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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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client: Don't reserve multicast address in the client setting case
When two multicast clients request specific transport configurations, and "transport.client-settings" parameter is set to true, it's wrong to actually require that these two clients request the same multicast group. Removed test_client_multicast_invalid_transport_specific test cases as they wrongly require that the requested destination address is supposed to be present in the address pool, also in the case when "transport.client-settings" parameter is set to true. Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa https://bugzilla.gnome.org/show_bug.cgi?id=793441
This commit is contained in:
parent
a7bb684e9b
commit
308480e762
2 changed files with 162 additions and 125 deletions
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@ -1990,17 +1990,16 @@ default_configure_client_transport (GstRTSPClient * client,
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goto error_allocating_ports;
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if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
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GstRTSPAddress *addr = NULL;
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/* FIXME: the address has been successfully allocated, however, in
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* the use_client_settings case we need to verify that the allocated
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* address is the one requested by the client and if this address is
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* an allowed destination. Verifying this via the address pool in not
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* the proper way as the address pool should only be used for choosing
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* the server-selected address/port pairs. */
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if (use_client_settings) {
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/* the address has been successfully allocated, let's check if it's
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* the one requested by the client */
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addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination,
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ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
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if (!use_client_settings) {
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GstRTSPAddress *addr = NULL;
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if (addr == NULL)
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goto no_address;
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} else {
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g_free (ct->destination);
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addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
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if (addr == NULL)
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@ -2009,9 +2008,8 @@ default_configure_client_transport (GstRTSPClient * client,
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ct->port.min = addr->port;
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ct->port.max = addr->port + addr->n_ports - 1;
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ct->ttl = addr->ttl;
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gst_rtsp_address_free (addr);
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}
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gst_rtsp_address_free (addr);
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} else {
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GstRTSPUrl *url;
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@ -525,7 +525,6 @@ test_setup_response_200_multicast (GstRTSPClient * client,
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session_pool = gst_rtsp_client_get_session_pool (client);
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fail_unless (session_pool != NULL);
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fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 1);
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session = gst_rtsp_session_pool_find (session_pool, session_hdr_params[0]);
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g_strfreev (session_hdr_params);
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@ -727,117 +726,6 @@ GST_START_TEST (test_client_multicast_ignore_transport_specific)
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GST_END_TEST;
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static gboolean
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test_setup_response_461 (GstRTSPClient * client,
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GstRTSPMessage * response, gboolean close, gpointer user_data)
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{
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GstRTSPStatusCode code;
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const gchar *reason;
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GstRTSPVersion version;
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gchar *str;
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fail_unless (expected_transport == NULL);
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fail_unless (gst_rtsp_message_get_type (response) ==
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GST_RTSP_MESSAGE_RESPONSE);
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fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
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&version)
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== GST_RTSP_OK);
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fail_unless (code == GST_RTSP_STS_UNSUPPORTED_TRANSPORT);
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fail_unless (g_str_equal (reason, "Unsupported transport"));
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fail_unless (version == GST_RTSP_VERSION_1_0);
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fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str,
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0) == GST_RTSP_OK);
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fail_unless (atoi (str) == cseq++);
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return TRUE;
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}
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GST_START_TEST (test_client_multicast_invalid_transport_specific)
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{
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GstRTSPClient *client;
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GstRTSPMessage request = { 0, };
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gchar *str;
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GstRTSPSessionPool *session_pool;
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GstRTSPContext ctx = { NULL };
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client = setup_multicast_client ();
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ctx.client = client;
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ctx.auth = gst_rtsp_auth_new ();
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ctx.token =
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gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
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G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
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"user", NULL);
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gst_rtsp_context_push_current (&ctx);
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/* simple SETUP with a valid URI and multicast, but an invalid ip */
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fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
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"rtsp://localhost/test/stream=0") == GST_RTSP_OK);
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str = g_strdup_printf ("%d", cseq);
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gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
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gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
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"RTP/AVP;multicast;destination=233.252.0.2;ttl=1;port=5000-5001;");
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gst_rtsp_client_set_send_func (client, test_setup_response_461, NULL, NULL);
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fail_unless (gst_rtsp_client_handle_message (client,
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&request) == GST_RTSP_OK);
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gst_rtsp_message_unset (&request);
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session_pool = gst_rtsp_client_get_session_pool (client);
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fail_unless (session_pool != NULL);
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fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 0);
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g_object_unref (session_pool);
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/* simple SETUP with a valid URI and multicast, but an invalid prt */
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fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
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"rtsp://localhost/test/stream=0") == GST_RTSP_OK);
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str = g_strdup_printf ("%d", cseq);
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gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
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gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
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"RTP/AVP;multicast;destination=233.252.0.1;ttl=1;port=6000-6001;");
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gst_rtsp_client_set_send_func (client, test_setup_response_461, NULL, NULL);
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fail_unless (gst_rtsp_client_handle_message (client,
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&request) == GST_RTSP_OK);
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gst_rtsp_message_unset (&request);
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session_pool = gst_rtsp_client_get_session_pool (client);
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fail_unless (session_pool != NULL);
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fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 0);
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g_object_unref (session_pool);
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/* simple SETUP with a valid URI and multicast, but an invalid ttl */
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fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
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"rtsp://localhost/test/stream=0") == GST_RTSP_OK);
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str = g_strdup_printf ("%d", cseq);
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gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
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gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
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"RTP/AVP;multicast;destination=233.252.0.1;ttl=2;port=5000-5001;");
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gst_rtsp_client_set_send_func (client, test_setup_response_461, NULL, NULL);
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fail_unless (gst_rtsp_client_handle_message (client,
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&request) == GST_RTSP_OK);
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gst_rtsp_message_unset (&request);
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session_pool = gst_rtsp_client_get_session_pool (client);
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fail_unless (session_pool != NULL);
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fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 0);
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g_object_unref (session_pool);
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teardown_client (client);
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g_object_unref (ctx.auth);
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gst_rtsp_token_unref (ctx.token);
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gst_rtsp_context_pop_current (&ctx);
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}
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GST_END_TEST;
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GST_START_TEST (test_client_multicast_transport_specific)
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{
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GstRTSPClient *client;
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@ -859,7 +747,7 @@ GST_START_TEST (test_client_multicast_transport_specific)
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expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
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"ttl=1;port=5000-5001;mode=\"PLAY\"";
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/* simple SETUP with a valid URI and multicast, but an invalid ip */
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/* simple SETUP with a valid URI */
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fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
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"rtsp://localhost/test/stream=0") == GST_RTSP_OK);
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str = g_strdup_printf ("%d", cseq);
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@ -1031,6 +919,156 @@ GST_START_TEST (test_client_sdp_with_no_bitrate_tags)
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GST_END_TEST;
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static void
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mcast_transport_specific_two_clients (gboolean shared)
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{
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GstRTSPClient *client, *client2;
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GstRTSPMessage request = { 0, };
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gchar *str;
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GstRTSPSessionPool *session_pool;
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GstRTSPContext ctx = { NULL };
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GstRTSPContext ctx2 = { NULL };
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GstRTSPMountPoints *mount_points;
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GstRTSPMediaFactory *factory;
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GstRTSPAddressPool *address_pool;
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GstRTSPThreadPool *thread_pool;
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gchar *session_id1;
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mount_points = gst_rtsp_mount_points_new ();
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factory = gst_rtsp_media_factory_new ();
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if (shared)
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gst_rtsp_media_factory_set_shared (factory, TRUE);
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gst_rtsp_media_factory_set_max_mcast_ttl (factory, 5);
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gst_rtsp_media_factory_set_launch (factory,
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"audiotestsrc ! audio/x-raw,rate=44100 ! audioconvert ! rtpL16pay name=pay0");
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address_pool = gst_rtsp_address_pool_new ();
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fail_unless (gst_rtsp_address_pool_add_range (address_pool,
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"233.252.0.1", "233.252.0.1", 5000, 5001, 1));
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gst_rtsp_media_factory_set_address_pool (factory, address_pool);
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gst_rtsp_media_factory_add_role (factory, "user",
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"media.factory.access", G_TYPE_BOOLEAN, TRUE,
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"media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
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gst_rtsp_mount_points_add_factory (mount_points, "/test", factory);
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session_pool = gst_rtsp_session_pool_new ();
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thread_pool = gst_rtsp_thread_pool_new ();
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/* first multicast client with transport specific request */
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client = gst_rtsp_client_new ();
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gst_rtsp_client_set_session_pool (client, session_pool);
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gst_rtsp_client_set_mount_points (client, mount_points);
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gst_rtsp_client_set_thread_pool (client, thread_pool);
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ctx.client = client;
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ctx.auth = gst_rtsp_auth_new ();
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ctx.token =
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gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
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G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
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"user", NULL);
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gst_rtsp_context_push_current (&ctx);
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expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
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"ttl=1;port=5000-5001;mode=\"PLAY\"";
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/* send SETUP request */
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fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
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"rtsp://localhost/test/stream=0") == GST_RTSP_OK);
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str = g_strdup_printf ("%d", cseq);
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gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
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gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
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expected_transport);
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gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
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NULL, NULL);
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fail_unless (gst_rtsp_client_handle_message (client,
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&request) == GST_RTSP_OK);
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gst_rtsp_message_unset (&request);
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expected_transport = NULL;
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/* send PLAY request */
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fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY,
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"rtsp://localhost/test") == GST_RTSP_OK);
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str = g_strdup_printf ("%d", cseq);
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gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
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gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
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gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL);
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fail_unless (gst_rtsp_client_handle_message (client,
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&request) == GST_RTSP_OK);
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gst_rtsp_message_unset (&request);
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gst_rtsp_context_pop_current (&ctx);
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session_id1 = session_id;
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/* second multicast client with transport specific request */
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cseq = 0;
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client2 = gst_rtsp_client_new ();
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gst_rtsp_client_set_session_pool (client2, session_pool);
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gst_rtsp_client_set_mount_points (client2, mount_points);
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gst_rtsp_client_set_thread_pool (client2, thread_pool);
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ctx2.client = client2;
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ctx2.auth = gst_rtsp_auth_new ();
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ctx2.token =
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gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
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G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
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"user", NULL);
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gst_rtsp_context_push_current (&ctx2);
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expected_transport = "RTP/AVP;multicast;destination=233.252.0.2;"
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"ttl=1;port=5002-5003;mode=\"PLAY\"";
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/* send SETUP request */
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fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
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"rtsp://localhost/test/stream=0") == GST_RTSP_OK);
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str = g_strdup_printf ("%d", cseq);
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gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
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gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
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expected_transport);
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gst_rtsp_client_set_send_func (client2, test_setup_response_200_multicast,
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NULL, NULL);
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fail_unless (gst_rtsp_client_handle_message (client2,
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&request) == GST_RTSP_OK);
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gst_rtsp_message_unset (&request);
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expected_transport = NULL;
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/* send PLAY request */
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fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY,
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"rtsp://localhost/test") == GST_RTSP_OK);
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str = g_strdup_printf ("%d", cseq);
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gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
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gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
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gst_rtsp_client_set_send_func (client2, test_response_200, NULL, NULL);
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fail_unless (gst_rtsp_client_handle_message (client2,
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&request) == GST_RTSP_OK);
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gst_rtsp_message_unset (&request);
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send_teardown (client2);
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gst_rtsp_context_pop_current (&ctx2);
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gst_rtsp_context_push_current (&ctx);
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session_id = session_id1;
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send_teardown (client);
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gst_rtsp_context_pop_current (&ctx);
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teardown_client (client);
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teardown_client (client2);
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g_object_unref (ctx.auth);
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g_object_unref (ctx2.auth);
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gst_rtsp_token_unref (ctx.token);
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gst_rtsp_token_unref (ctx2.token);
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g_object_unref (mount_points);
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g_object_unref (session_pool);
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g_object_unref (address_pool);
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g_object_unref (thread_pool);
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}
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/* test if two multicast clients can choose different transport settings */
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GST_START_TEST
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(test_client_multicast_transport_specific_two_clients_shared_media) {
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mcast_transport_specific_two_clients (TRUE);
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}
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GST_END_TEST;
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static Suite *
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rtspclient_suite (void)
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{
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@ -1046,12 +1084,13 @@ rtspclient_suite (void)
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tcase_add_test (tc, test_client_multicast_transport_404);
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tcase_add_test (tc, test_client_multicast_transport);
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tcase_add_test (tc, test_client_multicast_ignore_transport_specific);
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tcase_add_test (tc, test_client_multicast_invalid_transport_specific);
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tcase_add_test (tc, test_client_multicast_transport_specific);
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tcase_add_test (tc, test_client_sdp_with_max_bitrate_tag);
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tcase_add_test (tc, test_client_sdp_with_bitrate_tag);
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tcase_add_test (tc, test_client_sdp_with_max_bitrate_and_bitrate_tags);
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tcase_add_test (tc, test_client_sdp_with_no_bitrate_tags);
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tcase_add_test (tc,
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test_client_multicast_transport_specific_two_clients_shared_media);
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return s;
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}
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