rtsp-stream: Don't require address pool in the transport specific case

If "transport.client-settings" parameter is set to true, the client is
allowed to specify destination, ports and ttl.
There is no need for pre-configured address pool.

https://bugzilla.gnome.org/show_bug.cgi?id=793441

1 files changed, 47 insertions, 4 deletions

diff --git a/tests/check/gst/client.c b/tests/check/gst/client.c
index ab95509..d4fdd9e 100644
--- a/tests/check/gst/client.c
+++ b/tests/check/gst/client.c
@@ -726,7 +726,8 @@ GST_START_TEST (test_client_multicast_ignore_transport_specific)
 
 GST_END_TEST;
 
-GST_START_TEST (test_client_multicast_transport_specific)
+static void
+multicast_transport_specific (void)
 {
   GstRTSPClient *client;
   GstRTSPMessage request = { 0, };
@@ -760,7 +761,6 @@ GST_START_TEST (test_client_multicast_transport_specific)
   fail_unless (gst_rtsp_client_handle_message (client,
           &request) == GST_RTSP_OK);
   gst_rtsp_message_unset (&request);
-  expected_transport = NULL;
 
   gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
       NULL, NULL);
@@ -769,14 +769,44 @@ GST_START_TEST (test_client_multicast_transport_specific)
   fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 1);
   g_object_unref (session_pool);
 
-  send_teardown (client);
+  /* 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);
   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=6000-6001;mode=\"PLAY\"";
+  multicast_transport_specific ();
+  expected_transport = NULL;
+}
+
 GST_END_TEST;
 
 static gboolean
@@ -1061,7 +1091,8 @@ mcast_transport_specific_two_clients (gboolean shared)
   g_object_unref (thread_pool);
 }
 
-/* test if two multicast clients can choose different transport settings */
+/* 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) {
   mcast_transport_specific_two_clients (TRUE);
@@ -1069,6 +1100,15 @@ GST_START_TEST
 
 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)
+{
+  mcast_transport_specific_two_clients (FALSE);
+}
+
+GST_END_TEST;
+
 static Suite *
 rtspclient_suite (void)
 {
@@ -1091,6 +1131,9 @@ rtspclient_suite (void)
   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);
+  tcase_add_test (tc,
+      test_client_multicast_transport_specific_no_address_in_pool);
 
   return s;
 }