock != NULL);
|
|
|
|
res = gst_rtsp_connection_create_from_socket (server_sock, "127.0.0.1", 4444,
|
|
NULL, &rtsp_conn1);
|
|
fail_unless (res == GST_RTSP_OK);
|
|
fail_unless (rtsp_conn1 != NULL);
|
|
|
|
watch1 = gst_rtsp_watch_new (rtsp_conn1, &watch_funcs, NULL, NULL);
|
|
fail_unless (watch1 != NULL);
|
|
fail_unless (gst_rtsp_watch_attach (watch1, NULL) > 0);
|
|
g_source_unref ((GSource *) watch1);
|
|
|
|
ostream_get = g_io_stream_get_output_stream (G_IO_STREAM (client_get));
|
|
fail_unless (ostream_get != NULL);
|
|
|
|
istream_get = g_io_stream_get_input_stream (G_IO_STREAM (client_get));
|
|
fail_unless (istream_get != NULL);
|
|
|
|
/* initiate the tunnel by sending HTTP GET */
|
|
fail_unless (g_output_stream_write_all (ostream_get, get_msg,
|
|
strlen (get_msg), &size, NULL, NULL));
|
|
fail_unless (size == strlen (get_msg));
|
|
|
|
while (!g_main_context_iteration (NULL, TRUE));
|
|
fail_unless (tunnel_get_count == 1);
|
|
fail_unless (tunnel_post_count == 0);
|
|
fail_unless (tunnel_lost_count == 0);
|
|
fail_unless (closed_count == 0);
|
|
|
|
/* read the HTTP GET response */
|
|
size = g_input_stream_read (istream_get, buffer, 1024, NULL, NULL);
|
|
fail_unless (size > 0);
|
|
buffer[size] = 0;
|
|
fail_unless (g_strrstr (buffer, "HTTP/1.0 200 OK") != NULL);
|
|
|
|
/* create POST channel */
|
|
create_connection (&client_post, &server_post);
|
|
server_sock = g_socket_connection_get_socket (server_post);
|
|
fail_unless (server_sock != NULL);
|
|
|
|
res = gst_rtsp_connection_create_from_socket (server_sock, "127.0.0.1", 4444,
|
|
NULL, &rtsp_conn2);
|
|
fail_unless (res == GST_RTSP_OK);
|
|
fail_unless (rtsp_conn2 != NULL);
|
|
|
|
watch2 = gst_rtsp_watch_new (rtsp_conn2, &watch_funcs, NULL, NULL);
|
|
fail_unless (watch2 != NULL);
|
|
fail_unless (gst_rtsp_watch_attach (watch2, NULL) > 0);
|
|
g_source_unref ((GSource *) watch2);
|
|
|
|
ostream_post = g_io_stream_get_output_stream (G_IO_STREAM (client_post));
|
|
fail_unless (ostream_post != NULL);
|
|
|
|
/* complete the tunnel by sending HTTP POST */
|
|
fail_unless (g_output_stream_write_all (ostream_post, post_msg,
|
|
strlen (post_msg), &size, NULL, NULL));
|
|
fail_unless (size == strlen (post_msg));
|
|
|
|
while (!g_main_context_iteration (NULL, TRUE));
|
|
fail_unless (tunnel_get_count == 1);
|
|
fail_unless (tunnel_post_count == 1);
|
|
fail_unless (tunnel_lost_count == 0);
|
|
fail_unless (closed_count == 0);
|
|
|
|
/* merge the two connections together */
|
|
fail_unless (gst_rtsp_connection_do_tunnel (rtsp_conn1, rtsp_conn2) ==
|
|
GST_RTSP_OK);
|
|
gst_rtsp_watch_reset (watch1);
|
|
g_source_destroy ((GSource *) watch2);
|
|
gst_rtsp_connection_free (rtsp_conn2);
|
|
rtsp_conn2 = NULL;
|
|
|
|
/* it must be possible to reconnect the POST channel */
|
|
g_object_unref (client_post);
|
|
while (!g_main_context_iteration (NULL, TRUE));
|
|
fail_unless (tunnel_get_count == 1);
|
|
fail_unless (tunnel_post_count == 1);
|
|
fail_unless (tunnel_lost_count == 1);
|
|
fail_unless (closed_count == 0);
|
|
g_object_unref (server_post);
|
|
|
|
/* no other source should get dispatched */
|
|
fail_if (g_main_context_iteration (NULL, FALSE));
|
|
|
|
/* create new POST connection */
|
|
create_connection (&client_post, &server_post);
|
|
server_sock = g_socket_connection_get_socket (server_post);
|
|
fail_unless (server_sock != NULL);
|
|
|
|
res = gst_rtsp_connection_create_from_socket (server_sock, "127.0.0.1", 4444,
|
|
NULL, &rtsp_conn2);
|
|
fail_unless (res == GST_RTSP_OK);
|
|
fail_unless (rtsp_conn2 != NULL);
|
|
|
|
watch2 = gst_rtsp_watch_new (rtsp_conn2, &watch_funcs, NULL, NULL);
|
|
fail_unless (watch2 != NULL);
|
|
fail_unless (gst_rtsp_watch_attach (watch2, NULL) > 0);
|
|
g_source_unref ((GSource *) watch2);
|
|
|
|
ostream_post = g_io_stream_get_output_stream (G_IO_STREAM (client_post));
|
|
fail_unless (ostream_post != NULL);
|
|
|
|
/* complete the tunnel by sending HTTP POST */
|
|
fail_unless (g_output_stream_write_all (ostream_post, post_msg,
|
|
strlen (post_msg), &size, NULL, NULL));
|
|
fail_unless (size == strlen (post_msg));
|
|
|
|
while (!g_main_context_iteration (NULL, TRUE));
|
|
fail_unless (tunnel_get_count == 1);
|
|
fail_unless (tunnel_post_count == 2);
|
|
fail_unless (tunnel_lost_count == 1);
|
|
fail_unless (closed_count == 0);
|
|
|
|
/* merge the two connections together */
|
|
fail_unless (gst_rtsp_connection_do_tunnel (rtsp_conn1, rtsp_conn2) ==
|
|
GST_RTSP_OK);
|
|
gst_rtsp_watch_reset (watch1);
|
|
g_source_destroy ((GSource *) watch2);
|
|
gst_rtsp_connection_free (rtsp_conn2);
|
|
rtsp_conn2 = NULL;
|
|
|
|
/* check if rtspconnection can detect close of the get channel */
|
|
g_object_unref (client_get);
|
|
while (!g_main_context_iteration (NULL, TRUE));
|
|
fail_unless (tunnel_get_count == 1);
|
|
fail_unless (tunnel_post_count == 2);
|
|
fail_unless (tunnel_lost_count == 1);
|
|
fail_unless (closed_count == 1);
|
|
|
|
fail_unless (gst_rtsp_connection_close (rtsp_conn1) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_free (rtsp_conn1) == GST_RTSP_OK);
|
|
|
|
g_object_unref (client_post);
|
|
g_object_unref (server_post);
|
|
g_object_unref (server_get);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
/* sets up a new tunnel, starting with the read channel,
|
|
* then disconnects the read connection and creates it again
|
|
* ideally this test should be merged with test_rtspconnection_tunnel_setup but
|
|
* but it became quite messy */
|
|
GST_START_TEST (test_rtspconnection_tunnel_setup_post_first)
|
|
{
|
|
GstRTSPConnection *rtsp_conn1 = NULL;
|
|
GstRTSPConnection *rtsp_conn2 = NULL;
|
|
GstRTSPWatch *watch1;
|
|
GstRTSPWatch *watch2;
|
|
GstRTSPResult res;
|
|
GSocketConnection *client_get = NULL;
|
|
GSocketConnection *server_get = NULL;
|
|
GSocketConnection *client_post = NULL;
|
|
GSocketConnection *server_post = NULL;
|
|
GSocket *server_sock;
|
|
GOutputStream *ostream_get;
|
|
GInputStream *istream_get;
|
|
GOutputStream *ostream_post;
|
|
gsize size = 0;
|
|
gchar buffer[1024];
|
|
|
|
/* create POST channel */
|
|
create_connection (&client_post, &server_post);
|
|
server_sock = g_socket_connection_get_socket (server_post);
|
|
fail_unless (server_sock != NULL);
|
|
|
|
res = gst_rtsp_connection_create_from_socket (server_sock, "127.0.0.1", 4444,
|
|
NULL, &rtsp_conn1);
|
|
fail_unless (res == GST_RTSP_OK);
|
|
fail_unless (rtsp_conn1 != NULL);
|
|
|
|
watch1 = gst_rtsp_watch_new (rtsp_conn1, &watch_funcs, NULL, NULL);
|
|
fail_unless (watch1 != NULL);
|
|
fail_unless (gst_rtsp_watch_attach (watch1, NULL) > 0);
|
|
g_source_unref ((GSource *) watch1);
|
|
|
|
ostream_post = g_io_stream_get_output_stream (G_IO_STREAM (client_post));
|
|
fail_unless (ostream_post != NULL);
|
|
|
|
/* initiate the tunnel by sending HTTP POST */
|
|
fail_unless (g_output_stream_write_all (ostream_post, post_msg,
|
|
strlen (post_msg), &size, NULL, NULL));
|
|
fail_unless (size == strlen (post_msg));
|
|
|
|
while (!g_main_context_iteration (NULL, TRUE));
|
|
fail_unless (tunnel_get_count == 0);
|
|
fail_unless (tunnel_post_count == 1);
|
|
fail_unless (tunnel_lost_count == 0);
|
|
fail_unless (closed_count == 0);
|
|
|
|
/* create GET connection */
|
|
create_connection (&client_get, &server_get);
|
|
server_sock = g_socket_connection_get_socket (server_get);
|
|
fail_unless (server_sock != NULL);
|
|
|
|
res = gst_rtsp_connection_create_from_socket (server_sock, "127.0.0.1", 4444,
|
|
NULL, &rtsp_conn2);
|
|
fail_unless (res == GST_RTSP_OK);
|
|
fail_unless (rtsp_conn2 != NULL);
|
|
|
|
watch2 = gst_rtsp_watch_new (rtsp_conn2, &watch_funcs, NULL, NULL);
|
|
fail_unless (watch2 != NULL);
|
|
fail_unless (gst_rtsp_watch_attach (watch2, NULL) > 0);
|
|
g_source_unref ((GSource *) watch2);
|
|
|
|
ostream_get = g_io_stream_get_output_stream (G_IO_STREAM (client_get));
|
|
fail_unless (ostream_get != NULL);
|
|
|
|
istream_get = g_io_stream_get_input_stream (G_IO_STREAM (client_get));
|
|
fail_unless (istream_get != NULL);
|
|
|
|
/* complete the tunnel by sending HTTP GET */
|
|
fail_unless (g_output_stream_write_all (ostream_get, get_msg,
|
|
strlen (get_msg), &size, NULL, NULL));
|
|
fail_unless (size == strlen (get_msg));
|
|
|
|
while (!g_main_context_iteration (NULL, TRUE));
|
|
fail_unless (tunnel_get_count == 1);
|
|
fail_unless (tunnel_post_count == 1);
|
|
fail_unless (tunnel_lost_count == 0);
|
|
fail_unless (closed_count == 0);
|
|
|
|
/* read the HTTP GET response */
|
|
size = g_input_stream_read (istream_get, buffer, 1024, NULL, NULL);
|
|
fail_unless (size > 0);
|
|
buffer[size] = 0;
|
|
fail_unless (g_strrstr (buffer, "HTTP/1.0 200 OK") != NULL);
|
|
|
|
/* merge the two connections together */
|
|
fail_unless (gst_rtsp_connection_do_tunnel (rtsp_conn1, rtsp_conn2) ==
|
|
GST_RTSP_OK);
|
|
gst_rtsp_watch_reset (watch1);
|
|
g_source_destroy ((GSource *) watch2);
|
|
gst_rtsp_connection_free (rtsp_conn2);
|
|
rtsp_conn2 = NULL;
|
|
|
|
/* it must be possible to reconnect the POST channel */
|
|
g_object_unref (client_post);
|
|
while (!g_main_context_iteration (NULL, TRUE));
|
|
fail_unless (tunnel_get_count == 1);
|
|
fail_unless (tunnel_post_count == 1);
|
|
fail_unless (tunnel_lost_count == 1);
|
|
fail_unless (closed_count == 0);
|
|
g_object_unref (server_post);
|
|
|
|
/* no other source should get dispatched */
|
|
fail_if (g_main_context_iteration (NULL, FALSE));
|
|
|
|
/* create new POST connection */
|
|
create_connection (&client_post, &server_post);
|
|
server_sock = g_socket_connection_get_socket (server_post);
|
|
fail_unless (server_sock != NULL);
|
|
|
|
res = gst_rtsp_connection_create_from_socket (server_sock, "127.0.0.1", 4444,
|
|
NULL, &rtsp_conn2);
|
|
fail_unless (res == GST_RTSP_OK);
|
|
fail_unless (rtsp_conn2 != NULL);
|
|
|
|
watch2 = gst_rtsp_watch_new (rtsp_conn2, &watch_funcs, NULL, NULL);
|
|
fail_unless (watch2 != NULL);
|
|
fail_unless (gst_rtsp_watch_attach (watch2, NULL) > 0);
|
|
g_source_unref ((GSource *) watch2);
|
|
|
|
ostream_post = g_io_stream_get_output_stream (G_IO_STREAM (client_post));
|
|
fail_unless (ostream_post != NULL);
|
|
|
|
/* complete the tunnel by sending HTTP POST */
|
|
fail_unless (g_output_stream_write_all (ostream_post, post_msg,
|
|
strlen (post_msg), &size, NULL, NULL));
|
|
fail_unless (size == strlen (post_msg));
|
|
|
|
while (!g_main_context_iteration (NULL, TRUE));
|
|
fail_unless (tunnel_get_count == 1);
|
|
fail_unless (tunnel_post_count == 2);
|
|
fail_unless (tunnel_lost_count == 1);
|
|
fail_unless (closed_count == 0);
|
|
|
|
/* merge the two connections together */
|
|
fail_unless (gst_rtsp_connection_do_tunnel (rtsp_conn1, rtsp_conn2) ==
|
|
GST_RTSP_OK);
|
|
gst_rtsp_watch_reset (watch1);
|
|
g_source_destroy ((GSource *) watch2);
|
|
gst_rtsp_connection_free (rtsp_conn2);
|
|
rtsp_conn2 = NULL;
|
|
|
|
/* check if rtspconnection can detect close of the get channel */
|
|
g_object_unref (client_get);
|
|
while (!g_main_context_iteration (NULL, TRUE));
|
|
fail_unless (tunnel_get_count == 1);
|
|
fail_unless (tunnel_post_count == 2);
|
|
fail_unless (tunnel_lost_count == 1);
|
|
fail_unless (closed_count == 1);
|
|
|
|
fail_unless (gst_rtsp_connection_close (rtsp_conn1) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_free (rtsp_conn1) == GST_RTSP_OK);
|
|
|
|
g_object_unref (client_post);
|
|
g_object_unref (server_post);
|
|
g_object_unref (server_get);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_rtspconnection_send_receive)
|
|
{
|
|
GSocketConnection *input_conn = NULL;
|
|
GSocketConnection *output_conn = NULL;
|
|
GSocket *input_sock;
|
|
GSocket *output_sock;
|
|
GstRTSPConnection *rtsp_output_conn;
|
|
GstRTSPConnection *rtsp_input_conn;
|
|
GstRTSPMessage *msg;
|
|
gchar body[] = "message body";
|
|
gchar *recv_body;
|
|
guint recv_body_len;
|
|
|
|
create_connection (&input_conn, &output_conn);
|
|
input_sock = g_socket_connection_get_socket (input_conn);
|
|
fail_unless (input_sock != NULL);
|
|
output_sock = g_socket_connection_get_socket (output_conn);
|
|
fail_unless (output_sock != NULL);
|
|
|
|
fail_unless (gst_rtsp_connection_create_from_socket (input_sock, "127.0.0.1",
|
|
4444, NULL, &rtsp_input_conn) == GST_RTSP_OK);
|
|
fail_unless (rtsp_input_conn != NULL);
|
|
|
|
fail_unless (gst_rtsp_connection_create_from_socket (output_sock, "127.0.0.1",
|
|
4444, NULL, &rtsp_output_conn) == GST_RTSP_OK);
|
|
fail_unless (rtsp_output_conn != NULL);
|
|
|
|
/* send data message */
|
|
fail_unless (gst_rtsp_message_new_data (&msg, 1) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_message_set_body (msg, (guint8 *) body,
|
|
sizeof (body)) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_send (rtsp_output_conn, msg,
|
|
NULL) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_message_free (msg) == GST_RTSP_OK);
|
|
msg = NULL;
|
|
|
|
/* receive data message and make sure it is correct */
|
|
fail_unless (gst_rtsp_message_new (&msg) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_receive (rtsp_input_conn, msg, NULL) ==
|
|
GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_message_get_type (msg) == GST_RTSP_MESSAGE_DATA);
|
|
fail_unless (gst_rtsp_message_get_body (msg, (guint8 **) & recv_body,
|
|
&recv_body_len) == GST_RTSP_OK);
|
|
/* RTSPConnection adds an extra byte for the trailing '\0' */
|
|
fail_unless_equals_int (recv_body_len, sizeof (body) + 1);
|
|
fail_unless_equals_string (recv_body, body);
|
|
fail_unless (gst_rtsp_message_free (msg) == GST_RTSP_OK);
|
|
msg = NULL;
|
|
|
|
/* send request message */
|
|
fail_unless (gst_rtsp_message_new_request (&msg, GST_RTSP_OPTIONS,
|
|
"example.org") == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_message_set_body (msg, (guint8 *) body,
|
|
sizeof (body)) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_send (rtsp_output_conn, msg,
|
|
NULL) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_message_free (msg) == GST_RTSP_OK);
|
|
msg = NULL;
|
|
|
|
/* receive request message and make sure it is correct */
|
|
fail_unless (gst_rtsp_message_new (&msg) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_receive (rtsp_input_conn, msg, NULL) ==
|
|
GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_message_get_type (msg) == GST_RTSP_MESSAGE_REQUEST);
|
|
fail_unless (gst_rtsp_message_get_body (msg, (guint8 **) & recv_body,
|
|
&recv_body_len) == GST_RTSP_OK);
|
|
/* RTSPConnection adds an extra byte for the trailing '\0' */
|
|
fail_unless_equals_int (recv_body_len, sizeof (body) + 1);
|
|
fail_unless_equals_string (recv_body, body);
|
|
fail_unless (gst_rtsp_message_free (msg) == GST_RTSP_OK);
|
|
msg = NULL;
|
|
|
|
fail_unless (gst_rtsp_connection_close (rtsp_input_conn) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_free (rtsp_input_conn) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_close (rtsp_output_conn) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_free (rtsp_output_conn) == GST_RTSP_OK);
|
|
|
|
g_object_unref (input_conn);
|
|
g_object_unref (output_conn);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_rtspconnection_send_receive_check_headers)
|
|
{
|
|
GSocketConnection *input_conn = NULL;
|
|
GSocketConnection *output_conn = NULL;
|
|
GSocket *input_sock;
|
|
GSocket *output_sock;
|
|
GstRTSPConnection *rtsp_output_conn;
|
|
GstRTSPConnection *rtsp_input_conn;
|
|
GstRTSPMessage *msg;
|
|
gchar *header_val;
|
|
|
|
create_connection (&input_conn, &output_conn);
|
|
input_sock = g_socket_connection_get_socket (input_conn);
|
|
fail_unless (input_sock != NULL);
|
|
output_sock = g_socket_connection_get_socket (output_conn);
|
|
fail_unless (output_sock != NULL);
|
|
|
|
fail_unless (gst_rtsp_connection_create_from_socket (input_sock, "127.0.0.1",
|
|
4444, NULL, &rtsp_input_conn) == GST_RTSP_OK);
|
|
fail_unless (rtsp_input_conn != NULL);
|
|
|
|
fail_unless (gst_rtsp_connection_create_from_socket (output_sock, "127.0.0.1",
|
|
4444, NULL, &rtsp_output_conn) == GST_RTSP_OK);
|
|
fail_unless (rtsp_output_conn != NULL);
|
|
|
|
/* send request message */
|
|
fail_unless (gst_rtsp_message_new_request (&msg, GST_RTSP_SETUP,
|
|
"rtsp://example.com/") == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_message_add_header (msg, GST_RTSP_HDR_BLOCKSIZE,
|
|
"1024") == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_message_add_header_by_name (msg, "Custom-Header",
|
|
"lol") == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_send (rtsp_output_conn, msg,
|
|
NULL) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_message_free (msg) == GST_RTSP_OK);
|
|
msg = NULL;
|
|
|
|
/* receive request message and make sure it is correct */
|
|
fail_unless (gst_rtsp_message_new (&msg) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_receive (rtsp_input_conn, msg, NULL) ==
|
|
GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_message_get_type (msg) == GST_RTSP_MESSAGE_REQUEST);
|
|
/* check headers */
|
|
fail_unless (gst_rtsp_message_get_header (msg, GST_RTSP_HDR_BLOCKSIZE,
|
|
&header_val, 0) == GST_RTSP_OK);
|
|
fail_unless (!g_strcmp0 (header_val, "1024"));
|
|
fail_unless (gst_rtsp_message_get_header_by_name (msg, "Custom-Header",
|
|
&header_val, 0) == GST_RTSP_OK);
|
|
fail_unless (!g_strcmp0 (header_val, "lol"));
|
|
fail_unless (gst_rtsp_message_free (msg) == GST_RTSP_OK);
|
|
msg = NULL;
|
|
|
|
fail_unless (gst_rtsp_connection_close (rtsp_input_conn) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_free (rtsp_input_conn) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_close (rtsp_output_conn) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_free (rtsp_output_conn) == GST_RTSP_OK);
|
|
|
|
g_object_unref (input_conn);
|
|
g_object_unref (output_conn);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_rtspconnection_connect)
|
|
{
|
|
ServiceData *data;
|
|
GThread *service_thread;
|
|
GSocketConnection *socket_conn;
|
|
GstRTSPConnection *rtsp_conn = NULL;
|
|
GstRTSPUrl *url = NULL;
|
|
gchar *path;
|
|
|
|
data = g_new0 (ServiceData, 1);
|
|
g_mutex_init (&data->mutex);
|
|
g_cond_init (&data->cond);
|
|
|
|
/* create socket service */
|
|
service_thread = g_thread_new ("service thread", service_thread_func, data);
|
|
fail_unless (service_thread != NULL);
|
|
|
|
/* wait for the service to start */
|
|
g_mutex_lock (&data->mutex);
|
|
while (!data->started) {
|
|
g_cond_wait (&data->cond, &data->mutex);
|
|
}
|
|
g_mutex_unlock (&data->mutex);
|
|
|
|
/* connect to our service using the RTSPConnection API */
|
|
path = g_strdup_printf ("rtsp://localhost:%d", data->port);
|
|
fail_unless (gst_rtsp_url_parse (path, &url) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_create (url, &rtsp_conn) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_connect (rtsp_conn, NULL) == GST_RTSP_OK);
|
|
g_free (path);
|
|
gst_rtsp_url_free (url);
|
|
|
|
/* wait for the other end and check whether it is connected */
|
|
g_thread_join (service_thread);
|
|
socket_conn = data->conn;
|
|
data->conn = NULL;
|
|
fail_unless (g_socket_connection_is_connected (socket_conn));
|
|
|
|
fail_unless (gst_rtsp_connection_close (rtsp_conn) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_free (rtsp_conn) == GST_RTSP_OK);
|
|
g_object_unref (socket_conn);
|
|
g_mutex_clear (&data->mutex);
|
|
g_cond_clear (&data->cond);
|
|
g_free (data);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_rtspconnection_poll)
|
|
{
|
|
GSocketConnection *conn1 = NULL;
|
|
GSocketConnection *conn2 = NULL;
|
|
GSocket *sock;
|
|
GstRTSPConnection *rtsp_conn;
|
|
GstRTSPEvent event;
|
|
GOutputStream *ostream;
|
|
gsize size;
|
|
gint64 timeout;
|
|
|
|
create_connection (&conn1, &conn2);
|
|
sock = g_socket_connection_get_socket (conn1);
|
|
fail_unless (sock != NULL);
|
|
|
|
ostream = g_io_stream_get_output_stream (G_IO_STREAM (conn2));
|
|
fail_unless (ostream != NULL);
|
|
|
|
fail_unless (gst_rtsp_connection_create_from_socket (sock, "127.0.0.1",
|
|
4444, NULL, &rtsp_conn) == GST_RTSP_OK);
|
|
fail_unless (rtsp_conn != NULL);
|
|
|
|
/* should be possible to write on socket */
|
|
fail_unless (gst_rtsp_connection_poll (rtsp_conn, GST_RTSP_EV_WRITE, &event,
|
|
NULL) == GST_RTSP_OK);
|
|
fail_unless (event & GST_RTSP_EV_WRITE);
|
|
|
|
/* but not read, add timeout so that we don't block forever */
|
|
timeout = G_USEC_PER_SEC;
|
|
fail_unless (gst_rtsp_connection_poll_usec (rtsp_conn, GST_RTSP_EV_READ,
|
|
&event, timeout) == GST_RTSP_ETIMEOUT);
|
|
fail_if (event & GST_RTSP_EV_READ);
|
|
|
|
/* write on the other end and make sure socket can be read */
|
|
fail_unless (g_output_stream_write_all (ostream, "data", 5, &size, NULL,
|
|
NULL));
|
|
fail_unless (gst_rtsp_connection_poll (rtsp_conn, GST_RTSP_EV_READ, &event,
|
|
NULL) == GST_RTSP_OK);
|
|
fail_unless (event & GST_RTSP_EV_READ);
|
|
|
|
fail_unless (gst_rtsp_connection_close (rtsp_conn) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_free (rtsp_conn) == GST_RTSP_OK);
|
|
g_object_unref (conn1);
|
|
g_object_unref (conn2);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_rtspconnection_backlog)
|
|
{
|
|
GSocketConnection *conn1 = NULL;
|
|
GSocketConnection *conn2 = NULL;
|
|
GSocket *sock;
|
|
GstRTSPConnection *rtsp_conn = NULL;
|
|
GstRTSPWatch *watch;
|
|
GInputStream *istream;
|
|
guint8 *buffer;
|
|
guint8 recv[1024];
|
|
gsize count;
|
|
GstRTSPResult res = GST_RTSP_OK;
|
|
guint num_queued;
|
|
guint num_sent;
|
|
|
|
create_connection (&conn1, &conn2);
|
|
sock = g_socket_connection_get_socket (conn1);
|
|
fail_unless (sock != NULL);
|
|
|
|
fail_unless (gst_rtsp_connection_create_from_socket (sock, "127.0.0.1",
|
|
4444, NULL, &rtsp_conn) == GST_RTSP_OK);
|
|
fail_unless (rtsp_conn != NULL);
|
|
|
|
watch = gst_rtsp_watch_new (rtsp_conn, &watch_funcs, NULL, NULL);
|
|
fail_unless (watch != NULL);
|
|
fail_unless (gst_rtsp_watch_attach (watch, NULL) > 0);
|
|
g_source_unref ((GSource *) watch);
|
|
|
|
gst_rtsp_watch_set_send_backlog (watch, 1024, 0);
|
|
|
|
/* write until we fill tcp window and writes result in would_block,
|
|
* data will then start getting queued until the backlog also gets full */
|
|
num_queued = 0;
|
|
num_sent = 0;
|
|
while (res == GST_RTSP_OK) {
|
|
guint id = 0;
|
|
buffer = malloc (1024);
|
|
memset (buffer, 0, 1024);
|
|
res = gst_rtsp_watch_write_data (watch, buffer, 1024, &id);
|
|
if (id > 0)
|
|
num_queued++;
|
|
if (res == GST_RTSP_OK)
|
|
num_sent++;
|
|
}
|
|
|
|
/* make sure we got enomem and at least 1 message got queued */
|
|
fail_unless (res == GST_RTSP_ENOMEM);
|
|
fail_unless (num_queued > 0);
|
|
|
|
istream = g_io_stream_get_input_stream (G_IO_STREAM (conn2));
|
|
fail_unless (istream != NULL);
|
|
|
|
/* read a bit from the socket and make sure queued data gets sent */
|
|
while (num_queued > 0) {
|
|
fail_unless (g_input_stream_read_all (istream, recv, 1024, &count, NULL,
|
|
NULL));
|
|
num_sent--;
|
|
|
|
g_main_context_iteration (NULL, FALSE);
|
|
num_queued -= message_sent_count;
|
|
fail_unless (num_queued >= 0);
|
|
}
|
|
|
|
/* make sure we can read the rest of the data */
|
|
while (num_sent > 0) {
|
|
fail_unless (g_input_stream_read_all (istream, recv, 1024, &count, NULL,
|
|
NULL));
|
|
num_sent--;
|
|
}
|
|
|
|
g_source_destroy ((GSource *) watch);
|
|
fail_unless (gst_rtsp_connection_close (rtsp_conn) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_free (rtsp_conn) == GST_RTSP_OK);
|
|
g_object_unref (conn1);
|
|
g_object_unref (conn2);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_rtspconnection_ip)
|
|
{
|
|
GstRTSPConnection *conn = NULL;
|
|
GstRTSPUrl *url = NULL;
|
|
|
|
fail_unless (gst_rtsp_url_parse ("rtsp://127.0.0.1:42", &url) == GST_RTSP_OK);
|
|
fail_unless (url != NULL);
|
|
fail_unless (gst_rtsp_connection_create (url, &conn) == GST_RTSP_OK);
|
|
fail_unless (conn != NULL);
|
|
|
|
gst_rtsp_connection_set_ip (conn, "127.0.0.1");
|
|
fail_unless_equals_string (gst_rtsp_connection_get_ip (conn), "127.0.0.1");
|
|
|
|
gst_rtsp_url_free (url);
|
|
fail_unless (gst_rtsp_connection_free (conn) == GST_RTSP_OK);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_rtspconnection_send_receive_content_length)
|
|
{
|
|
GSocketConnection *input_conn = NULL;
|
|
GSocketConnection *output_conn = NULL;
|
|
GSocket *input_sock;
|
|
GSocket *output_sock;
|
|
GstRTSPConnection *rtsp_output_conn;
|
|
GstRTSPConnection *rtsp_input_conn;
|
|
GstRTSPMessage *msg;
|
|
|
|
create_connection (&input_conn, &output_conn);
|
|
input_sock = g_socket_connection_get_socket (input_conn);
|
|
fail_unless (input_sock != NULL);
|
|
output_sock = g_socket_connection_get_socket (output_conn);
|
|
fail_unless (output_sock != NULL);
|
|
|
|
fail_unless (gst_rtsp_connection_create_from_socket (input_sock, "127.0.0.1",
|
|
4444, NULL, &rtsp_input_conn) == GST_RTSP_OK);
|
|
fail_unless (rtsp_input_conn != NULL);
|
|
|
|
fail_unless (gst_rtsp_connection_create_from_socket (output_sock, "127.0.0.1",
|
|
4444, NULL, &rtsp_output_conn) == GST_RTSP_OK);
|
|
fail_unless (rtsp_output_conn != NULL);
|
|
|
|
/* send request message with to big payload */
|
|
fail_unless (gst_rtsp_message_new_request (&msg, GST_RTSP_SETUP,
|
|
"rtsp://example.com/") == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_message_add_header (msg, GST_RTSP_HDR_CONTENT_LENGTH,
|
|
"2000") == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_send (rtsp_output_conn, msg,
|
|
NULL) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_message_free (msg) == GST_RTSP_OK);
|
|
msg = NULL;
|
|
|
|
/* receive request message, expect ENOMEM */
|
|
gst_rtsp_connection_set_content_length_limit (rtsp_input_conn, 1000);
|
|
fail_unless (gst_rtsp_message_new (&msg) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_receive (rtsp_input_conn, msg, NULL) ==
|
|
GST_RTSP_ENOMEM);
|
|
fail_unless (gst_rtsp_message_free (msg) == GST_RTSP_OK);
|
|
msg = NULL;
|
|
|
|
/* send request message with negative payload */
|
|
fail_unless (gst_rtsp_message_new_request (&msg, GST_RTSP_SETUP,
|
|
"rtsp://example.com/") == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_message_add_header (msg, GST_RTSP_HDR_CONTENT_LENGTH,
|
|
"-2000") == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_send (rtsp_output_conn, msg,
|
|
NULL) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_message_free (msg) == GST_RTSP_OK);
|
|
msg = NULL;
|
|
|
|
/* receive request message, expect EPARSE */
|
|
fail_unless (gst_rtsp_message_new (&msg) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_receive (rtsp_input_conn, msg, NULL) ==
|
|
GST_RTSP_EPARSE);
|
|
fail_unless (gst_rtsp_message_free (msg) == GST_RTSP_OK);
|
|
msg = NULL;
|
|
|
|
fail_unless (gst_rtsp_connection_close (rtsp_input_conn) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_free (rtsp_input_conn) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_close (rtsp_output_conn) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_connection_free (rtsp_output_conn) == GST_RTSP_OK);
|
|
|
|
g_object_unref (input_conn);
|
|
g_object_unref (output_conn);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static Suite *
|
|
rtspconnection_suite (void)
|
|
{
|
|
Suite *s = suite_create ("rtsp support library(rtspconnection)");
|
|
TCase *tc_chain = tcase_create ("general");
|
|
|
|
suite_add_tcase (s, tc_chain);
|
|
tcase_add_test (tc_chain, test_rtspconnection_tunnel_setup);
|
|
tcase_add_test (tc_chain, test_rtspconnection_tunnel_setup_post_first);
|
|
tcase_add_test (tc_chain, test_rtspconnection_send_receive);
|
|
tcase_add_test (tc_chain, test_rtspconnection_send_receive_check_headers);
|
|
tcase_add_test (tc_chain, test_rtspconnection_connect);
|
|
tcase_add_test (tc_chain, test_rtspconnection_poll);
|
|
tcase_add_test (tc_chain, test_rtspconnection_backlog);
|
|
tcase_add_test (tc_chain, test_rtspconnection_ip);
|
|
tcase_add_test (tc_chain, test_rtspconnection_send_receive_content_length);
|
|
|
|
return s;
|
|
}
|
|
|
|
GST_CHECK_MAIN (rtspconnection);
|