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884 lines
29 KiB
C
884 lines
29 KiB
C
/* GStreamer unit tests for the GstRTSPConnection API (RTSP support
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* library)
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*
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* Copyright (C) 2014 Ognyan Tonchev <ognyan axis com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/check/gstcheck.h>
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#include <gst/rtsp/gstrtspconnection.h>
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#include <sys/socket.h>
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#include <netinet/in.h>
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#include <string.h>
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static const gchar *get_msg =
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"GET /example/url HTTP/1.0\r\n"
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"Host: 127.0.0.1\r\n" "x-sessioncookie: 805849328\r\n\r\n";
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static const gchar *post_msg =
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"POST /example/url HTTP/1.0\r\n"
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"Host: 127.0.0.1\r\n"
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"x-sessioncookie: 805849328\r\n"
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"Content-Length: 0\r\n"
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"Content-Type: application/x-rtsp-tunnelled\r\n\r\n";
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static guint tunnel_get_count;
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static guint tunnel_post_count;
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static guint tunnel_lost_count;
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static guint closed_count;
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static guint message_sent_count;
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typedef struct
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{
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GMainLoop *loop;
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guint16 port;
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GSocketConnection *conn;
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GMutex mutex;
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GCond cond;
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gboolean started;
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} ServiceData;
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static gboolean
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incoming_callback (GSocketService * service, GSocketConnection * connection,
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GObject * source_object, gpointer user_data)
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{
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ServiceData *data = user_data;
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GST_DEBUG ("new incoming connection");
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data->conn = g_object_ref (connection);
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g_main_loop_quit (data->loop);
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return FALSE;
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}
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static gpointer
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service_thread_func (gpointer user_data)
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{
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ServiceData *data = user_data;
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GMainContext *service_context;
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GSocketService *service;
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service_context = g_main_context_new ();
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g_main_context_push_thread_default (service_context);
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data->loop = g_main_loop_new (service_context, FALSE);
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/* find available port and start service */
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service = g_socket_service_new ();
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data->port = g_socket_listener_add_any_inet_port ((GSocketListener *) service,
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NULL, NULL);
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fail_unless (data->port != 0);
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/* get notified upon new connection */
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g_signal_connect (service, "incoming", G_CALLBACK (incoming_callback), data);
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g_socket_service_start (service);
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/* service is started */
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g_mutex_lock (&data->mutex);
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data->started = TRUE;
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g_cond_signal (&data->cond);
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g_mutex_unlock (&data->mutex);
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/* our service will run in the main context of this main loop */
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g_main_loop_run (data->loop);
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g_main_context_pop_thread_default (service_context);
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g_main_loop_unref (data->loop);
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data->loop = NULL;
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return NULL;
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}
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static void
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create_connection (GSocketConnection ** client_conn,
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GSocketConnection ** server_conn)
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{
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ServiceData *data;
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GThread *service_thread;
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GSocketClient *client = g_socket_client_new ();
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data = g_new0 (ServiceData, 1);
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g_mutex_init (&data->mutex);
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g_cond_init (&data->cond);
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service_thread = g_thread_new ("service thread", service_thread_func, data);
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fail_unless (service_thread != NULL);
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/* wait for the service to start */
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g_mutex_lock (&data->mutex);
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while (!data->started) {
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g_cond_wait (&data->cond, &data->mutex);
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}
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g_mutex_unlock (&data->mutex);
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/* create the tcp link */
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*client_conn = g_socket_client_connect_to_host (client, (gchar *) "localhost",
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data->port, NULL, NULL);
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fail_unless (*client_conn != NULL);
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fail_unless (g_socket_connection_is_connected (*client_conn));
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g_thread_join (service_thread);
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*server_conn = data->conn;
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data->conn = NULL;
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fail_unless (g_socket_connection_is_connected (*server_conn));
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g_mutex_clear (&data->mutex);
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g_cond_clear (&data->cond);
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g_free (data);
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g_object_unref (client);
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}
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static GstRTSPStatusCode
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tunnel_get (GstRTSPWatch * watch, gpointer user_data)
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{
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tunnel_get_count++;
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return GST_RTSP_STS_OK;
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}
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static GstRTSPResult
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tunnel_post (GstRTSPWatch * watch, gpointer user_data)
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{
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tunnel_post_count++;
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return GST_RTSP_OK;
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}
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static GstRTSPResult
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tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
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{
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tunnel_lost_count++;
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return GST_RTSP_OK;
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}
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static GstRTSPResult
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closed (GstRTSPWatch * watch, gpointer user_data)
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{
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closed_count++;
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return GST_RTSP_OK;
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}
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static GstRTSPResult
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message_sent (GstRTSPWatch * watch, guint id, gpointer user_data)
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{
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message_sent_count++;
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return GST_RTSP_OK;
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}
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static GstRTSPWatchFuncs watch_funcs = {
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NULL,
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message_sent,
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closed,
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NULL,
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tunnel_get,
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tunnel_post,
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NULL,
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tunnel_lost
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};
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/* setts up a new tunnel, then disconnects the read connection and creates it
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* again */
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GST_START_TEST (test_rtspconnection_tunnel_setup)
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{
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GstRTSPConnection *rtsp_conn1 = NULL;
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GstRTSPConnection *rtsp_conn2 = NULL;
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GstRTSPWatch *watch1;
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GstRTSPWatch *watch2;
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GstRTSPResult res;
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GSocketConnection *client_get = NULL;
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GSocketConnection *server_get = NULL;
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GSocketConnection *client_post = NULL;
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GSocketConnection *server_post = NULL;
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GSocket *server_sock;
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GOutputStream *ostream_get;
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GInputStream *istream_get;
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GOutputStream *ostream_post;
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gsize size = 0;
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gchar buffer[1024];
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/* create GET connection */
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create_connection (&client_get, &server_get);
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server_sock = g_socket_connection_get_socket (server_get);
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fail_unless (server_sock != NULL);
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res = gst_rtsp_connection_create_from_socket (server_sock, "127.0.0.1", 4444,
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NULL, &rtsp_conn1);
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fail_unless (res == GST_RTSP_OK);
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fail_unless (rtsp_conn1 != NULL);
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watch1 = gst_rtsp_watch_new (rtsp_conn1, &watch_funcs, NULL, NULL);
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fail_unless (watch1 != NULL);
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fail_unless (gst_rtsp_watch_attach (watch1, NULL) > 0);
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g_source_unref ((GSource *) watch1);
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ostream_get = g_io_stream_get_output_stream (G_IO_STREAM (client_get));
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fail_unless (ostream_get != NULL);
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istream_get = g_io_stream_get_input_stream (G_IO_STREAM (client_get));
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fail_unless (istream_get != NULL);
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/* initiate the tunnel by sending HTTP GET */
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fail_unless (g_output_stream_write_all (ostream_get, get_msg,
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strlen (get_msg), &size, NULL, NULL));
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fail_unless (size == strlen (get_msg));
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while (!g_main_context_iteration (NULL, TRUE));
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fail_unless (tunnel_get_count == 1);
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fail_unless (tunnel_post_count == 0);
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fail_unless (tunnel_lost_count == 0);
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fail_unless (closed_count == 0);
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/* read the HTTP GET response */
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size = g_input_stream_read (istream_get, buffer, 1024, NULL, NULL);
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fail_unless (size > 0);
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buffer[size] = 0;
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fail_unless (g_strrstr (buffer, "HTTP/1.0 200 OK") != NULL);
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/* create POST channel */
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create_connection (&client_post, &server_post);
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server_sock = g_socket_connection_get_socket (server_post);
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fail_unless (server_sock != NULL);
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res = gst_rtsp_connection_create_from_socket (server_sock, "127.0.0.1", 4444,
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NULL, &rtsp_conn2);
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fail_unless (res == GST_RTSP_OK);
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fail_unless (rtsp_conn2 != NULL);
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watch2 = gst_rtsp_watch_new (rtsp_conn2, &watch_funcs, NULL, NULL);
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fail_unless (watch2 != NULL);
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fail_unless (gst_rtsp_watch_attach (watch2, NULL) > 0);
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g_source_unref ((GSource *) watch2);
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ostream_post = g_io_stream_get_output_stream (G_IO_STREAM (client_post));
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fail_unless (ostream_post != NULL);
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/* complete the tunnel by sending HTTP POST */
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fail_unless (g_output_stream_write_all (ostream_post, post_msg,
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strlen (post_msg), &size, NULL, NULL));
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fail_unless (size == strlen (post_msg));
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while (!g_main_context_iteration (NULL, TRUE));
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fail_unless (tunnel_get_count == 1);
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fail_unless (tunnel_post_count == 1);
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fail_unless (tunnel_lost_count == 0);
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fail_unless (closed_count == 0);
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/* merge the two connections together */
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fail_unless (gst_rtsp_connection_do_tunnel (rtsp_conn1, rtsp_conn2) ==
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GST_RTSP_OK);
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gst_rtsp_watch_reset (watch1);
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g_source_destroy ((GSource *) watch2);
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gst_rtsp_connection_free (rtsp_conn2);
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rtsp_conn2 = NULL;
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/* it must be possible to reconnect the POST channel */
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g_object_unref (client_post);
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while (!g_main_context_iteration (NULL, TRUE));
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fail_unless (tunnel_get_count == 1);
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fail_unless (tunnel_post_count == 1);
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fail_unless (tunnel_lost_count == 1);
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fail_unless (closed_count == 0);
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g_object_unref (server_post);
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/* no other source should get dispatched */
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fail_if (g_main_context_iteration (NULL, FALSE));
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/* create new POST connection */
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create_connection (&client_post, &server_post);
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server_sock = g_socket_connection_get_socket (server_post);
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fail_unless (server_sock != NULL);
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res = gst_rtsp_connection_create_from_socket (server_sock, "127.0.0.1", 4444,
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NULL, &rtsp_conn2);
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fail_unless (res == GST_RTSP_OK);
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fail_unless (rtsp_conn2 != NULL);
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watch2 = gst_rtsp_watch_new (rtsp_conn2, &watch_funcs, NULL, NULL);
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fail_unless (watch2 != NULL);
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fail_unless (gst_rtsp_watch_attach (watch2, NULL) > 0);
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g_source_unref ((GSource *) watch2);
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ostream_post = g_io_stream_get_output_stream (G_IO_STREAM (client_post));
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fail_unless (ostream_post != NULL);
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/* complete the tunnel by sending HTTP POST */
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fail_unless (g_output_stream_write_all (ostream_post, post_msg,
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strlen (post_msg), &size, NULL, NULL));
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fail_unless (size == strlen (post_msg));
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while (!g_main_context_iteration (NULL, TRUE));
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fail_unless (tunnel_get_count == 1);
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fail_unless (tunnel_post_count == 2);
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fail_unless (tunnel_lost_count == 1);
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fail_unless (closed_count == 0);
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/* merge the two connections together */
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fail_unless (gst_rtsp_connection_do_tunnel (rtsp_conn1, rtsp_conn2) ==
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GST_RTSP_OK);
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gst_rtsp_watch_reset (watch1);
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g_source_destroy ((GSource *) watch2);
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gst_rtsp_connection_free (rtsp_conn2);
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rtsp_conn2 = NULL;
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/* check if rtspconnection can detect close of the get channel */
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g_object_unref (client_get);
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while (!g_main_context_iteration (NULL, TRUE));
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fail_unless (tunnel_get_count == 1);
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fail_unless (tunnel_post_count == 2);
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fail_unless (tunnel_lost_count == 1);
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fail_unless (closed_count == 1);
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fail_unless (gst_rtsp_connection_close (rtsp_conn1) == GST_RTSP_OK);
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fail_unless (gst_rtsp_connection_free (rtsp_conn1) == GST_RTSP_OK);
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g_object_unref (client_post);
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g_object_unref (server_post);
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g_object_unref (server_get);
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}
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GST_END_TEST;
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/* setts up a new tunnel, starting with the read channel,
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* then disconnects the read connection and creates it again
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* ideally this test should be merged with test_rtspconnection_tunnel_setup but
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* but it became quite messy */
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GST_START_TEST (test_rtspconnection_tunnel_setup_post_first)
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{
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GstRTSPConnection *rtsp_conn1 = NULL;
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GstRTSPConnection *rtsp_conn2 = NULL;
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GstRTSPWatch *watch1;
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GstRTSPWatch *watch2;
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GstRTSPResult res;
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GSocketConnection *client_get = NULL;
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GSocketConnection *server_get = NULL;
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GSocketConnection *client_post = NULL;
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GSocketConnection *server_post = NULL;
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GSocket *server_sock;
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GOutputStream *ostream_get;
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GInputStream *istream_get;
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GOutputStream *ostream_post;
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gsize size = 0;
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gchar buffer[1024];
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/* create POST channel */
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create_connection (&client_post, &server_post);
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server_sock = g_socket_connection_get_socket (server_post);
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fail_unless (server_sock != NULL);
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res = gst_rtsp_connection_create_from_socket (server_sock, "127.0.0.1", 4444,
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NULL, &rtsp_conn1);
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fail_unless (res == GST_RTSP_OK);
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fail_unless (rtsp_conn1 != NULL);
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watch1 = gst_rtsp_watch_new (rtsp_conn1, &watch_funcs, NULL, NULL);
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fail_unless (watch1 != NULL);
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fail_unless (gst_rtsp_watch_attach (watch1, NULL) > 0);
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g_source_unref ((GSource *) watch1);
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ostream_post = g_io_stream_get_output_stream (G_IO_STREAM (client_post));
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fail_unless (ostream_post != NULL);
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/* initiate the tunnel by sending HTTP POST */
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fail_unless (g_output_stream_write_all (ostream_post, post_msg,
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strlen (post_msg), &size, NULL, NULL));
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fail_unless (size == strlen (post_msg));
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while (!g_main_context_iteration (NULL, TRUE));
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fail_unless (tunnel_get_count == 0);
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fail_unless (tunnel_post_count == 1);
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fail_unless (tunnel_lost_count == 0);
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fail_unless (closed_count == 0);
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/* create GET connection */
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create_connection (&client_get, &server_get);
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server_sock = g_socket_connection_get_socket (server_get);
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fail_unless (server_sock != NULL);
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res = gst_rtsp_connection_create_from_socket (server_sock, "127.0.0.1", 4444,
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NULL, &rtsp_conn2);
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fail_unless (res == GST_RTSP_OK);
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fail_unless (rtsp_conn2 != NULL);
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watch2 = gst_rtsp_watch_new (rtsp_conn2, &watch_funcs, NULL, NULL);
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fail_unless (watch2 != NULL);
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fail_unless (gst_rtsp_watch_attach (watch2, NULL) > 0);
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g_source_unref ((GSource *) watch2);
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ostream_get = g_io_stream_get_output_stream (G_IO_STREAM (client_get));
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fail_unless (ostream_get != NULL);
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istream_get = g_io_stream_get_input_stream (G_IO_STREAM (client_get));
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fail_unless (istream_get != NULL);
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/* complete the tunnel by sending HTTP GET */
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fail_unless (g_output_stream_write_all (ostream_get, get_msg,
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strlen (get_msg), &size, NULL, NULL));
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fail_unless (size == strlen (get_msg));
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while (!g_main_context_iteration (NULL, TRUE));
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fail_unless (tunnel_get_count == 1);
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fail_unless (tunnel_post_count == 1);
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fail_unless (tunnel_lost_count == 0);
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fail_unless (closed_count == 0);
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/* read the HTTP GET response */
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size = g_input_stream_read (istream_get, buffer, 1024, NULL, NULL);
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fail_unless (size > 0);
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buffer[size] = 0;
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fail_unless (g_strrstr (buffer, "HTTP/1.0 200 OK") != NULL);
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/* merge the two connections together */
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fail_unless (gst_rtsp_connection_do_tunnel (rtsp_conn1, rtsp_conn2) ==
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GST_RTSP_OK);
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gst_rtsp_watch_reset (watch1);
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g_source_destroy ((GSource *) watch2);
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gst_rtsp_connection_free (rtsp_conn2);
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rtsp_conn2 = NULL;
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/* it must be possible to reconnect the POST channel */
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g_object_unref (client_post);
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while (!g_main_context_iteration (NULL, TRUE));
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fail_unless (tunnel_get_count == 1);
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fail_unless (tunnel_post_count == 1);
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fail_unless (tunnel_lost_count == 1);
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fail_unless (closed_count == 0);
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g_object_unref (server_post);
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/* no other source should get dispatched */
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fail_if (g_main_context_iteration (NULL, FALSE));
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/* create new POST connection */
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create_connection (&client_post, &server_post);
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server_sock = g_socket_connection_get_socket (server_post);
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fail_unless (server_sock != NULL);
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|
|
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;
|
|
GTimeVal tv;
|
|
|
|
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 */
|
|
tv.tv_sec = 1;
|
|
tv.tv_usec = 0;
|
|
fail_unless (gst_rtsp_connection_poll (rtsp_conn, GST_RTSP_EV_READ, &event,
|
|
&tv) == 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;
|
|
|
|
|
|
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);
|
|
|
|
return s;
|
|
}
|
|
|
|
GST_CHECK_MAIN (rtspconnection);
|