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tests: add some unit tests for ANNOUNCE and RECORD
https://bugzilla.gnome.org/show_bug.cgi?id=743175
This commit is contained in:
Tim-Philipp Müller
Sebastian Dröge
parent
e9ce91634c
commit
6e5b156b0d
1 changed files with 345 additions and 10 deletions
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@ -1,9 +1,8 @@
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/* GStreamer
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*
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* unit test for GstRTSPServer
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*
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/* GStreamer unit test for GstRTSPServer
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* Copyright (C) 2012 Axis Communications <dev-gstreamer at axis dot com>
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* @author David Svensson Fors <davidsf at axis dot com>
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* @author David Svensson Fors <davidsf at axis dot com>
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* Copyright (C) 2015 Centricular Ltd
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* @author Tim-Philipp Müller <tim@centricular.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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@ -182,6 +181,40 @@ start_server (void)
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GST_DEBUG ("rtsp server listening on port %d", test_port);
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}
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/* start the testing rtsp server for RECORD mode */
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static GstRTSPMediaFactory *
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start_record_server (const gchar * launch_line)
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{
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GstRTSPMediaFactory *factory;
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GstRTSPMountPoints *mounts;
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gchar *service;
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mounts = gst_rtsp_server_get_mount_points (server);
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factory = gst_rtsp_media_factory_new ();
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gst_rtsp_media_factory_set_record (factory, TRUE);
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gst_rtsp_media_factory_set_launch (factory, launch_line);
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gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
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g_object_unref (mounts);
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/* set port to any */
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gst_rtsp_server_set_service (server, "0");
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/* attach to default main context */
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source_id = gst_rtsp_server_attach (server, NULL);
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fail_if (source_id == 0);
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/* get port */
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service = gst_rtsp_server_get_service (server);
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test_port = atoi (service);
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fail_unless (test_port != 0);
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g_free (service);
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GST_DEBUG ("rtsp server listening on port %d", test_port);
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return factory;
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}
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/* stop the tested rtsp server */
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static void
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stop_server (void)
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@ -453,12 +486,11 @@ do_setup_full (GstRTSPConnection * conn, const gchar * control,
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}
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if (use_tcp_transport) {
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transport_string_in =
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g_strdup_printf (TEST_PROTO_TCP ";unicast");
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transport_string_in = g_strdup_printf (TEST_PROTO_TCP ";unicast");
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} else {
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transport_string_in =
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g_strdup_printf (TEST_PROTO ";unicast;client_port=%d-%d",
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client_ports->min, client_ports->max);
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g_strdup_printf (TEST_PROTO ";unicast;client_port=%d-%d",
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client_ports->min, client_ports->max);
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}
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code =
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do_request_full (conn, GST_RTSP_SETUP, control, session_in,
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@ -596,6 +628,27 @@ GST_START_TEST (test_describe)
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GST_END_TEST;
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GST_START_TEST (test_describe_record_media)
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{
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GstRTSPConnection *conn;
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start_record_server ("( fakesink name=depay0 )");
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conn = connect_to_server (test_port, TEST_MOUNT_POINT);
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/* send DESCRIBE request */
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fail_unless_equals_int (do_request (conn, GST_RTSP_DESCRIBE, NULL, NULL, NULL,
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NULL, NULL, NULL, NULL, NULL, NULL, NULL),
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GST_RTSP_STS_METHOD_NOT_ALLOWED);
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/* clean up and iterate so the clean-up can finish */
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gst_rtsp_connection_free (conn);
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stop_server ();
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iterate ();
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}
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GST_END_TEST;
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GST_START_TEST (test_describe_non_existing_mount_point)
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{
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GstRTSPConnection *conn;
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@ -797,7 +850,7 @@ GST_START_TEST (test_setup_twice)
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/* session can not be the same */
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fail_unless (strcmp (session1, session2));
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/* send TEARDOWN request for the first session*/
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/* send TEARDOWN request for the first session */
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fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
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session1) == GST_RTSP_STS_OK);
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@ -1576,6 +1629,285 @@ GST_START_TEST (test_play_smpte_range)
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GST_END_TEST;
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GST_START_TEST (test_announce_without_sdp)
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{
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GstRTSPConnection *conn;
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GstRTSPStatusCode status;
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GstRTSPMessage *request;
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GstRTSPMessage *response;
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start_record_server ("( fakesink name=depay0 )");
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conn = connect_to_server (test_port, TEST_MOUNT_POINT);
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/* create and send ANNOUNCE request */
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request = create_request (conn, GST_RTSP_ANNOUNCE, NULL);
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fail_unless (send_request (conn, request));
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iterate ();
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response = read_response (conn);
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/* check response */
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gst_rtsp_message_parse_response (response, &status, NULL, NULL);
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fail_unless_equals_int (status, GST_RTSP_STS_BAD_REQUEST);
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gst_rtsp_message_free (response);
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/* try again, this type with content-type, but still no SDP */
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gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
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"application/sdp");
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fail_unless (send_request (conn, request));
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iterate ();
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response = read_response (conn);
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/* check response */
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gst_rtsp_message_parse_response (response, &status, NULL, NULL);
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fail_unless_equals_int (status, GST_RTSP_STS_BAD_REQUEST);
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gst_rtsp_message_free (response);
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/* try again, this type with an unknown content-type */
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gst_rtsp_message_remove_header (request, GST_RTSP_HDR_CONTENT_TYPE, -1);
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gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
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"application/x-something");
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fail_unless (send_request (conn, request));
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iterate ();
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response = read_response (conn);
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/* check response */
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gst_rtsp_message_parse_response (response, &status, NULL, NULL);
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fail_unless_equals_int (status, GST_RTSP_STS_BAD_REQUEST);
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gst_rtsp_message_free (response);
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/* clean up and iterate so the clean-up can finish */
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gst_rtsp_message_free (request);
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gst_rtsp_connection_free (conn);
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stop_server ();
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iterate ();
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}
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GST_END_TEST;
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static GstRTSPStatusCode
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do_announce (GstRTSPConnection * conn, GstSDPMessage * sdp)
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{
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GstRTSPMessage *request;
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GstRTSPMessage *response;
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GstRTSPStatusCode code;
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gchar *str;
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/* create request */
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request = create_request (conn, GST_RTSP_ANNOUNCE, NULL);
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gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
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"application/sdp");
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/* add SDP to the response body */
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str = gst_sdp_message_as_text (sdp);
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gst_rtsp_message_take_body (request, (guint8 *) str, strlen (str));
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gst_sdp_message_free (sdp);
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/* send request */
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fail_unless (send_request (conn, request));
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gst_rtsp_message_free (request);
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iterate ();
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/* read response */
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response = read_response (conn);
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/* check status line */
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gst_rtsp_message_parse_response (response, &code, NULL, NULL);
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gst_rtsp_message_free (response);
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return code;
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}
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static void
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media_constructed_cb (GstRTSPMediaFactory * mfactory, GstRTSPMedia * media,
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gpointer user_data)
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{
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GstElement **p_sink = user_data;
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GstElement *bin;
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bin = gst_rtsp_media_get_element (media);
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*p_sink = gst_bin_get_by_name (GST_BIN (bin), "sink");
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GST_INFO ("media constructed!: %" GST_PTR_FORMAT, *p_sink);
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}
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#define RECORD_N_BUFS 10
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GST_START_TEST (test_record_tcp)
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{
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GstRTSPMediaFactory *mfactory;
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GstRTSPConnection *conn;
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GstRTSPStatusCode status;
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GstRTSPMessage *response;
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GstRTSPMessage *request;
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GstSDPMessage *sdp;
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GstRTSPResult rres;
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GSocketAddress *sa;
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GInetAddress *ia;
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GstElement *sink = NULL;
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GSocket *conn_socket;
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const gchar *proto;
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gchar *client_ip, *sess_id, *session = NULL;
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gint i;
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mfactory =
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start_record_server ("( rtppcmadepay name=depay0 ! appsink name=sink )");
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g_signal_connect (mfactory, "media-constructed",
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G_CALLBACK (media_constructed_cb), &sink);
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conn = connect_to_server (test_port, TEST_MOUNT_POINT);
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conn_socket = gst_rtsp_connection_get_read_socket (conn);
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sa = g_socket_get_local_address (conn_socket, NULL);
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ia = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (sa));
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client_ip = g_inet_address_to_string (ia);
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if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV6)
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proto = "IP6";
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else if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV4)
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proto = "IP4";
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else
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g_assert_not_reached ();
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g_object_unref (sa);
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gst_sdp_message_new (&sdp);
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/* some standard things first */
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gst_sdp_message_set_version (sdp, "0");
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/* session ID doesn't have to be super-unique in this case */
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sess_id = g_strdup_printf ("%u", g_random_int ());
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gst_sdp_message_set_origin (sdp, "-", sess_id, "1", "IN", proto, client_ip);
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g_free (sess_id);
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g_free (client_ip);
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gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
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gst_sdp_message_set_information (sdp, "rtsp-server-test");
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gst_sdp_message_add_time (sdp, "0", "0", NULL);
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gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
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/* add stream 0 */
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{
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GstSDPMedia *smedia;
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gst_sdp_media_new (&smedia);
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gst_sdp_media_set_media (smedia, "audio");
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gst_sdp_media_add_format (smedia, "8"); /* pcma/alaw */
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gst_sdp_media_set_port_info (smedia, 0, 1);
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gst_sdp_media_set_proto (smedia, "RTP/AVP");
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gst_sdp_media_add_attribute (smedia, "rtpmap", "8 PCMA/8000");
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gst_sdp_message_add_media (sdp, smedia);
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gst_sdp_media_free (smedia);
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}
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/* send ANNOUNCE request */
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status = do_announce (conn, sdp);
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fail_unless_equals_int (status, GST_RTSP_STS_OK);
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/* create and send SETUP request */
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request = create_request (conn, GST_RTSP_SETUP, NULL);
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gst_rtsp_message_add_header (request, GST_RTSP_HDR_TRANSPORT,
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"RTP/AVP/TCP;interleaved=0;mode=record");
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fail_unless (send_request (conn, request));
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gst_rtsp_message_free (request);
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iterate ();
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response = read_response (conn);
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gst_rtsp_message_parse_response (response, &status, NULL, NULL);
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fail_unless_equals_int (status, GST_RTSP_STS_OK);
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rres =
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gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &session, 0);
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session = g_strdup (session);
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fail_unless_equals_int (rres, GST_RTSP_OK);
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gst_rtsp_message_free (response);
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/* send RECORD */
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request = create_request (conn, GST_RTSP_RECORD, NULL);
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gst_rtsp_message_add_header (request, GST_RTSP_HDR_SESSION, session);
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fail_unless (send_request (conn, request));
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gst_rtsp_message_free (request);
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iterate ();
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response = read_response (conn);
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gst_rtsp_message_parse_response (response, &status, NULL, NULL);
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fail_unless_equals_int (status, GST_RTSP_STS_OK);
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gst_rtsp_message_free (response);
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/* send some data */
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{
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GstElement *pipeline, *src, *enc, *pay, *sink;
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pipeline = gst_pipeline_new ("send-pipeline");
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src = gst_element_factory_make ("audiotestsrc", NULL);
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g_object_set (src, "num-buffers", RECORD_N_BUFS,
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"samplesperbuffer", 1000, NULL);
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enc = gst_element_factory_make ("alawenc", NULL);
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pay = gst_element_factory_make ("rtppcmapay", NULL);
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sink = gst_element_factory_make ("appsink", NULL);
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fail_unless (pipeline && src && enc && pay && sink);
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gst_bin_add_many (GST_BIN (pipeline), src, enc, pay, sink, NULL);
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gst_element_link_many (src, enc, pay, sink, NULL);
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gst_element_set_state (pipeline, GST_STATE_PLAYING);
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do {
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GstRTSPMessage *data_msg;
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GstMapInfo map = GST_MAP_INFO_INIT;
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GstRTSPResult rres;
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GstSample *sample = NULL;
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GstBuffer *buf;
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g_signal_emit_by_name (G_OBJECT (sink), "pull-sample", &sample);
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if (sample == NULL)
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break;
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buf = gst_sample_get_buffer (sample);
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rres = gst_rtsp_message_new_data (&data_msg, 0);
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fail_unless_equals_int (rres, GST_RTSP_OK);
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gst_buffer_map (buf, &map, GST_MAP_READ);
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GST_INFO ("sending %u bytes of data on channel 0", (guint) map.size);
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GST_MEMDUMP ("data on channel 0", map.data, map.size);
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rres = gst_rtsp_message_set_body (data_msg, map.data, map.size);
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fail_unless_equals_int (rres, GST_RTSP_OK);
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gst_buffer_unmap (buf, &map);
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rres = gst_rtsp_connection_send (conn, data_msg, NULL);
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fail_unless_equals_int (rres, GST_RTSP_OK);
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gst_rtsp_message_free (data_msg);
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gst_sample_unref (sample);
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} while (TRUE);
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gst_element_set_state (pipeline, GST_STATE_NULL);
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gst_object_unref (pipeline);
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}
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/* check received data (we assume every buffer created by audiotestsrc and
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* subsequently encoded by mulawenc results in exactly one RTP packet) */
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for (i = 0; i < RECORD_N_BUFS; ++i) {
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GstSample *sample = NULL;
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g_signal_emit_by_name (G_OBJECT (sink), "pull-sample", &sample);
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GST_INFO ("%2d recv sample: %p", i, sample);
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gst_sample_unref (sample);
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}
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fail_unless_equals_int (GST_STATE (sink), GST_STATE_PLAYING);
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/* clean up and iterate so the clean-up can finish */
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gst_rtsp_connection_free (conn);
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stop_server ();
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iterate ();
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g_free (session);
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}
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GST_END_TEST;
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static Suite *
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rtspserver_suite (void)
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@ -1589,6 +1921,7 @@ rtspserver_suite (void)
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tcase_add_test (tc, test_connect);
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tcase_add_test (tc, test_describe);
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tcase_add_test (tc, test_describe_non_existing_mount_point);
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tcase_add_test (tc, test_describe_record_media);
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tcase_add_test (tc, test_setup);
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tcase_add_test (tc, test_setup_tcp);
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tcase_add_test (tc, test_setup_twice);
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@ -1604,6 +1937,8 @@ rtspserver_suite (void)
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tcase_add_test (tc, test_play_disconnect);
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tcase_add_test (tc, test_play_specific_server_port);
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tcase_add_test (tc, test_play_smpte_range);
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tcase_add_test (tc, test_announce_without_sdp);
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tcase_add_test (tc, test_record_tcp);
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return s;
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}
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