-/* GStreamer
- *
- * unit test for GstRTSPServer
- *
+/* GStreamer unit test for GstRTSPServer
* Copyright (C) 2012 Axis Communications <dev-gstreamer at axis dot com>
- * @author David Svensson Fors <davidsf at axis dot com>
+ * @author David Svensson Fors <davidsf at axis dot com>
+ * Copyright (C) 2015 Centricular Ltd
+ * @author Tim-Philipp Müller <tim@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
GST_DEBUG ("rtsp server listening on port %d", test_port);
}
+/* 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_record (factory, TRUE);
+ 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)
}
if (use_tcp_transport) {
- transport_string_in =
- g_strdup_printf (TEST_PROTO_TCP ";unicast");
+ transport_string_in = g_strdup_printf (TEST_PROTO_TCP ";unicast");
} else {
transport_string_in =
- g_strdup_printf (TEST_PROTO ";unicast;client_port=%d-%d",
- client_ports->min, client_ports->max);
+ g_strdup_printf (TEST_PROTO ";unicast;client_port=%d-%d",
+ client_ports->min, client_ports->max);
}
code =
do_request_full (conn, GST_RTSP_SETUP, control, session_in,
GST_END_TEST;
+GST_START_TEST (test_describe_record_media)
+{
+ GstRTSPConnection *conn;
+
+ start_record_server ("( fakesink name=depay0 )");
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ /* send DESCRIBE request */
+ fail_unless_equals_int (do_request (conn, GST_RTSP_DESCRIBE, NULL, NULL, NULL,
+ NULL, NULL, NULL, NULL, NULL, NULL, NULL),
+ GST_RTSP_STS_METHOD_NOT_ALLOWED);
+
+ /* clean up and iterate so the clean-up can finish */
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
GST_START_TEST (test_describe_non_existing_mount_point)
{
GstRTSPConnection *conn;
/* session can not be the same */
fail_unless (strcmp (session1, session2));
- /* send TEARDOWN request for the first session*/
+ /* send TEARDOWN request for the first session */
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
session1) == GST_RTSP_STS_OK);
GST_END_TEST;
+GST_START_TEST (test_announce_without_sdp)
+{
+ GstRTSPConnection *conn;
+ GstRTSPStatusCode status;
+ GstRTSPMessage *request;
+ GstRTSPMessage *response;
+
+ start_record_server ("( fakesink name=depay0 )");
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ /* create and send ANNOUNCE request */
+ request = create_request (conn, GST_RTSP_ANNOUNCE, NULL);
+
+ fail_unless (send_request (conn, request));
+
+ iterate ();
+
+ response = read_response (conn);
+
+ /* check response */
+ gst_rtsp_message_parse_response (response, &status, NULL, NULL);
+ fail_unless_equals_int (status, GST_RTSP_STS_BAD_REQUEST);
+ gst_rtsp_message_free (response);
+
+ /* try again, this type with content-type, but still no SDP */
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
+ "application/sdp");
+
+ fail_unless (send_request (conn, request));
+
+ iterate ();
+
+ response = read_response (conn);
+
+ /* check response */
+ gst_rtsp_message_parse_response (response, &status, NULL, NULL);
+ fail_unless_equals_int (status, GST_RTSP_STS_BAD_REQUEST);
+ gst_rtsp_message_free (response);
+
+ /* try again, this type with an unknown content-type */
+ gst_rtsp_message_remove_header (request, GST_RTSP_HDR_CONTENT_TYPE, -1);
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
+ "application/x-something");
+
+ fail_unless (send_request (conn, request));
+
+ iterate ();
+
+ response = read_response (conn);
+
+ /* check response */
+ gst_rtsp_message_parse_response (response, &status, NULL, NULL);
+ fail_unless_equals_int (status, GST_RTSP_STS_BAD_REQUEST);
+ gst_rtsp_message_free (response);
+
+ /* clean up and iterate so the clean-up can finish */
+ gst_rtsp_message_free (request);
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+static GstRTSPStatusCode
+do_announce (GstRTSPConnection * conn, GstSDPMessage * sdp)
+{
+ GstRTSPMessage *request;
+ GstRTSPMessage *response;
+ GstRTSPStatusCode code;
+ gchar *str;
+
+ /* create request */
+ request = create_request (conn, GST_RTSP_ANNOUNCE, NULL);
+
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
+ "application/sdp");
+
+ /* add SDP to the response body */
+ str = gst_sdp_message_as_text (sdp);
+ gst_rtsp_message_take_body (request, (guint8 *) str, strlen (str));
+ gst_sdp_message_free (sdp);
+
+ /* send request */
+ fail_unless (send_request (conn, request));
+ gst_rtsp_message_free (request);
+
+ iterate ();
+
+ /* read response */
+ response = read_response (conn);
+
+ /* check status line */
+ gst_rtsp_message_parse_response (response, &code, NULL, NULL);
+
+ gst_rtsp_message_free (response);
+ return code;
+}
+
+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);
+}
+
+#define RECORD_N_BUFS 10
+
+GST_START_TEST (test_record_tcp)
+{
+ GstRTSPMediaFactory *mfactory;
+ GstRTSPConnection *conn;
+ GstRTSPStatusCode status;
+ GstRTSPMessage *response;
+ GstRTSPMessage *request;
+ GstSDPMessage *sdp;
+ GstRTSPResult rres;
+ GSocketAddress *sa;
+ GInetAddress *ia;
+ GstElement *sink = NULL;
+ GSocket *conn_socket;
+ const gchar *proto;
+ gchar *client_ip, *sess_id, *session = NULL;
+ gint i;
+
+ mfactory =
+ start_record_server ("( rtppcmadepay name=depay0 ! appsink name=sink )");
+
+ g_signal_connect (mfactory, "media-constructed",
+ G_CALLBACK (media_constructed_cb), &sink);
+
+ conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+ conn_socket = gst_rtsp_connection_get_read_socket (conn);
+
+ sa = g_socket_get_local_address (conn_socket, NULL);
+ ia = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (sa));
+ client_ip = g_inet_address_to_string (ia);
+ if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV6)
+ proto = "IP6";
+ else if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV4)
+ proto = "IP4";
+ else
+ g_assert_not_reached ();
+ g_object_unref (sa);
+
+ gst_sdp_message_new (&sdp);
+
+ /* some standard things first */
+ gst_sdp_message_set_version (sdp, "0");
+
+ /* session ID doesn't have to be super-unique in this case */
+ sess_id = g_strdup_printf ("%u", g_random_int ());
+ gst_sdp_message_set_origin (sdp, "-", sess_id, "1", "IN", proto, client_ip);
+ g_free (sess_id);
+ g_free (client_ip);
+
+ gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
+ gst_sdp_message_set_information (sdp, "rtsp-server-test");
+ gst_sdp_message_add_time (sdp, "0", "0", NULL);
+ gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
+
+ /* add stream 0 */
+ {
+ GstSDPMedia *smedia;
+
+ gst_sdp_media_new (&smedia);
+ gst_sdp_media_set_media (smedia, "audio");
+ gst_sdp_media_add_format (smedia, "8"); /* pcma/alaw */
+ gst_sdp_media_set_port_info (smedia, 0, 1);
+ gst_sdp_media_set_proto (smedia, "RTP/AVP");
+ gst_sdp_media_add_attribute (smedia, "rtpmap", "8 PCMA/8000");
+ gst_sdp_message_add_media (sdp, smedia);
+ gst_sdp_media_free (smedia);
+ }
+
+ /* send ANNOUNCE request */
+ status = do_announce (conn, sdp);
+ fail_unless_equals_int (status, GST_RTSP_STS_OK);
+
+ /* create and send SETUP request */
+ request = create_request (conn, GST_RTSP_SETUP, NULL);
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP/TCP;interleaved=0;mode=record");
+ fail_unless (send_request (conn, request));
+ gst_rtsp_message_free (request);
+ iterate ();
+ response = read_response (conn);
+ gst_rtsp_message_parse_response (response, &status, NULL, NULL);
+ fail_unless_equals_int (status, GST_RTSP_STS_OK);
+
+ rres =
+ gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &session, 0);
+ session = g_strdup (session);
+ fail_unless_equals_int (rres, GST_RTSP_OK);
+ gst_rtsp_message_free (response);
+
+ /* send RECORD */
+ request = create_request (conn, GST_RTSP_RECORD, NULL);
+ gst_rtsp_message_add_header (request, GST_RTSP_HDR_SESSION, session);
+ fail_unless (send_request (conn, request));
+ gst_rtsp_message_free (request);
+ iterate ();
+ response = read_response (conn);
+ gst_rtsp_message_parse_response (response, &status, NULL, NULL);
+ fail_unless_equals_int (status, GST_RTSP_STS_OK);
+ gst_rtsp_message_free (response);
+
+ /* send some data */
+ {
+ GstElement *pipeline, *src, *enc, *pay, *sink;
+
+ pipeline = gst_pipeline_new ("send-pipeline");
+ src = gst_element_factory_make ("audiotestsrc", NULL);
+ g_object_set (src, "num-buffers", RECORD_N_BUFS,
+ "samplesperbuffer", 1000, NULL);
+ enc = gst_element_factory_make ("alawenc", NULL);
+ pay = gst_element_factory_make ("rtppcmapay", NULL);
+ sink = gst_element_factory_make ("appsink", NULL);
+ fail_unless (pipeline && src && enc && pay && sink);
+ gst_bin_add_many (GST_BIN (pipeline), src, enc, pay, sink, NULL);
+ gst_element_link_many (src, enc, pay, sink, NULL);
+ gst_element_set_state (pipeline, GST_STATE_PLAYING);
+
+ do {
+ GstRTSPMessage *data_msg;
+ GstMapInfo map = GST_MAP_INFO_INIT;
+ GstRTSPResult rres;
+ GstSample *sample = NULL;
+ GstBuffer *buf;
+
+ g_signal_emit_by_name (G_OBJECT (sink), "pull-sample", &sample);
+ if (sample == NULL)
+ break;
+ buf = gst_sample_get_buffer (sample);
+ rres = gst_rtsp_message_new_data (&data_msg, 0);
+ fail_unless_equals_int (rres, GST_RTSP_OK);
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ GST_INFO ("sending %u bytes of data on channel 0", (guint) map.size);
+ GST_MEMDUMP ("data on channel 0", map.data, map.size);
+ rres = gst_rtsp_message_set_body (data_msg, map.data, map.size);
+ fail_unless_equals_int (rres, GST_RTSP_OK);
+ gst_buffer_unmap (buf, &map);
+ rres = gst_rtsp_connection_send (conn, data_msg, NULL);
+ fail_unless_equals_int (rres, GST_RTSP_OK);
+ gst_rtsp_message_free (data_msg);
+ gst_sample_unref (sample);
+ } while (TRUE);
+
+ gst_element_set_state (pipeline, GST_STATE_NULL);
+ gst_object_unref (pipeline);
+ }
+
+ /* 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 (sink), "pull-sample", &sample);
+ GST_INFO ("%2d recv sample: %p", i, sample);
+ gst_sample_unref (sample);
+ }
+
+ fail_unless_equals_int (GST_STATE (sink), GST_STATE_PLAYING);
+
+ /* clean up and iterate so the clean-up can finish */
+ gst_rtsp_connection_free (conn);
+ stop_server ();
+ iterate ();
+ g_free (session);
+}
+
+GST_END_TEST;
static Suite *
rtspserver_suite (void)
tcase_add_test (tc, test_connect);
tcase_add_test (tc, test_describe);
tcase_add_test (tc, test_describe_non_existing_mount_point);
+ tcase_add_test (tc, test_describe_record_media);
tcase_add_test (tc, test_setup);
tcase_add_test (tc, test_setup_tcp);
tcase_add_test (tc, test_setup_twice);
tcase_add_test (tc, test_play_disconnect);
tcase_add_test (tc, test_play_specific_server_port);
tcase_add_test (tc, test_play_smpte_range);
+ tcase_add_test (tc, test_announce_without_sdp);
+ tcase_add_test (tc, test_record_tcp);
return s;
}
projects / platform / upstream / gstreamer.git / commitdiff