mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-07-03 13:16:04 +00:00
rtspconnection: New unit test
See https://bugzilla.gnome.org/show_bug.cgi?id=724720
This commit is contained in:
Ognyan Tonchev
Wim Taymans
parent
ebe3530f51
commit
6bf215fa09
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@ -197,6 +197,7 @@ check_PROGRAMS = \
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libs/rtp \
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libs/rtp-basepayloading \
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libs/rtsp \
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libs/rtspconnection \
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libs/sdp \
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libs/tag \
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libs/video \
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@ -376,6 +377,14 @@ libs_rtsp_LDADD = \
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$(top_builddir)/gst-libs/gst/rtsp/libgstrtsp-@GST_API_VERSION@.la \
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$(GST_BASE_LIBS) $(LDADD)
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libs_rtspconnection_CFLAGS = \
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$(GST_PLUGINS_BASE_CFLAGS) \
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$(GIO_CFLAGS) \
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$(AM_CFLAGS)
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libs_rtspconnection_LDADD = \
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$(top_builddir)/gst-libs/gst/rtsp/libgstrtsp-@GST_API_VERSION@.la \
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$(GST_BASE_LIBS) $(GIO_LIBS) $(LDADD)
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libs_tag_CFLAGS = \
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$(GST_PLUGINS_BASE_CFLAGS) \
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$(GST_BASE_CFLAGS) \
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370
tests/check/libs/rtspconnection.c
Normal file
370
tests/check/libs/rtspconnection.c
Normal file
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@ -0,0 +1,370 @@
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/* 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"
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"x-sessioncookie: 805849328\r\n"
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"\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"
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"\r\n";
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static guint tunnel_start_count;
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static guint tunnel_complete_count;
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static guint tunnel_lost_count;
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static guint closed_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_start (GstRTSPWatch *watch, gpointer user_data)
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{
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tunnel_start_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_complete (GstRTSPWatch *watch, gpointer user_data)
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{
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tunnel_complete_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 GstRTSPWatchFuncs watch_funcs = {
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NULL,
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NULL,
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closed,
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NULL,
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tunnel_start,
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tunnel_complete,
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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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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_start_count == 1);
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fail_unless (tunnel_complete_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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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_start_count == 1);
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fail_unless (tunnel_complete_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_start_count == 1);
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fail_unless (tunnel_complete_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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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_start_count == 1);
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fail_unless (tunnel_complete_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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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 (client_get);
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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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static Suite *
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rtspconnection_suite (void)
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{
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Suite *s = suite_create ("rtsp support library(rtspconnection)");
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TCase *tc_chain = tcase_create ("general");
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suite_add_tcase (s, tc_chain);
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tcase_add_test (tc_chain, test_rtspconnection_tunnel_setup);
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return s;
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}
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int
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main (int argc, char **argv)
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{
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int nf;
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Suite *s = rtspconnection_suite ();
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SRunner *sr = srunner_create (s);
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gst_check_init (&argc, &argv);
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srunner_run_all (sr, CK_NORMAL);
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nf = srunner_ntests_failed (sr);
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srunner_free (sr);
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return nf;
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}
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