/* GStreamer * Copyright (C) 2012 Wim Taymans * * 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 #define VIDEO_PIPELINE "videotestsrc ! " \ "video/x-raw,width=352,height=288 ! " \ "rtpgstpay name=pay0 pt=96" #define AUDIO_PIPELINE "audiotestsrc ! " \ "audio/x-raw,rate=8000 ! " \ "rtpgstpay name=pay1 pt=97" static gchar *session_id; static gint cseq; static guint expected_session_timeout = 60; static const gchar *expected_unsupported_header; static const gchar *expected_scale_header; static const gchar *expected_speed_header; static gdouble fake_rate_value = 0; static gdouble fake_applied_rate_value = 0; static gboolean test_response_200 (GstRTSPClient * client, GstRTSPMessage * response, gboolean close, gpointer user_data) { GstRTSPStatusCode code; const gchar *reason; GstRTSPVersion version; fail_unless (gst_rtsp_message_get_type (response) == GST_RTSP_MESSAGE_RESPONSE); fail_unless (gst_rtsp_message_parse_response (response, &code, &reason, &version) == GST_RTSP_OK); fail_unless (code == GST_RTSP_STS_OK); fail_unless (g_str_equal (reason, "OK")); fail_unless (version == GST_RTSP_VERSION_1_0); return TRUE; } static gboolean test_response_play_200 (GstRTSPClient * client, GstRTSPMessage * response, gboolean close, gpointer user_data) { GstRTSPStatusCode code; const gchar *reason; GstRTSPVersion version; gchar *str; gchar **session_hdr_params; gchar *pattern; fail_unless_equals_int (gst_rtsp_message_get_type (response), GST_RTSP_MESSAGE_RESPONSE); fail_unless (gst_rtsp_message_parse_response (response, &code, &reason, &version) == GST_RTSP_OK); fail_unless_equals_int (code, GST_RTSP_STS_OK); fail_unless_equals_string (reason, "OK"); fail_unless_equals_int (version, GST_RTSP_VERSION_1_0); /* Verify mandatory headers according to RFC 2326 */ /* verify mandatory CSeq header */ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str, 0) == GST_RTSP_OK); fail_unless (atoi (str) == cseq++); /* verify mandatory Session header */ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &str, 0) == GST_RTSP_OK); session_hdr_params = g_strsplit (str, ";", -1); fail_unless (session_hdr_params[0] != NULL); g_strfreev (session_hdr_params); /* verify mandatory RTP-Info header */ fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_RTP_INFO, &str, 0) == GST_RTSP_OK); pattern = g_strdup_printf ("^url=rtsp://.+;seq=[0-9]+;rtptime=[0-9]+"); fail_unless (g_regex_match_simple (pattern, str, 0, 0), "GST_RTSP_HDR_RTP_INFO '%s' doesn't match pattern '%s'", str, pattern); g_free (pattern); return TRUE; } static gboolean test_response_400 (GstRTSPClient * client, GstRTSPMessage * response, gboolean close, gpointer user_data) { GstRTSPStatusCode code; const gchar *reason; GstRTSPVersion version; fail_unless (gst_rtsp_message_get_type (response) == GST_RTSP_MESSAGE_RESPONSE); fail_unless (gst_rtsp_message_parse_response (response, &code, &reason, &version) == GST_RTSP_OK); fail_unless (code == GST_RTSP_STS_BAD_REQUEST); fail_unless (g_str_equal (reason, "Bad Request")); fail_unless (version == GST_RTSP_VERSION_1_0); return TRUE; } static gboolean test_response_404 (GstRTSPClient * client, GstRTSPMessage * response, gboolean close, gpointer user_data) { GstRTSPStatusCode code; const gchar *reason; GstRTSPVersion version; fail_unless (gst_rtsp_message_get_type (response) == GST_RTSP_MESSAGE_RESPONSE); fail_unless (gst_rtsp_message_parse_response (response, &code, &reason, &version) == GST_RTSP_OK); fail_unless (code == GST_RTSP_STS_NOT_FOUND); fail_unless (g_str_equal (reason, "Not Found")); fail_unless (version == GST_RTSP_VERSION_1_0); return TRUE; } static gboolean test_response_454 (GstRTSPClient * client, GstRTSPMessage * response, gboolean close, gpointer user_data) { GstRTSPStatusCode code; const gchar *reason; GstRTSPVersion version; fail_unless (gst_rtsp_message_get_type (response) == GST_RTSP_MESSAGE_RESPONSE); fail_unless (gst_rtsp_message_parse_response (response, &code, &reason, &version) == GST_RTSP_OK); fail_unless (code == GST_RTSP_STS_SESSION_NOT_FOUND); fail_unless (g_str_equal (reason, "Session Not Found")); fail_unless (version == GST_RTSP_VERSION_1_0); return TRUE; } static gboolean test_response_551 (GstRTSPClient * client, GstRTSPMessage * response, gboolean close, gpointer user_data) { GstRTSPStatusCode code; const gchar *reason; GstRTSPVersion version; gchar *options; fail_unless (gst_rtsp_message_get_type (response) == GST_RTSP_MESSAGE_RESPONSE); fail_unless (gst_rtsp_message_parse_response (response, &code, &reason, &version) == GST_RTSP_OK); fail_unless (code == GST_RTSP_STS_OPTION_NOT_SUPPORTED); fail_unless (g_str_equal (reason, "Option not supported")); fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_UNSUPPORTED, &options, 0) == GST_RTSP_OK); fail_unless (!g_strcmp0 (expected_unsupported_header, options)); fail_unless (version == GST_RTSP_VERSION_1_0); return TRUE; } static void create_connection (GstRTSPConnection ** conn) { GSocket *sock; GError *error = NULL; sock = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_STREAM, G_SOCKET_PROTOCOL_TCP, &error); g_assert_no_error (error); fail_unless (gst_rtsp_connection_create_from_socket (sock, "127.0.0.1", 444, NULL, conn) == GST_RTSP_OK); g_object_unref (sock); } static GstRTSPClient * setup_client (const gchar * launch_line, const gchar * mount_point, gboolean enable_rtcp) { GstRTSPClient *client; GstRTSPSessionPool *session_pool; GstRTSPMountPoints *mount_points; GstRTSPMediaFactory *factory; GstRTSPThreadPool *thread_pool; client = gst_rtsp_client_new (); session_pool = gst_rtsp_session_pool_new (); gst_rtsp_client_set_session_pool (client, session_pool); mount_points = gst_rtsp_mount_points_new (); factory = gst_rtsp_media_factory_new (); if (launch_line == NULL) gst_rtsp_media_factory_set_launch (factory, "( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )"); else gst_rtsp_media_factory_set_launch (factory, launch_line); gst_rtsp_media_factory_set_enable_rtcp (factory, enable_rtcp); gst_rtsp_mount_points_add_factory (mount_points, mount_point, factory); gst_rtsp_client_set_mount_points (client, mount_points); thread_pool = gst_rtsp_thread_pool_new (); gst_rtsp_client_set_thread_pool (client, thread_pool); g_object_unref (mount_points); g_object_unref (session_pool); g_object_unref (thread_pool); return client; } static void teardown_client (GstRTSPClient * client) { gst_rtsp_client_set_thread_pool (client, NULL); g_object_unref (client); } static gchar * check_requirements_cb (GstRTSPClient * client, GstRTSPContext * ctx, gchar ** req, gpointer user_data) { int index = 0; GString *result = g_string_new (""); while (req[index] != NULL) { if (g_strcmp0 (req[index], "test-requirements")) { if (result->len > 0) g_string_append (result, ", "); g_string_append (result, req[index]); } index++; } return g_string_free (result, FALSE); } GST_START_TEST (test_require) { GstRTSPClient *client; GstRTSPMessage request = { 0, }; gchar *str; client = gst_rtsp_client_new (); /* require header without handler */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS, "rtsp://localhost/test") == GST_RTSP_OK); str = g_strdup_printf ("test-not-supported1"); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str); g_free (str); expected_unsupported_header = "test-not-supported1"; gst_rtsp_client_set_send_func (client, test_response_551, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); g_signal_connect (G_OBJECT (client), "check-requirements", G_CALLBACK (check_requirements_cb), NULL); /* one supported option */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS, "rtsp://localhost/test") == GST_RTSP_OK); str = g_strdup_printf ("test-requirements"); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str); g_free (str); gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); /* unsupported option */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS, "rtsp://localhost/test") == GST_RTSP_OK); str = g_strdup_printf ("test-not-supported1"); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str); g_free (str); expected_unsupported_header = "test-not-supported1"; gst_rtsp_client_set_send_func (client, test_response_551, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); /* more than one unsupported options */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS, "rtsp://localhost/test") == GST_RTSP_OK); str = g_strdup_printf ("test-not-supported1"); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str); g_free (str); str = g_strdup_printf ("test-not-supported2"); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str); g_free (str); expected_unsupported_header = "test-not-supported1, test-not-supported2"; gst_rtsp_client_set_send_func (client, test_response_551, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); /* supported and unsupported together */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS, "rtsp://localhost/test") == GST_RTSP_OK); str = g_strdup_printf ("test-not-supported1"); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str); g_free (str); str = g_strdup_printf ("test-requirements"); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str); g_free (str); str = g_strdup_printf ("test-not-supported2"); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str); g_free (str); expected_unsupported_header = "test-not-supported1, test-not-supported2"; gst_rtsp_client_set_send_func (client, test_response_551, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); g_object_unref (client); } GST_END_TEST; GST_START_TEST (test_request) { GstRTSPClient *client; GstRTSPMessage request = { 0, }; gchar *str; GstRTSPConnection *conn; client = gst_rtsp_client_new (); /* OPTIONS with invalid url */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS, "foopy://padoop/") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str); g_free (str); gst_rtsp_client_set_send_func (client, test_response_400, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); /* OPTIONS with unknown session id */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS, "rtsp://localhost/test") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str); g_free (str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, "foobar"); gst_rtsp_client_set_send_func (client, test_response_454, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); /* OPTIONS with an absolute path instead of an absolute url */ /* set host information */ create_connection (&conn); fail_unless (gst_rtsp_client_set_connection (client, conn)); fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS, "/test") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str); g_free (str); gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); /* OPTIONS with an absolute path instead of an absolute url with invalid * host information */ g_object_unref (client); client = gst_rtsp_client_new (); fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS, "/test") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str); g_free (str); gst_rtsp_client_set_send_func (client, test_response_400, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); g_object_unref (client); } GST_END_TEST; static gboolean test_option_response_200 (GstRTSPClient * client, GstRTSPMessage * response, gboolean close, gpointer user_data) { GstRTSPStatusCode code; const gchar *reason; GstRTSPVersion version; gchar *str; GstRTSPMethod methods; fail_unless (gst_rtsp_message_get_type (response) == GST_RTSP_MESSAGE_RESPONSE); fail_unless (gst_rtsp_message_parse_response (response, &code, &reason, &version) == GST_RTSP_OK); fail_unless (code == GST_RTSP_STS_OK); fail_unless (g_str_equal (reason, "OK")); fail_unless (version == GST_RTSP_VERSION_1_0); fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str, 0) == GST_RTSP_OK); fail_unless (atoi (str) == cseq++); fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_PUBLIC, &str, 0) == GST_RTSP_OK); methods = gst_rtsp_options_from_text (str); fail_if (methods == 0); fail_unless (methods == (GST_RTSP_DESCRIBE | GST_RTSP_ANNOUNCE | GST_RTSP_OPTIONS | GST_RTSP_PAUSE | GST_RTSP_PLAY | GST_RTSP_RECORD | GST_RTSP_SETUP | GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN)); return TRUE; } GST_START_TEST (test_options) { GstRTSPClient *client; GstRTSPMessage request = { 0, }; gchar *str; client = gst_rtsp_client_new (); /* simple OPTIONS */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS, "rtsp://localhost/test") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str); g_free (str); gst_rtsp_client_set_send_func (client, test_option_response_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); g_object_unref (client); } GST_END_TEST; static void test_describe_sub (const gchar * mount_point, const gchar * url) { GstRTSPClient *client; GstRTSPMessage request = { 0, }; gchar *str; client = gst_rtsp_client_new (); /* simple DESCRIBE for non-existing url */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE, url) == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str); g_free (str); gst_rtsp_client_set_send_func (client, test_response_404, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); g_object_unref (client); /* simple DESCRIBE for an existing url */ client = setup_client (NULL, mount_point, TRUE); fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE, url) == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str); g_free (str); gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); teardown_client (client); } GST_START_TEST (test_describe) { test_describe_sub ("/test", "rtsp://localhost/test"); } GST_END_TEST; GST_START_TEST (test_describe_root_mount_point) { test_describe_sub ("/", "rtsp://localhost"); } GST_END_TEST; static const gchar *expected_transport = NULL; static gboolean test_setup_response_200 (GstRTSPClient * client, GstRTSPMessage * response, gboolean close, gpointer user_data) { GstRTSPStatusCode code; const gchar *reason; GstRTSPVersion version; gchar *str; gchar *pattern; GstRTSPSessionPool *session_pool; GstRTSPSession *session; gchar **session_hdr_params; fail_unless (expected_transport != NULL); fail_unless_equals_int (gst_rtsp_message_get_type (response), GST_RTSP_MESSAGE_RESPONSE); fail_unless (gst_rtsp_message_parse_response (response, &code, &reason, &version) == GST_RTSP_OK); fail_unless_equals_int (code, GST_RTSP_STS_OK); fail_unless_equals_string (reason, "OK"); fail_unless_equals_int (version, GST_RTSP_VERSION_1_0); fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str, 0) == GST_RTSP_OK); fail_unless (atoi (str) == cseq++); fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_TRANSPORT, &str, 0) == GST_RTSP_OK); pattern = g_strdup_printf ("^%s$", expected_transport); fail_unless (g_regex_match_simple (pattern, str, 0, 0), "Transport '%s' doesn't match pattern '%s'", str, pattern); g_free (pattern); fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &str, 0) == GST_RTSP_OK); session_hdr_params = g_strsplit (str, ";", -1); /* session-id value */ fail_unless (session_hdr_params[0] != NULL); if (expected_session_timeout != 60) { /* session timeout param */ gchar *timeout_str = g_strdup_printf ("timeout=%u", expected_session_timeout); fail_unless (session_hdr_params[1] != NULL); g_strstrip (session_hdr_params[1]); fail_unless (g_strcmp0 (session_hdr_params[1], timeout_str) == 0); g_free (timeout_str); } session_pool = gst_rtsp_client_get_session_pool (client); fail_unless (session_pool != NULL); session = gst_rtsp_session_pool_find (session_pool, session_hdr_params[0]); g_strfreev (session_hdr_params); /* remember session id to be able to send teardown */ if (session_id) g_free (session_id); session_id = g_strdup (gst_rtsp_session_get_sessionid (session)); fail_unless (session_id != NULL); fail_unless (session != NULL); g_object_unref (session); g_object_unref (session_pool); return TRUE; } static gboolean test_setup_response_461 (GstRTSPClient * client, GstRTSPMessage * response, gboolean close, gpointer user_data) { GstRTSPStatusCode code; const gchar *reason; GstRTSPVersion version; gchar *str; fail_unless (expected_transport == NULL); fail_unless (gst_rtsp_message_get_type (response) == GST_RTSP_MESSAGE_RESPONSE); fail_unless (gst_rtsp_message_parse_response (response, &code, &reason, &version) == GST_RTSP_OK); fail_unless (code == GST_RTSP_STS_UNSUPPORTED_TRANSPORT); fail_unless (g_str_equal (reason, "Unsupported transport")); fail_unless (version == GST_RTSP_VERSION_1_0); fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str, 0) == GST_RTSP_OK); fail_unless (atoi (str) == cseq++); return TRUE; } static gboolean test_teardown_response_200 (GstRTSPClient * client, GstRTSPMessage * response, gboolean close, gpointer user_data) { GstRTSPStatusCode code; const gchar *reason; GstRTSPVersion version; fail_unless (gst_rtsp_message_get_type (response) == GST_RTSP_MESSAGE_RESPONSE); fail_unless (gst_rtsp_message_parse_response (response, &code, &reason, &version) == GST_RTSP_OK); fail_unless (code == GST_RTSP_STS_OK); fail_unless (g_str_equal (reason, "OK")); fail_unless (version == GST_RTSP_VERSION_1_0); return TRUE; } static void send_teardown (GstRTSPClient * client, const gchar * url) { GstRTSPMessage request = { 0, }; gchar *str; fail_unless (session_id != NULL); fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, url) == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id); gst_rtsp_client_set_send_func (client, test_teardown_response_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); g_free (session_id); session_id = NULL; } static void test_setup_tcp_sub (const gchar * mount_point, const gchar * url1, const gchar * url2) { GstRTSPClient *client; GstRTSPConnection *conn; GstRTSPMessage request = { 0, }; gchar *str; client = setup_client (NULL, mount_point, TRUE); create_connection (&conn); fail_unless (gst_rtsp_client_set_connection (client, conn)); fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP, url1) == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str); g_free (str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, "RTP/AVP/TCP;unicast"); gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL); expected_transport = "RTP/AVP/TCP;unicast;interleaved=0-1;ssrc=.*;mode=\"PLAY\""; fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); send_teardown (client, url2); teardown_client (client); } GST_START_TEST (test_setup_tcp) { test_setup_tcp_sub ("/test", "rtsp://localhost/test/stream=0", "rtsp://localhost/test"); } GST_END_TEST; GST_START_TEST (test_setup_tcp_root_mount_point) { test_setup_tcp_sub ("/", "rtsp://localhost/stream=0", "rtsp://localhost"); } GST_END_TEST; GST_START_TEST (test_setup_no_rtcp) { GstRTSPClient *client; GstRTSPConnection *conn; GstRTSPMessage request = { 0, }; gchar *str; client = setup_client (NULL, "/test", FALSE); create_connection (&conn); fail_unless (gst_rtsp_client_set_connection (client, conn)); fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP, "rtsp://localhost/test/stream=0") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str); g_free (str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, "RTP/AVP;unicast;client_port=5000-5001"); gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL); /* We want to verify that server_port holds a single number, not a range */ expected_transport = "RTP/AVP;unicast;client_port=5000-5001;server_port=[0-9]+;ssrc=.*;mode=\"PLAY\""; fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); send_teardown (client, "rtsp://localhost/test"); teardown_client (client); } GST_END_TEST; static void test_setup_tcp_two_streams_same_channels_sub (const gchar * mount_point, const gchar * url1, const gchar * url2, const gchar * url3) { GstRTSPClient *client; GstRTSPConnection *conn; GstRTSPMessage request = { 0, }; gchar *str; client = setup_client (NULL, mount_point, TRUE); create_connection (&conn); fail_unless (gst_rtsp_client_set_connection (client, conn)); /* test SETUP of a video stream with 0-1 as interleaved channels */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP, url1) == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str); g_free (str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, "RTP/AVP/TCP;unicast;interleaved=0-1"); gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL); expected_transport = "RTP/AVP/TCP;unicast;interleaved=0-1;ssrc=.*;mode=\"PLAY\""; fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); /* test SETUP of an audio stream with *the same* interleaved channels. * we expect the server to allocate new channel numbers */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP, url2) == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str); g_free (str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, "RTP/AVP/TCP;unicast;interleaved=0-1"); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id); gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL); expected_transport = "RTP/AVP/TCP;unicast;interleaved=2-3;ssrc=.*;mode=\"PLAY\""; fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); send_teardown (client, url3); teardown_client (client); } GST_START_TEST (test_setup_tcp_two_streams_same_channels) { test_setup_tcp_two_streams_same_channels_sub ("/test", "rtsp://localhost/test/stream=0", "rtsp://localhost/test/stream=1", "rtsp://localhost/test"); } GST_END_TEST; GST_START_TEST (test_setup_tcp_two_streams_same_channels_root_mount_point) { test_setup_tcp_two_streams_same_channels_sub ("/", "rtsp://localhost/stream=0", "rtsp://localhost/stream=1", "rtsp://localhost"); } GST_END_TEST; static GstRTSPClient * setup_multicast_client (guint max_ttl, const gchar * mount_point) { GstRTSPClient *client; GstRTSPSessionPool *session_pool; GstRTSPMountPoints *mount_points; GstRTSPMediaFactory *factory; GstRTSPAddressPool *address_pool; GstRTSPThreadPool *thread_pool; client = gst_rtsp_client_new (); session_pool = gst_rtsp_session_pool_new (); gst_rtsp_client_set_session_pool (client, session_pool); mount_points = gst_rtsp_mount_points_new (); factory = gst_rtsp_media_factory_new (); gst_rtsp_media_factory_set_launch (factory, "audiotestsrc ! audio/x-raw,rate=44100 ! audioconvert ! rtpL16pay name=pay0"); address_pool = gst_rtsp_address_pool_new (); fail_unless (gst_rtsp_address_pool_add_range (address_pool, "233.252.0.1", "233.252.0.1", 5000, 5010, 1)); gst_rtsp_media_factory_set_address_pool (factory, address_pool); gst_rtsp_media_factory_add_role (factory, "user", "media.factory.access", G_TYPE_BOOLEAN, TRUE, "media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL); gst_rtsp_mount_points_add_factory (mount_points, mount_point, factory); gst_rtsp_client_set_mount_points (client, mount_points); gst_rtsp_media_factory_set_max_mcast_ttl (factory, max_ttl); thread_pool = gst_rtsp_thread_pool_new (); gst_rtsp_client_set_thread_pool (client, thread_pool); g_object_unref (mount_points); g_object_unref (session_pool); g_object_unref (address_pool); g_object_unref (thread_pool); return client; } GST_START_TEST (test_client_multicast_transport_404) { GstRTSPClient *client; GstRTSPMessage request = { 0, }; gchar *str; client = setup_multicast_client (1, "/test"); /* simple SETUP for non-existing url */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP, "rtsp://localhost/test2/stream=0") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, "RTP/AVP;multicast"); gst_rtsp_client_set_send_func (client, test_response_404, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); teardown_client (client); } GST_END_TEST; static void new_session_cb (GObject * client, GstRTSPSession * session, gpointer user_data) { GST_DEBUG ("%p: new session %p", client, session); gst_rtsp_session_set_timeout (session, expected_session_timeout); } GST_START_TEST (test_client_multicast_transport) { GstRTSPClient *client; GstRTSPMessage request = { 0, }; gchar *str; client = setup_multicast_client (1, "/test"); expected_session_timeout = 20; g_signal_connect (G_OBJECT (client), "new-session", G_CALLBACK (new_session_cb), NULL); /* simple SETUP with a valid URI and multicast */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP, "rtsp://localhost/test/stream=0") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, "RTP/AVP;multicast"); expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;" "ttl=1;port=5000-5001;mode=\"PLAY\""; gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); expected_transport = NULL; expected_session_timeout = 60; send_teardown (client, "rtsp://localhost/test"); teardown_client (client); } GST_END_TEST; GST_START_TEST (test_client_multicast_ignore_transport_specific) { GstRTSPClient *client; GstRTSPMessage request = { 0, }; gchar *str; client = setup_multicast_client (1, "/test"); /* simple SETUP with a valid URI and multicast and a specific dest, * but ignore it */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP, "rtsp://localhost/test/stream=0") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, "RTP/AVP;multicast;destination=233.252.0.2;ttl=2;port=5001-5006;"); expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;" "ttl=1;port=5000-5001;mode=\"PLAY\""; gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); expected_transport = NULL; send_teardown (client, "rtsp://localhost/test"); teardown_client (client); } GST_END_TEST; static void multicast_transport_specific (void) { GstRTSPClient *client; GstRTSPMessage request = { 0, }; gchar *str; GstRTSPSessionPool *session_pool; GstRTSPContext ctx = { NULL }; client = setup_multicast_client (1, "/test"); ctx.client = client; ctx.auth = gst_rtsp_auth_new (); ctx.token = gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS, G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING, "user", NULL); gst_rtsp_context_push_current (&ctx); /* simple SETUP with a valid URI */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP, "rtsp://localhost/test/stream=0") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, expected_transport); gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL); session_pool = gst_rtsp_client_get_session_pool (client); fail_unless (session_pool != NULL); fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 1); g_object_unref (session_pool); /* send PLAY request */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, "rtsp://localhost/test") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id); gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); send_teardown (client, "rtsp://localhost/test"); teardown_client (client); g_object_unref (ctx.auth); gst_rtsp_token_unref (ctx.token); gst_rtsp_context_pop_current (&ctx); } /* CASE: multicast address requested by the client exists in the address pool */ GST_START_TEST (test_client_multicast_transport_specific) { expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;" "ttl=1;port=5000-5001;mode=\"PLAY\""; multicast_transport_specific (); expected_transport = NULL; } GST_END_TEST; /* CASE: multicast address requested by the client does not exist in the address pool */ GST_START_TEST (test_client_multicast_transport_specific_no_address_in_pool) { expected_transport = "RTP/AVP;multicast;destination=234.252.0.3;" "ttl=1;port=10002-10004;mode=\"PLAY\""; multicast_transport_specific (); expected_transport = NULL; } GST_END_TEST; static gboolean test_response_sdp (GstRTSPClient * client, GstRTSPMessage * response, gboolean close, gpointer user_data) { guint8 *data; guint size; GstSDPMessage *sdp_msg; const GstSDPMedia *sdp_media; const GstSDPBandwidth *bw; gint bandwidth_val = GPOINTER_TO_INT (user_data); fail_unless (gst_rtsp_message_get_body (response, &data, &size) == GST_RTSP_OK); gst_sdp_message_new (&sdp_msg); fail_unless (gst_sdp_message_parse_buffer (data, size, sdp_msg) == GST_SDP_OK); /* session description */ /* v= */ fail_unless (gst_sdp_message_get_version (sdp_msg) != NULL); /* o= */ fail_unless (gst_sdp_message_get_origin (sdp_msg) != NULL); /* s= */ fail_unless (gst_sdp_message_get_session_name (sdp_msg) != NULL); /* t=0 0 */ fail_unless (gst_sdp_message_times_len (sdp_msg) == 0); /* verify number of medias */ fail_unless (gst_sdp_message_medias_len (sdp_msg) == 1); /* media description */ sdp_media = gst_sdp_message_get_media (sdp_msg, 0); fail_unless (sdp_media != NULL); /* m= */ fail_unless (gst_sdp_media_get_media (sdp_media) != NULL); /* media bandwidth */ if (bandwidth_val) { fail_unless (gst_sdp_media_bandwidths_len (sdp_media) == 1); bw = gst_sdp_media_get_bandwidth (sdp_media, 0); fail_unless (bw != NULL); fail_unless (g_strcmp0 (bw->bwtype, "AS") == 0); fail_unless (bw->bandwidth == bandwidth_val); } else { fail_unless (gst_sdp_media_bandwidths_len (sdp_media) == 0); } gst_sdp_message_free (sdp_msg); return TRUE; } static void test_client_sdp (const gchar * launch_line, guint * bandwidth_val) { GstRTSPClient *client; GstRTSPMessage request = { 0, }; gchar *str; /* simple DESCRIBE for an existing url */ client = setup_client (launch_line, "/test", TRUE); fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE, "rtsp://localhost/test") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str); g_free (str); gst_rtsp_client_set_send_func (client, test_response_sdp, (gpointer) bandwidth_val, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); teardown_client (client); } GST_START_TEST (test_client_sdp_with_max_bitrate_tag) { test_client_sdp ("videotestsrc " "! taginject tags=\"maximum-bitrate=(uint)50000000\" " "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96", GUINT_TO_POINTER (50000)); /* max-bitrate=0: no bandwidth line */ test_client_sdp ("videotestsrc " "! taginject tags=\"maximum-bitrate=(uint)0\" " "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96", GUINT_TO_POINTER (0)); } GST_END_TEST; GST_START_TEST (test_client_sdp_with_bitrate_tag) { test_client_sdp ("videotestsrc " "! taginject tags=\"bitrate=(uint)7000000\" " "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96", GUINT_TO_POINTER (7000)); /* bitrate=0: no bandwdith line */ test_client_sdp ("videotestsrc " "! taginject tags=\"bitrate=(uint)0\" " "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96", GUINT_TO_POINTER (0)); } GST_END_TEST; GST_START_TEST (test_client_sdp_with_max_bitrate_and_bitrate_tags) { test_client_sdp ("videotestsrc " "! taginject tags=\"bitrate=(uint)7000000,maximum-bitrate=(uint)50000000\" " "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96", GUINT_TO_POINTER (50000)); /* max-bitrate is zero: fallback to bitrate */ test_client_sdp ("videotestsrc " "! taginject tags=\"bitrate=(uint)7000000,maximum-bitrate=(uint)0\" " "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96", GUINT_TO_POINTER (7000)); /* max-bitrate=bitrate=0o: no bandwidth line */ test_client_sdp ("videotestsrc " "! taginject tags=\"bitrate=(uint)0,maximum-bitrate=(uint)0\" " "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96", GUINT_TO_POINTER (0)); } GST_END_TEST; GST_START_TEST (test_client_sdp_with_no_bitrate_tags) { test_client_sdp ("videotestsrc " "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96", NULL); } GST_END_TEST; static void mcast_transport_two_clients (gboolean shared, const gchar * transport1, const gchar * expected_transport1, const gchar * addr1, const gchar * transport2, const gchar * expected_transport2, const gchar * addr2, gboolean bind_mcast_address) { GstRTSPClient *client1, *client2; GstRTSPMessage request = { 0, }; gchar *str; GstRTSPSessionPool *session_pool; GstRTSPContext ctx = { NULL }; GstRTSPContext ctx2 = { NULL }; GstRTSPMountPoints *mount_points; GstRTSPMediaFactory *factory; GstRTSPAddressPool *address_pool; GstRTSPThreadPool *thread_pool; gchar *session_id1; gchar *client_addr = NULL; mount_points = gst_rtsp_mount_points_new (); factory = gst_rtsp_media_factory_new (); if (shared) gst_rtsp_media_factory_set_shared (factory, TRUE); gst_rtsp_media_factory_set_max_mcast_ttl (factory, 5); gst_rtsp_media_factory_set_bind_mcast_address (factory, bind_mcast_address); gst_rtsp_media_factory_set_launch (factory, "audiotestsrc ! audio/x-raw,rate=44100 ! audioconvert ! rtpL16pay name=pay0"); address_pool = gst_rtsp_address_pool_new (); fail_unless (gst_rtsp_address_pool_add_range (address_pool, "233.252.0.1", "233.252.0.1", 5000, 5001, 1)); gst_rtsp_media_factory_set_address_pool (factory, address_pool); gst_rtsp_media_factory_add_role (factory, "user", "media.factory.access", G_TYPE_BOOLEAN, TRUE, "media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL); gst_rtsp_mount_points_add_factory (mount_points, "/test", factory); session_pool = gst_rtsp_session_pool_new (); thread_pool = gst_rtsp_thread_pool_new (); /* first multicast client with transport specific request */ client1 = gst_rtsp_client_new (); gst_rtsp_client_set_session_pool (client1, session_pool); gst_rtsp_client_set_mount_points (client1, mount_points); gst_rtsp_client_set_thread_pool (client1, thread_pool); ctx.client = client1; ctx.auth = gst_rtsp_auth_new (); ctx.token = gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS, G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING, "user", NULL); gst_rtsp_context_push_current (&ctx); expected_transport = expected_transport1; /* send SETUP request */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP, "rtsp://localhost/test/stream=0") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transport1); gst_rtsp_client_set_send_func (client1, test_setup_response_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client1, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); expected_transport = NULL; /* send PLAY request */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, "rtsp://localhost/test") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id); gst_rtsp_client_set_send_func (client1, test_response_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client1, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); /* check address */ client_addr = gst_rtsp_stream_get_multicast_client_addresses (ctx.stream); fail_if (client_addr == NULL); fail_unless (g_str_equal (client_addr, addr1)); g_free (client_addr); gst_rtsp_context_pop_current (&ctx); session_id1 = g_strdup (session_id); /* second multicast client with transport specific request */ cseq = 0; client2 = gst_rtsp_client_new (); gst_rtsp_client_set_session_pool (client2, session_pool); gst_rtsp_client_set_mount_points (client2, mount_points); gst_rtsp_client_set_thread_pool (client2, thread_pool); ctx2.client = client2; ctx2.auth = gst_rtsp_auth_new (); ctx2.token = gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS, G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING, "user", NULL); gst_rtsp_context_push_current (&ctx2); expected_transport = expected_transport2; /* send SETUP request */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP, "rtsp://localhost/test/stream=0") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transport2); gst_rtsp_client_set_send_func (client2, test_setup_response_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client2, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); expected_transport = NULL; /* send PLAY request */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, "rtsp://localhost/test") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id); gst_rtsp_client_set_send_func (client2, test_response_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client2, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); /* check addresses */ client_addr = gst_rtsp_stream_get_multicast_client_addresses (ctx2.stream); fail_if (client_addr == NULL); if (shared) { if (g_str_equal (addr1, addr2)) { fail_unless (g_str_equal (client_addr, addr1)); } else { gchar *addr_str = g_strdup_printf ("%s,%s", addr2, addr1); fail_unless (g_str_equal (client_addr, addr_str)); g_free (addr_str); } } else { fail_unless (g_str_equal (client_addr, addr2)); } g_free (client_addr); send_teardown (client2, "rtsp://localhost/test"); gst_rtsp_context_pop_current (&ctx2); gst_rtsp_context_push_current (&ctx); session_id = session_id1; send_teardown (client1, "rtsp://localhost/test"); gst_rtsp_context_pop_current (&ctx); teardown_client (client1); teardown_client (client2); g_object_unref (ctx.auth); g_object_unref (ctx2.auth); gst_rtsp_token_unref (ctx.token); gst_rtsp_token_unref (ctx2.token); g_object_unref (mount_points); g_object_unref (session_pool); g_object_unref (address_pool); g_object_unref (thread_pool); } /* CASE: media is shared. * client 1: SETUP ---> * client 1: PLAY ---> * client 2: SETUP ---> * client 1: TEARDOWN ---> * client 2: PLAY ---> * client 2: TEARDOWN ---> */ static void mcast_transport_two_clients_teardown_play (const gchar * transport1, const gchar * expected_transport1, const gchar * transport2, const gchar * expected_transport2, gboolean bind_mcast_address, gboolean is_shared) { GstRTSPClient *client1, *client2; GstRTSPMessage request = { 0, }; gchar *str; GstRTSPSessionPool *session_pool; GstRTSPContext ctx = { NULL }; GstRTSPContext ctx2 = { NULL }; GstRTSPMountPoints *mount_points; GstRTSPMediaFactory *factory; GstRTSPAddressPool *address_pool; GstRTSPThreadPool *thread_pool; gchar *session_id1, *session_id2; mount_points = gst_rtsp_mount_points_new (); factory = gst_rtsp_media_factory_new (); gst_rtsp_media_factory_set_shared (factory, is_shared); gst_rtsp_media_factory_set_max_mcast_ttl (factory, 5); gst_rtsp_media_factory_set_bind_mcast_address (factory, bind_mcast_address); gst_rtsp_media_factory_set_launch (factory, "audiotestsrc ! audio/x-raw,rate=44100 ! audioconvert ! rtpL16pay name=pay0"); address_pool = gst_rtsp_address_pool_new (); if (is_shared) fail_unless (gst_rtsp_address_pool_add_range (address_pool, "233.252.0.1", "233.252.0.1", 5000, 5001, 1)); else fail_unless (gst_rtsp_address_pool_add_range (address_pool, "233.252.0.1", "233.252.0.1", 5000, 5003, 1)); gst_rtsp_media_factory_set_address_pool (factory, address_pool); gst_rtsp_media_factory_add_role (factory, "user", "media.factory.access", G_TYPE_BOOLEAN, TRUE, "media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL); gst_rtsp_mount_points_add_factory (mount_points, "/test", factory); session_pool = gst_rtsp_session_pool_new (); thread_pool = gst_rtsp_thread_pool_new (); /* client 1 configuration */ client1 = gst_rtsp_client_new (); gst_rtsp_client_set_session_pool (client1, session_pool); gst_rtsp_client_set_mount_points (client1, mount_points); gst_rtsp_client_set_thread_pool (client1, thread_pool); ctx.client = client1; ctx.auth = gst_rtsp_auth_new (); ctx.token = gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS, G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING, "user", NULL); gst_rtsp_context_push_current (&ctx); expected_transport = expected_transport1; /* client 1 sends SETUP request */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP, "rtsp://localhost/test/stream=0") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transport1); gst_rtsp_client_set_send_func (client1, test_setup_response_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client1, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); expected_transport = NULL; /* client 1 sends PLAY request */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, "rtsp://localhost/test") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id); gst_rtsp_client_set_send_func (client1, test_response_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client1, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); gst_rtsp_context_pop_current (&ctx); session_id1 = g_strdup (session_id); /* client 2 configuration */ cseq = 0; client2 = gst_rtsp_client_new (); gst_rtsp_client_set_session_pool (client2, session_pool); gst_rtsp_client_set_mount_points (client2, mount_points); gst_rtsp_client_set_thread_pool (client2, thread_pool); ctx2.client = client2; ctx2.auth = gst_rtsp_auth_new (); ctx2.token = gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS, G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING, "user", NULL); gst_rtsp_context_push_current (&ctx2); expected_transport = expected_transport2; /* client 2 sends SETUP request */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP, "rtsp://localhost/test/stream=0") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transport2); gst_rtsp_client_set_send_func (client2, test_setup_response_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client2, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); expected_transport = NULL; session_id2 = g_strdup (session_id); g_free (session_id); gst_rtsp_context_pop_current (&ctx2); /* the first client sends TEARDOWN request */ gst_rtsp_context_push_current (&ctx); session_id = session_id1; send_teardown (client1, "rtsp://localhost/test"); gst_rtsp_context_pop_current (&ctx); teardown_client (client1); /* the second client sends PLAY request */ gst_rtsp_context_push_current (&ctx2); session_id = session_id2; fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, "rtsp://localhost/test") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id); gst_rtsp_client_set_send_func (client2, test_response_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client2, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); /* client 2 sends TEARDOWN request */ send_teardown (client2, "rtsp://localhost/test"); gst_rtsp_context_pop_current (&ctx2); teardown_client (client2); g_object_unref (ctx.auth); g_object_unref (ctx2.auth); gst_rtsp_token_unref (ctx.token); gst_rtsp_token_unref (ctx2.token); g_object_unref (mount_points); g_object_unref (session_pool); g_object_unref (address_pool); g_object_unref (thread_pool); } /* test if two multicast clients can choose different transport settings * CASE: media is shared */ GST_START_TEST (test_client_multicast_transport_specific_two_clients_shared_media) { const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;" "ttl=1;port=5000-5001;mode=\"PLAY\""; const gchar *expected_transport_1 = transport_client_1; const gchar *addr_client_1 = "233.252.0.1:5000"; const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;" "ttl=1;port=5002-5003;mode=\"PLAY\""; const gchar *expected_transport_2 = transport_client_2; const gchar *addr_client_2 = "233.252.0.2:5002"; mcast_transport_two_clients (TRUE, transport_client_1, expected_transport_1, addr_client_1, transport_client_2, expected_transport_2, addr_client_2, FALSE); } GST_END_TEST; /* test if two multicast clients can choose different transport settings * CASE: media is not shared */ GST_START_TEST (test_client_multicast_transport_specific_two_clients) { const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;" "ttl=1;port=5000-5001;mode=\"PLAY\""; const gchar *expected_transport_1 = transport_client_1; const gchar *addr_client_1 = "233.252.0.1:5000"; const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;" "ttl=1;port=5002-5003;mode=\"PLAY\""; const gchar *expected_transport_2 = transport_client_2; const gchar *addr_client_2 = "233.252.0.2:5002"; mcast_transport_two_clients (FALSE, transport_client_1, expected_transport_1, addr_client_1, transport_client_2, expected_transport_2, addr_client_2, FALSE); } GST_END_TEST; /* test if two multicast clients can choose the same ports but different * multicast destinations * CASE: media is not shared */ GST_START_TEST (test_client_multicast_transport_specific_two_clients_same_ports) { const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;" "ttl=1;port=9000-9001;mode=\"PLAY\""; const gchar *expected_transport_1 = transport_client_1; const gchar *addr_client_1 = "233.252.0.1:9000"; const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;" "ttl=1;port=9000-9001;mode=\"PLAY\""; const gchar *expected_transport_2 = transport_client_2; const gchar *addr_client_2 = "233.252.0.2:9000"; /* configure the multicast socket to be bound to the requested multicast address instead of INADDR_ANY. * The clients request the same rtp/rtcp borts and having the socket that are bound to ANY would result * in bind() failure */ gboolean allow_bind_mcast_address = TRUE; mcast_transport_two_clients (FALSE, transport_client_1, expected_transport_1, addr_client_1, transport_client_2, expected_transport_2, addr_client_2, allow_bind_mcast_address); } GST_END_TEST; /* test if two multicast clients can choose the same multicast destination but different * ports * CASE: media is not shared */ GST_START_TEST (test_client_multicast_transport_specific_two_clients_same_destination) { const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.2;" "ttl=1;port=9002-9003;mode=\"PLAY\""; const gchar *expected_transport_1 = transport_client_1; const gchar *addr_client_1 = "233.252.0.2:9002"; const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;" "ttl=1;port=9004-9005;mode=\"PLAY\""; const gchar *expected_transport_2 = transport_client_2; const gchar *addr_client_2 = "233.252.0.2:9004"; mcast_transport_two_clients (FALSE, transport_client_1, expected_transport_1, addr_client_1, transport_client_2, expected_transport_2, addr_client_2, FALSE); } GST_END_TEST; /* test if two multicast clients can choose the same transport settings. * CASE: media is shared */ GST_START_TEST (test_client_multicast_transport_specific_two_clients_shared_media_same_transport) { const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;" "ttl=1;port=5000-5001;mode=\"PLAY\""; const gchar *expected_transport_1 = transport_client_1; const gchar *addr_client_1 = "233.252.0.1:5000"; const gchar *transport_client_2 = transport_client_1; const gchar *expected_transport_2 = expected_transport_1; const gchar *addr_client_2 = addr_client_1; mcast_transport_two_clients (TRUE, transport_client_1, expected_transport_1, addr_client_1, transport_client_2, expected_transport_2, addr_client_2, FALSE); } GST_END_TEST; /* test if two multicast clients get the same transport settings without * requesting specific transport. * CASE: media is shared */ GST_START_TEST (test_client_multicast_two_clients_shared_media) { const gchar *transport_client_1 = "RTP/AVP;multicast;mode=\"PLAY\""; const gchar *expected_transport_1 = "RTP/AVP;multicast;destination=233.252.0.1;" "ttl=1;port=5000-5001;mode=\"PLAY\""; const gchar *addr_client_1 = "233.252.0.1:5000"; const gchar *transport_client_2 = transport_client_1; const gchar *expected_transport_2 = expected_transport_1; const gchar *addr_client_2 = addr_client_1; mcast_transport_two_clients (TRUE, transport_client_1, expected_transport_1, addr_client_1, transport_client_2, expected_transport_2, addr_client_2, FALSE); } GST_END_TEST; /* test if it's possible to play the shared media, after one of the clients * has terminated its session. */ GST_START_TEST (test_client_multicast_two_clients_shared_media_teardown_play) { const gchar *transport_client_1 = "RTP/AVP;multicast;mode=\"PLAY\""; const gchar *expected_transport_1 = "RTP/AVP;multicast;destination=233.252.0.1;" "ttl=1;port=5000-5001;mode=\"PLAY\""; const gchar *transport_client_2 = transport_client_1; const gchar *expected_transport_2 = expected_transport_1; mcast_transport_two_clients_teardown_play (transport_client_1, expected_transport_1, transport_client_2, expected_transport_2, FALSE, TRUE); } GST_END_TEST; /* test if it's possible to play the shared media, after one of the clients * has terminated its session. */ GST_START_TEST (test_client_multicast_two_clients_not_shared_media_teardown_play) { const gchar *transport_client_1 = "RTP/AVP;multicast;mode=\"PLAY\""; const gchar *expected_transport_1 = "RTP/AVP;multicast;destination=233.252.0.1;" "ttl=1;port=5000-5001;mode=\"PLAY\""; const gchar *transport_client_2 = transport_client_1; const gchar *expected_transport_2 = "RTP/AVP;multicast;destination=233.252.0.1;" "ttl=1;port=5002-5003;mode=\"PLAY\""; mcast_transport_two_clients_teardown_play (transport_client_1, expected_transport_1, transport_client_2, expected_transport_2, FALSE, FALSE); } GST_END_TEST; /* test if two multicast clients get the different transport settings: the first client * requests the specific transport configuration while the second client lets * the server select the multicast address and the ports. * CASE: media is shared */ GST_START_TEST (test_client_multicast_two_clients_first_specific_transport_shared_media) { const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;" "ttl=1;port=5000-5001;mode=\"PLAY\""; const gchar *expected_transport_1 = transport_client_1; const gchar *addr_client_1 = "233.252.0.1:5000"; const gchar *transport_client_2 = "RTP/AVP;multicast;mode=\"PLAY\""; const gchar *expected_transport_2 = expected_transport_1; const gchar *addr_client_2 = addr_client_1; mcast_transport_two_clients (TRUE, transport_client_1, expected_transport_1, addr_client_1, transport_client_2, expected_transport_2, addr_client_2, FALSE); } GST_END_TEST; /* test if two multicast clients get the different transport settings: the first client lets * the server select the multicast address and the ports while the second client requests * the specific transport configuration. * CASE: media is shared */ GST_START_TEST (test_client_multicast_two_clients_second_specific_transport_shared_media) { const gchar *transport_client_1 = "RTP/AVP;multicast;mode=\"PLAY\""; const gchar *expected_transport_1 = "RTP/AVP;multicast;destination=233.252.0.1;" "ttl=1;port=5000-5001;mode=\"PLAY\""; const gchar *addr_client_1 = "233.252.0.1:5000"; const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;" "ttl=2;port=5004-5005;mode=\"PLAY\""; const gchar *expected_transport_2 = transport_client_2; const gchar *addr_client_2 = "233.252.0.2:5004"; mcast_transport_two_clients (TRUE, transport_client_1, expected_transport_1, addr_client_1, transport_client_2, expected_transport_2, addr_client_2, FALSE); } GST_END_TEST; /* test if the maximum ttl multicast value is chosen by the server * CASE: the first client provides the highest ttl value */ GST_START_TEST (test_client_multicast_max_ttl_first_client) { const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;" "ttl=3;port=5000-5001;mode=\"PLAY\""; const gchar *expected_transport_1 = transport_client_1; const gchar *addr_client_1 = "233.252.0.1:5000"; const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;" "ttl=1;port=5002-5003;mode=\"PLAY\""; const gchar *expected_transport_2 = "RTP/AVP;multicast;destination=233.252.0.2;" "ttl=3;port=5002-5003;mode=\"PLAY\""; const gchar *addr_client_2 = "233.252.0.2:5002"; mcast_transport_two_clients (TRUE, transport_client_1, expected_transport_1, addr_client_1, transport_client_2, expected_transport_2, addr_client_2, FALSE); } GST_END_TEST; /* test if the maximum ttl multicast value is chosen by the server * CASE: the second client provides the highest ttl value */ GST_START_TEST (test_client_multicast_max_ttl_second_client) { const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;" "ttl=2;port=5000-5001;mode=\"PLAY\""; const gchar *expected_transport_1 = transport_client_1; const gchar *addr_client_1 = "233.252.0.1:5000"; const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;" "ttl=4;port=5002-5003;mode=\"PLAY\""; const gchar *expected_transport_2 = transport_client_2; const gchar *addr_client_2 = "233.252.0.2:5002"; mcast_transport_two_clients (TRUE, transport_client_1, expected_transport_1, addr_client_1, transport_client_2, expected_transport_2, addr_client_2, FALSE); } GST_END_TEST; GST_START_TEST (test_client_multicast_invalid_ttl) { GstRTSPClient *client; GstRTSPMessage request = { 0, }; gchar *str; GstRTSPSessionPool *session_pool; GstRTSPContext ctx = { NULL }; client = setup_multicast_client (3, "/test"); ctx.client = client; ctx.auth = gst_rtsp_auth_new (); ctx.token = gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS, G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING, "user", NULL); gst_rtsp_context_push_current (&ctx); /* simple SETUP with an invalid ttl=0 */ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP, "rtsp://localhost/test/stream=0") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, "RTP/AVP;multicast;destination=233.252.0.1;ttl=0;port=5000-5001;"); gst_rtsp_client_set_send_func (client, test_setup_response_461, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); session_pool = gst_rtsp_client_get_session_pool (client); fail_unless (session_pool != NULL); fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 0); g_object_unref (session_pool); teardown_client (client); g_object_unref (ctx.auth); gst_rtsp_token_unref (ctx.token); gst_rtsp_context_pop_current (&ctx); } GST_END_TEST; static gboolean test_response_scale_speed (GstRTSPClient * client, GstRTSPMessage * response, gboolean close, gpointer user_data) { GstRTSPStatusCode code; const gchar *reason; GstRTSPVersion version; gchar *header_value; fail_unless (gst_rtsp_message_get_type (response) == GST_RTSP_MESSAGE_RESPONSE); fail_unless (gst_rtsp_message_parse_response (response, &code, &reason, &version) == GST_RTSP_OK); fail_unless (code == GST_RTSP_STS_OK); fail_unless (g_str_equal (reason, "OK")); fail_unless (version == GST_RTSP_VERSION_1_0); fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_RANGE, &header_value, 0) == GST_RTSP_OK); if (expected_scale_header != NULL) { fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SCALE, &header_value, 0) == GST_RTSP_OK); ck_assert_str_eq (header_value, expected_scale_header); } else { fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SCALE, &header_value, 0) == GST_RTSP_ENOTIMPL); } if (expected_speed_header != NULL) { fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SPEED, &header_value, 0) == GST_RTSP_OK); ck_assert_str_eq (header_value, expected_speed_header); } else { fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SPEED, &header_value, 0) == GST_RTSP_ENOTIMPL); } return TRUE; } /* Probe that tweaks segment events according to the values of the * fake_rate_value and fake_applied_rate_value variables. Used to simulate * seek results with different combinations of rate and applied rate. */ static GstPadProbeReturn rate_tweaking_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data) { GstEvent *event = GST_PAD_PROBE_INFO_EVENT (info); GstSegment segment; if (GST_EVENT_TYPE (event) == GST_EVENT_SEGMENT) { GST_DEBUG ("got segment event %" GST_PTR_FORMAT, event); gst_event_copy_segment (event, &segment); if (fake_applied_rate_value) segment.applied_rate = fake_applied_rate_value; if (fake_rate_value) segment.rate = fake_rate_value; gst_event_unref (event); info->data = gst_event_new_segment (&segment); GST_DEBUG ("forwarding segment event %" GST_PTR_FORMAT, GST_EVENT (info->data)); } return GST_PAD_PROBE_OK; } static void attach_rate_tweaking_probe (void) { GstRTSPContext *ctx; GstRTSPMedia *media; GstRTSPStream *stream; GstPad *srcpad; fail_unless ((ctx = gst_rtsp_context_get_current ()) != NULL); media = ctx->media; fail_unless (media != NULL); stream = gst_rtsp_media_get_stream (media, 0); fail_unless (stream != NULL); srcpad = gst_rtsp_stream_get_srcpad (stream); fail_unless (srcpad != NULL); GST_DEBUG ("adding rate_tweaking_probe"); gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM, rate_tweaking_probe, NULL, NULL); gst_object_unref (srcpad); } static void do_test_scale_and_speed (const gchar * scale, const gchar * speed, GstRTSPStatusCode expected_response_code) { GstRTSPClient *client; GstRTSPMessage request = { 0, }; gchar *str; GstRTSPContext ctx = { NULL }; client = setup_multicast_client (1, "/test"); ctx.client = client; ctx.auth = gst_rtsp_auth_new (); ctx.token = gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS, G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING, "user", NULL); gst_rtsp_context_push_current (&ctx); expected_session_timeout = 20; g_signal_connect (G_OBJECT (client), "new-session", G_CALLBACK (new_session_cb), NULL); fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP, "rtsp://localhost/test/stream=0") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, "RTP/AVP;multicast"); expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;" "ttl=1;port=.*;mode=\"PLAY\""; gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); expected_transport = NULL; expected_session_timeout = 60; if (fake_applied_rate_value || fake_rate_value) attach_rate_tweaking_probe (); fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, "rtsp://localhost/test") == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id); if (scale != NULL) gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, scale); if (speed != NULL) gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, speed); if (expected_response_code == GST_RTSP_STS_BAD_REQUEST) gst_rtsp_client_set_send_func (client, test_response_400, NULL, NULL); else gst_rtsp_client_set_send_func (client, test_response_scale_speed, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); send_teardown (client, "rtsp://localhost/test"); teardown_client (client); g_object_unref (ctx.auth); gst_rtsp_token_unref (ctx.token); gst_rtsp_context_pop_current (&ctx); } GST_START_TEST (test_scale_and_speed) { /* no scale/speed requested, no scale/speed should be received */ expected_scale_header = NULL; expected_speed_header = NULL; do_test_scale_and_speed (NULL, NULL, GST_RTSP_STS_OK); /* scale requested, scale should be received */ fake_applied_rate_value = 2; fake_rate_value = 1; expected_scale_header = "2.000"; expected_speed_header = NULL; do_test_scale_and_speed ("2.000", NULL, GST_RTSP_STS_OK); /* speed requested, speed should be received */ fake_applied_rate_value = 0; fake_rate_value = 0; expected_scale_header = NULL; expected_speed_header = "2.000"; do_test_scale_and_speed (NULL, "2.000", GST_RTSP_STS_OK); /* both requested, both should be received */ fake_applied_rate_value = 2; fake_rate_value = 2; expected_scale_header = "2.000"; expected_speed_header = "2.000"; do_test_scale_and_speed ("2", "2", GST_RTSP_STS_OK); /* scale requested but media doesn't handle scaling so both should be * received, with scale set to 1.000 and speed set to (requested scale * requested speed) */ fake_applied_rate_value = 0; fake_rate_value = 5; expected_scale_header = "1.000"; expected_speed_header = "5.000"; do_test_scale_and_speed ("5", NULL, GST_RTSP_STS_OK); /* both requested but media only handles scaling so both should be received, * with scale set to (requested scale * requested speed) and speed set to 1.00 */ fake_rate_value = 1.000; fake_applied_rate_value = 4.000; expected_scale_header = "4.000"; expected_speed_header = "1.000"; do_test_scale_and_speed ("2", "2", GST_RTSP_STS_OK); /* test invalid values */ fake_applied_rate_value = 0; fake_rate_value = 0; expected_scale_header = NULL; expected_speed_header = NULL; /* scale or speed not decimal values */ do_test_scale_and_speed ("x", NULL, GST_RTSP_STS_BAD_REQUEST); do_test_scale_and_speed (NULL, "y", GST_RTSP_STS_BAD_REQUEST); /* scale or speed illegal decimal values */ do_test_scale_and_speed ("0", NULL, GST_RTSP_STS_BAD_REQUEST); do_test_scale_and_speed (NULL, "0", GST_RTSP_STS_BAD_REQUEST); do_test_scale_and_speed (NULL, "-2", GST_RTSP_STS_BAD_REQUEST); } GST_END_TEST static void test_client_play_sub (const gchar * mount_point, const gchar * url1, const gchar * url2) { GstRTSPClient *client; GstRTSPMessage request = { 0, }; gchar *str; GstRTSPContext ctx = { NULL }; client = setup_multicast_client (1, mount_point); ctx.client = client; ctx.auth = gst_rtsp_auth_new (); ctx.token = gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING, "user", NULL); gst_rtsp_context_push_current (&ctx); fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP, url1) == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, "RTP/AVP;multicast"); /* destination is from adress pool */ expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;" "ttl=1;port=.*;mode=\"PLAY\""; gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); expected_transport = NULL; fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, url2) == GST_RTSP_OK); str = g_strdup_printf ("%d", cseq); gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id); gst_rtsp_client_set_send_func (client, test_response_play_200, NULL, NULL); fail_unless (gst_rtsp_client_handle_message (client, &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); send_teardown (client, url2); teardown_client (client); g_object_unref (ctx.auth); gst_rtsp_token_unref (ctx.token); gst_rtsp_context_pop_current (&ctx); } GST_START_TEST (test_client_play) { test_client_play_sub ("/test", "rtsp://localhost/test/stream=0", "rtsp://localhost/test"); } GST_END_TEST; GST_START_TEST (test_client_play_root_mount_point) { test_client_play_sub ("/", "rtsp://localhost/stream=0", "rtsp://localhost"); } GST_END_TEST; #define RTSP_CLIENT_TEST_TYPE (rtsp_client_test_get_type ()) #define RTSP_CLIENT_TEST_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), RTSP_CLIENT_TEST_TYPE, RtspClientTestClass)) typedef struct RtspClientTest { GstRTSPClient parent; } RtspClientTest; typedef struct RtspClientTestClass { GstRTSPClientClass parent_class; } RtspClientTestClass; GType rtsp_client_test_get_type (void); G_DEFINE_TYPE (RtspClientTest, rtsp_client_test, GST_TYPE_RTSP_CLIENT); static void rtsp_client_test_init (RtspClientTest * client) { } static void rtsp_client_test_class_init (RtspClientTestClass * klass) { } static GstRTSPStatusCode adjust_error_code_cb (GstRTSPClient * client, GstRTSPContext * ctx, GstRTSPStatusCode code) { return GST_RTSP_STS_NOT_FOUND; } GST_START_TEST (test_adjust_error_code) { RtspClientTest *client; RtspClientTestClass *klass; GstRTSPClientClass *base_klass; GstRTSPMessage request = { 0, }; client = g_object_new (RTSP_CLIENT_TEST_TYPE, NULL); /* invalid request to trigger error response */ ck_assert (gst_rtsp_message_init_request (&request, GST_RTSP_INVALID, "foopy://padoop/") == GST_RTSP_OK); /* expect non-adjusted error response 400 */ gst_rtsp_client_set_send_func (GST_RTSP_CLIENT (client), test_response_400, NULL, NULL); ck_assert (gst_rtsp_client_handle_message (GST_RTSP_CLIENT (client), &request) == GST_RTSP_OK); /* override virtual function for adjusting error code */ klass = RTSP_CLIENT_TEST_GET_CLASS (client); base_klass = GST_RTSP_CLIENT_CLASS (klass); base_klass->adjust_error_code = adjust_error_code_cb; /* expect error adjusted to 404 */ gst_rtsp_client_set_send_func (GST_RTSP_CLIENT (client), test_response_404, NULL, NULL); ck_assert (gst_rtsp_client_handle_message (GST_RTSP_CLIENT (client), &request) == GST_RTSP_OK); gst_rtsp_message_unset (&request); g_object_unref (client); } GST_END_TEST; static Suite * rtspclient_suite (void) { Suite *s = suite_create ("rtspclient"); TCase *tc = tcase_create ("general"); suite_add_tcase (s, tc); tcase_set_timeout (tc, 20); tcase_add_test (tc, test_require); tcase_add_test (tc, test_request); tcase_add_test (tc, test_options); tcase_add_test (tc, test_describe); tcase_add_test (tc, test_describe_root_mount_point); tcase_add_test (tc, test_setup_tcp); tcase_add_test (tc, test_setup_tcp_root_mount_point); tcase_add_test (tc, test_setup_no_rtcp); tcase_add_test (tc, test_setup_tcp_two_streams_same_channels); tcase_add_test (tc, test_setup_tcp_two_streams_same_channels_root_mount_point); tcase_add_test (tc, test_client_multicast_transport_404); tcase_add_test (tc, test_client_multicast_transport); tcase_add_test (tc, test_client_multicast_ignore_transport_specific); tcase_add_test (tc, test_client_multicast_transport_specific); tcase_add_test (tc, test_client_sdp_with_max_bitrate_tag); tcase_add_test (tc, test_client_sdp_with_bitrate_tag); tcase_add_test (tc, test_client_sdp_with_max_bitrate_and_bitrate_tags); tcase_add_test (tc, test_client_sdp_with_no_bitrate_tags); tcase_add_test (tc, test_client_multicast_transport_specific_two_clients_shared_media); tcase_add_test (tc, test_client_multicast_transport_specific_two_clients); #ifndef G_OS_WIN32 tcase_add_test (tc, test_client_multicast_transport_specific_two_clients_same_ports); #else /* skip the test on windows as the test restricts the multicast sockets to multicast traffic only, * by specifying the multicast IP as the bind address and this currently doesn't work on Windows */ tcase_skip_broken_test (tc, test_client_multicast_transport_specific_two_clients_same_ports); #endif tcase_add_test (tc, test_client_multicast_transport_specific_two_clients_same_destination); tcase_add_test (tc, test_client_multicast_transport_specific_two_clients_shared_media_same_transport); tcase_add_test (tc, test_client_multicast_two_clients_shared_media); tcase_add_test (tc, test_client_multicast_two_clients_shared_media_teardown_play); tcase_add_test (tc, test_client_multicast_two_clients_not_shared_media_teardown_play); tcase_add_test (tc, test_client_multicast_two_clients_first_specific_transport_shared_media); tcase_add_test (tc, test_client_multicast_two_clients_second_specific_transport_shared_media); tcase_add_test (tc, test_client_multicast_transport_specific_no_address_in_pool); tcase_add_test (tc, test_client_multicast_max_ttl_first_client); tcase_add_test (tc, test_client_multicast_max_ttl_second_client); tcase_add_test (tc, test_client_multicast_invalid_ttl); tcase_add_test (tc, test_scale_and_speed); tcase_add_test (tc, test_client_play); tcase_add_test (tc, test_client_play_root_mount_point); tcase_add_test (tc, test_adjust_error_code); return s; } GST_CHECK_MAIN (rtspclient);