/* GStreamer unit test for rtspclientsink * Copyright (C) 2012 Axis Communications * @author David Svensson Fors * Copyright (C) 2015 Centricular Ltd * @author Tim-Philipp Müller * @author Jan Schmidt * * 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. */ #include #include #include #include #include #include #include "rtsp-server.h" #define TEST_MOUNT_POINT "/test" /* tested rtsp server */ static GstRTSPServer *server = NULL; /* tcp port that the test server listens for rtsp requests on */ static gint test_port = 0; /* id of the server's source within the GMainContext */ static guint source_id; /* iterate the default main context until there are no events to dispatch */ static void iterate (void) { while (g_main_context_iteration (NULL, FALSE)) { GST_DEBUG ("iteration"); } } /* start the testing rtsp server for RECORD mode */ static GstRTSPMediaFactory * start_record_server (const gchar * launch_line) { GstRTSPMediaFactory *factory; GstRTSPMountPoints *mounts; gchar *service; mounts = gst_rtsp_server_get_mount_points (server); factory = gst_rtsp_media_factory_new (); gst_rtsp_media_factory_set_transport_mode (factory, GST_RTSP_TRANSPORT_MODE_RECORD); gst_rtsp_media_factory_set_launch (factory, launch_line); gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory); g_object_unref (mounts); /* set port to any */ gst_rtsp_server_set_service (server, "0"); /* attach to default main context */ source_id = gst_rtsp_server_attach (server, NULL); fail_if (source_id == 0); /* get port */ service = gst_rtsp_server_get_service (server); test_port = atoi (service); fail_unless (test_port != 0); g_free (service); GST_DEBUG ("rtsp server listening on port %d", test_port); return factory; } /* stop the tested rtsp server */ static void stop_server (void) { g_source_remove (source_id); source_id = 0; GST_DEBUG ("rtsp server stopped"); } /* fixture setup function */ static void setup (void) { server = gst_rtsp_server_new (); } /* fixture clean-up function */ static void teardown (void) { if (server) { g_object_unref (server); server = NULL; } test_port = 0; } /* create an rtsp connection to the server on test_port */ static gchar * get_server_uri (gint port, const gchar * mount_point) { gchar *address; gchar *uri_string; GstRTSPUrl *url = NULL; address = gst_rtsp_server_get_address (server); uri_string = g_strdup_printf ("rtsp://%s:%d%s", address, port, mount_point); g_free (address); fail_unless (gst_rtsp_url_parse (uri_string, &url) == GST_RTSP_OK); gst_rtsp_url_free (url); return uri_string; } static void media_constructed_cb (GstRTSPMediaFactory * mfactory, GstRTSPMedia * media, gpointer user_data) { GstElement **p_sink = user_data; GstElement *bin; bin = gst_rtsp_media_get_element (media); *p_sink = gst_bin_get_by_name (GST_BIN (bin), "sink"); GST_INFO ("media constructed!: %" GST_PTR_FORMAT, *p_sink); gst_object_unref (bin); } #define AUDIO_PIPELINE "audiotestsrc num-buffers=%d ! " \ "audio/x-raw,rate=8000 ! alawenc ! rtspclientsink name=sink location=%s" #define RECORD_N_BUFS 10 GST_START_TEST (test_record) { GstRTSPMediaFactory *mfactory; GstElement *server_sink = NULL; gint i; mfactory = start_record_server ("( rtppcmadepay name=depay0 ! appsink name=sink )"); g_signal_connect (mfactory, "media-constructed", G_CALLBACK (media_constructed_cb), &server_sink); /* Create an rtspclientsink and send some data */ { gchar *uri = get_server_uri (test_port, TEST_MOUNT_POINT); gchar *pipe_str; GstMessage *msg; GstElement *pipeline; GstBus *bus; pipe_str = g_strdup_printf (AUDIO_PIPELINE, RECORD_N_BUFS, uri); g_free (uri); pipeline = gst_parse_launch (pipe_str, NULL); g_free (pipe_str); fail_unless (pipeline != NULL); bus = gst_element_get_bus (pipeline); fail_if (bus == NULL); gst_element_set_state (pipeline, GST_STATE_PLAYING); msg = gst_bus_poll (bus, GST_MESSAGE_EOS | GST_MESSAGE_ERROR, -1); fail_if (GST_MESSAGE_TYPE (msg) != GST_MESSAGE_EOS); gst_message_unref (msg); gst_element_set_state (pipeline, GST_STATE_NULL); gst_object_unref (pipeline); } iterate (); /* check received data (we assume every buffer created by audiotestsrc and * subsequently encoded by mulawenc results in exactly one RTP packet) */ for (i = 0; i < RECORD_N_BUFS; ++i) { GstSample *sample = NULL; g_signal_emit_by_name (G_OBJECT (server_sink), "pull-sample", &sample); GST_INFO ("%2d recv sample: %p", i, sample); if (sample) gst_sample_unref (sample); } /* clean up and iterate so the clean-up can finish */ stop_server (); iterate (); } GST_END_TEST; static Suite * rtspclientsink_suite (void) { Suite *s = suite_create ("rtspclientsink"); TCase *tc = tcase_create ("general"); suite_add_tcase (s, tc); tcase_add_checked_fixture (tc, setup, teardown); tcase_set_timeout (tc, 120); tcase_add_test (tc, test_record); return s; } GST_CHECK_MAIN (rtspclientsink);