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authorGöran Jönsson <goranjn@axis.com>2019-02-19 09:45:08 +0100
committerGöran Jönsson <goranjn@axis.com>2019-02-19 12:12:34 +0100
commit7e01dfd151c35868ac679ca12f30caaefa9bbb8b (patch)
tree32199f33f192b0354aa6fdaec574dfdc6904783d
parentafb27f91cfec706d2a4dbf8a6c787504731035a3 (diff)
rtsp-media: Fix multicast use case with common media
Use case client 1: SETUP client 1: PLAY client 2: SETUP client 1: TEARDOWN client 2: PLAY client 2: TEARDOWN
-rw-r--r--gst/rtsp-server/rtsp-media.c6
-rw-r--r--tests/check/gst/client.c207
2 files changed, 211 insertions, 2 deletions
diff --git a/gst/rtsp-server/rtsp-media.c b/gst/rtsp-server/rtsp-media.c
index bfcba34..be98afd 100644
--- a/gst/rtsp-server/rtsp-media.c
+++ b/gst/rtsp-server/rtsp-media.c
@@ -4434,8 +4434,10 @@ gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
/* we just activated the first media, do the playing state change */
if (old_active == 0 && activate)
do_state = TRUE;
- /* if we have no more active media, do the downward state changes */
- else if (priv->n_active == 0)
+ /* if we have no more active media and prepare count is not indicate
+ * that there are new session/sessions ongoing,
+ * do the downward state changes */
+ else if (priv->n_active == 0 && priv->prepare_count <= 1)
do_state = TRUE;
else
do_state = FALSE;
diff --git a/tests/check/gst/client.c b/tests/check/gst/client.c
index 6c5240e..cbe846e 100644
--- a/tests/check/gst/client.c
+++ b/tests/check/gst/client.c
@@ -1238,6 +1238,167 @@ mcast_transport_two_clients (gboolean shared, const gchar * transport1,
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);
+ 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);
+ 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
@@ -1372,6 +1533,48 @@ GST_START_TEST (test_client_multicast_two_clients_shared_media)
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.
@@ -1544,6 +1747,10 @@ rtspclient_suite (void)
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);