1 /* GStreamer unit test for rtspclientsink
2 * Copyright (C) 2012 Axis Communications <dev-gstreamer at axis dot com>
3 * @author David Svensson Fors <davidsf at axis dot com>
4 * Copyright (C) 2015 Centricular Ltd
5 * @author Tim-Philipp Müller <tim@centricular.com>
6 * @author Jan Schmidt <jan@centricular.com>
8 * This library is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Library General Public
10 * License as published by the Free Software Foundation; either
11 * version 2 of the License, or (at your option) any later version.
13 * This library is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16 * Library General Public License for more details.
18 * You should have received a copy of the GNU Library General Public
19 * License along with this library; if not, write to the
20 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
21 * Boston, MA 02110-1301, USA.
24 #include <gst/check/gstcheck.h>
25 #include <gst/sdp/gstsdpmessage.h>
26 #include <gst/rtp/gstrtpbuffer.h>
27 #include <gst/rtp/gstrtcpbuffer.h>
30 #include <netinet/in.h>
32 #include "rtsp-server.h"
34 #define TEST_MOUNT_POINT "/test"
36 /* tested rtsp server */
37 static GstRTSPServer *server = NULL;
39 /* tcp port that the test server listens for rtsp requests on */
40 static gint test_port = 0;
42 /* id of the server's source within the GMainContext */
43 static guint source_id;
45 /* iterate the default main context until there are no events to dispatch */
49 while (g_main_context_iteration (NULL, FALSE)) {
50 GST_DEBUG ("iteration");
54 /* start the testing rtsp server for RECORD mode */
55 static GstRTSPMediaFactory *
56 start_record_server (const gchar * launch_line)
58 GstRTSPMediaFactory *factory;
59 GstRTSPMountPoints *mounts;
62 mounts = gst_rtsp_server_get_mount_points (server);
64 factory = gst_rtsp_media_factory_new ();
66 gst_rtsp_media_factory_set_transport_mode (factory,
67 GST_RTSP_TRANSPORT_MODE_RECORD);
68 gst_rtsp_media_factory_set_launch (factory, launch_line);
69 gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
70 g_object_unref (mounts);
73 gst_rtsp_server_set_service (server, "0");
75 /* attach to default main context */
76 source_id = gst_rtsp_server_attach (server, NULL);
77 fail_if (source_id == 0);
80 service = gst_rtsp_server_get_service (server);
81 test_port = atoi (service);
82 fail_unless (test_port != 0);
85 GST_DEBUG ("rtsp server listening on port %d", test_port);
89 /* stop the tested rtsp server */
93 g_source_remove (source_id);
96 GST_DEBUG ("rtsp server stopped");
99 /* fixture setup function */
103 server = gst_rtsp_server_new ();
106 /* fixture clean-up function */
111 g_object_unref (server);
117 /* create an rtsp connection to the server on test_port */
119 get_server_uri (gint port, const gchar * mount_point)
123 GstRTSPUrl *url = NULL;
125 address = gst_rtsp_server_get_address (server);
126 uri_string = g_strdup_printf ("rtsp://%s:%d%s", address, port, mount_point);
129 fail_unless (gst_rtsp_url_parse (uri_string, &url) == GST_RTSP_OK);
130 gst_rtsp_url_free (url);
136 media_constructed_cb (GstRTSPMediaFactory * mfactory, GstRTSPMedia * media,
139 GstElement **p_sink = user_data;
142 bin = gst_rtsp_media_get_element (media);
143 *p_sink = gst_bin_get_by_name (GST_BIN (bin), "sink");
144 GST_INFO ("media constructed!: %" GST_PTR_FORMAT, *p_sink);
147 #define AUDIO_PIPELINE "audiotestsrc num-buffers=%d ! " \
148 "audio/x-raw,rate=8000 ! alawenc ! rtspclientsink name=sink location=%s"
149 #define RECORD_N_BUFS 10
151 GST_START_TEST (test_record)
153 GstRTSPMediaFactory *mfactory;
154 GstElement *server_sink = NULL;
158 start_record_server ("( rtppcmadepay name=depay0 ! appsink name=sink )");
160 g_signal_connect (mfactory, "media-constructed",
161 G_CALLBACK (media_constructed_cb), &server_sink);
163 /* Create an rtspclientsink and send some data */
165 gchar *uri = get_server_uri (test_port, TEST_MOUNT_POINT);
166 gchar *pipe_str = g_strdup_printf (AUDIO_PIPELINE,
169 GstElement *pipeline;
172 pipeline = gst_parse_launch (pipe_str, NULL);
173 fail_unless (pipeline != NULL);
175 bus = gst_element_get_bus (pipeline);
176 fail_if (bus == NULL);
178 gst_element_set_state (pipeline, GST_STATE_PLAYING);
180 msg = gst_bus_poll (bus, GST_MESSAGE_EOS | GST_MESSAGE_ERROR, -1);
181 fail_if (GST_MESSAGE_TYPE (msg) != GST_MESSAGE_EOS);
182 gst_message_unref (msg);
184 gst_element_set_state (pipeline, GST_STATE_NULL);
185 gst_object_unref (pipeline);
190 /* check received data (we assume every buffer created by audiotestsrc and
191 * subsequently encoded by mulawenc results in exactly one RTP packet) */
192 for (i = 0; i < RECORD_N_BUFS; ++i) {
193 GstSample *sample = NULL;
195 g_signal_emit_by_name (G_OBJECT (server_sink), "pull-sample", &sample);
196 GST_INFO ("%2d recv sample: %p", i, sample);
198 gst_sample_unref (sample);
201 /* clean up and iterate so the clean-up can finish */
209 rtspclientsink_suite (void)
211 Suite *s = suite_create ("rtspclientsink");
212 TCase *tc = tcase_create ("general");
214 suite_add_tcase (s, tc);
215 tcase_add_checked_fixture (tc, setup, teardown);
216 tcase_set_timeout (tc, 120);
217 tcase_add_test (tc, test_record);
221 GST_CHECK_MAIN (rtspclientsink);