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rtsp-media: Fix multicast use case with common media

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Use case
client 1: SETUP
client 1: PLAY
client 2: SETUP
client 1: TEARDOWN
client 2: PLAY
client 2: TEARDOWN
This commit is contained in:
Göran Jönsson 2019-02-19 09:45:08 +01:00
parent afb27f91cf
commit 7e01dfd151
2 changed files with 211 additions and 2 deletions
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6
gst/rtsp-server/rtsp-media.c
View file

@ -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;

207
tests/check/gst/client.c
View file

@ -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.
@ -1543,6 +1746,10 @@ rtspclient_suite (void)
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,