*/
/* FIXME: could be more adequately solved by making it possible
* to set a socket on multiudpsink after it has already been started */
- if (!gst_rtsp_stream_allocate_udp_sockets (ctx->stream, G_SOCKET_FAMILY_IPV4, ct,
- use_client_settings) && family == G_SOCKET_FAMILY_IPV4)
+ if (!gst_rtsp_stream_allocate_udp_sockets (ctx->stream,
+ G_SOCKET_FAMILY_IPV4, ct, use_client_settings)
+ && family == G_SOCKET_FAMILY_IPV4)
goto error_allocating_ports;
- if (!gst_rtsp_stream_allocate_udp_sockets (ctx->stream, G_SOCKET_FAMILY_IPV6, ct,
- use_client_settings) && family == G_SOCKET_FAMILY_IPV6)
+ if (!gst_rtsp_stream_allocate_udp_sockets (ctx->stream,
+ G_SOCKET_FAMILY_IPV6, ct, use_client_settings)
+ && family == G_SOCKET_FAMILY_IPV6)
goto error_allocating_ports;
if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
- GstRTSPAddress *addr = NULL;
+ /* FIXME: the address has been successfully allocated, however, in
+ * the use_client_settings case we need to verify that the allocated
+ * address is the one requested by the client and if this address is
+ * an allowed destination. Verifying this via the address pool in not
+ * the proper way as the address pool should only be used for choosing
+ * the server-selected address/port pairs. */
- if (use_client_settings) {
- /* the address has been successfully allocated, let's check if it's
- * the one requested by the client */
- addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination,
- ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
+ if (!use_client_settings) {
+ GstRTSPAddress *addr = NULL;
- if (addr == NULL)
- goto no_address;
- } else {
g_free (ct->destination);
addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
if (addr == NULL)
ct->port.min = addr->port;
ct->port.max = addr->port + addr->n_ports - 1;
ct->ttl = addr->ttl;
+ gst_rtsp_address_free (addr);
}
-
- gst_rtsp_address_free (addr);
} else {
GstRTSPUrl *url;
GstRTSPTransportMode transport_mode;
gboolean stop_on_disconnect;
gchar *multicast_iface;
+ guint max_mcast_ttl;
GstClockTime rtx_time;
guint latency;
GST_RTSP_LOWER_TRANS_TCP
#define DEFAULT_BUFFER_SIZE 0x80000
#define DEFAULT_LATENCY 200
+#define DEFAULT_MAX_MCAST_TTL 255
#define DEFAULT_TRANSPORT_MODE GST_RTSP_TRANSPORT_MODE_PLAY
#define DEFAULT_STOP_ON_DISCONNECT TRUE
#define DEFAULT_DO_RETRANSMISSION FALSE
PROP_TRANSPORT_MODE,
PROP_STOP_ON_DISCONNECT,
PROP_CLOCK,
+ PROP_MAX_MCAST_TTL,
PROP_LAST
};
"medias of this factory", GST_TYPE_CLOCK,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_MAX_MCAST_TTL,
+ g_param_spec_uint ("max-mcast-ttl", "Maximum multicast ttl",
+ "The maximum time-to-live value of outgoing multicast packets", 1,
+ 255, DEFAULT_MAX_MCAST_TTL,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
gst_rtsp_media_factory_signals[SIGNAL_MEDIA_CONSTRUCTED] =
g_signal_new ("media-constructed", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaFactoryClass,
priv->stop_on_disconnect = DEFAULT_STOP_ON_DISCONNECT;
priv->publish_clock_mode = GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK;
priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
+ priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
g_mutex_init (&priv->lock);
g_mutex_init (&priv->medias_lock);
case PROP_CLOCK:
g_value_take_object (value, gst_rtsp_media_factory_get_clock (factory));
break;
+ case PROP_MAX_MCAST_TTL:
+ g_value_set_uint (value,
+ gst_rtsp_media_factory_get_max_mcast_ttl (factory));
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
case PROP_CLOCK:
gst_rtsp_media_factory_set_clock (factory, g_value_get_object (value));
break;
+ case PROP_MAX_MCAST_TTL:
+ gst_rtsp_media_factory_set_max_mcast_ttl (factory,
+ g_value_get_uint (value));
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
return ret;
}
+/**
+ * gst_rtsp_media_factory_set_max_mcast_ttl:
+ * @factory: a #GstRTSPMedia
+ * @ttl: the new multicast ttl value
+ *
+ * Set the maximum time-to-live value of outgoing multicast packets.
+ *
+ * Returns: %TRUE if the requested ttl has been set successfully.
+ */
+gboolean
+gst_rtsp_media_factory_set_max_mcast_ttl (GstRTSPMediaFactory * factory,
+ guint ttl)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), FALSE);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ if (ttl == 0 || ttl > DEFAULT_MAX_MCAST_TTL) {
+ GST_WARNING_OBJECT (factory, "The requested mcast TTL value is not valid.");
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+ return FALSE;
+ }
+ priv->max_mcast_ttl = ttl;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_media_factory_get_max_mcast_ttl:
+ * @factory: a #GstRTSPMedia
+ *
+ * Get the the maximum time-to-live value of outgoing multicast packets.
+ *
+ * Returns: the maximum time-to-live value of outgoing multicast packets.
+ */
+guint
+gst_rtsp_media_factory_get_max_mcast_ttl (GstRTSPMediaFactory * factory)
+{
+ GstRTSPMediaFactoryPrivate *priv;
+ guint result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), 0);
+
+ priv = factory->priv;
+
+ GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+ result = priv->max_mcast_ttl;
+ GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+ return result;
+}
+
static gchar *
default_gen_key (GstRTSPMediaFactory * factory, const GstRTSPUrl * url)
{
GstClock *clock;
gchar *multicast_iface;
GstRTSPPublishClockMode publish_clock_mode;
+ guint ttl;
/* configure the sharedness */
GST_RTSP_MEDIA_FACTORY_LOCK (factory);
stop_on_disconnect = priv->stop_on_disconnect;
clock = priv->clock ? gst_object_ref (priv->clock) : NULL;
publish_clock_mode = priv->publish_clock_mode;
+ ttl = priv->max_mcast_ttl;
GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
gst_rtsp_media_set_suspend_mode (media, suspend_mode);
gst_rtsp_media_set_transport_mode (media, transport_mode);
gst_rtsp_media_set_stop_on_disconnect (media, stop_on_disconnect);
gst_rtsp_media_set_publish_clock_mode (media, publish_clock_mode);
+ gst_rtsp_media_set_max_mcast_ttl (media, ttl);
if (clock) {
gst_rtsp_media_set_clock (media, clock);
GST_RTSP_SERVER_API
GstRTSPPublishClockMode gst_rtsp_media_factory_get_publish_clock_mode (GstRTSPMediaFactory * factory);
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_factory_set_max_mcast_ttl (GstRTSPMediaFactory * factory,
+ guint ttl);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_media_factory_get_max_mcast_ttl (GstRTSPMediaFactory * factory);
+
/* creating the media from the factory and a url */
GST_RTSP_SERVER_API
guint buffer_size;
GstRTSPAddressPool *pool;
gchar *multicast_iface;
+ guint max_mcast_ttl;
gboolean blocked;
GstRTSPTransportMode transport_mode;
gboolean stop_on_disconnect;
#define DEFAULT_LATENCY 200
#define DEFAULT_TRANSPORT_MODE GST_RTSP_TRANSPORT_MODE_PLAY
#define DEFAULT_STOP_ON_DISCONNECT TRUE
+#define DEFAULT_MAX_MCAST_TTL 255
#define DEFAULT_DO_RETRANSMISSION FALSE
PROP_TRANSPORT_MODE,
PROP_STOP_ON_DISCONNECT,
PROP_CLOCK,
+ PROP_MAX_MCAST_TTL,
PROP_LAST
};
"Clock to be used by the media pipeline",
GST_TYPE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_MAX_MCAST_TTL,
+ g_param_spec_uint ("max-mcast-ttl", "Maximum multicast ttl",
+ "The maximum time-to-live value of outgoing multicast packets", 1,
+ 255, DEFAULT_MAX_MCAST_TTL,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
priv->stop_on_disconnect = DEFAULT_STOP_ON_DISCONNECT;
priv->publish_clock_mode = GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK;
priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
+ priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
}
static void
case PROP_CLOCK:
g_value_take_object (value, gst_rtsp_media_get_clock (media));
break;
+ case PROP_MAX_MCAST_TTL:
+ g_value_set_uint (value, gst_rtsp_media_get_max_mcast_ttl (media));
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
case PROP_CLOCK:
gst_rtsp_media_set_clock (media, g_value_get_object (value));
break;
+ case PROP_MAX_MCAST_TTL:
+ gst_rtsp_media_set_max_mcast_ttl (media, g_value_get_uint (value));
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
return result;
}
+/**
+ * gst_rtsp_media_set_max_mcast_ttl:
+ * @media: a #GstRTSPMedia
+ * @ttl: the new multicast ttl value
+ *
+ * Set the maximum time-to-live value of outgoing multicast packets.
+ *
+ * Returns: %TRUE if the requested ttl has been set successfully.
+ */
+gboolean
+gst_rtsp_media_set_max_mcast_ttl (GstRTSPMedia * media, guint ttl)
+{
+ GstRTSPMediaPrivate *priv;
+ guint i;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ GST_LOG_OBJECT (media, "set max mcast ttl %u", ttl);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+
+ if (ttl == 0 || ttl > DEFAULT_MAX_MCAST_TTL) {
+ GST_WARNING_OBJECT (media, "The reqested mcast TTL value is not valid.");
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+ priv->max_mcast_ttl = ttl;
+
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+ gst_rtsp_stream_set_max_mcast_ttl (stream, ttl);
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_media_get_max_mcast_ttl:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the the maximum time-to-live value of outgoing multicast packets.
+ *
+ * Returns: the maximum time-to-live value of outgoing multicast packets.
+ */
+guint
+gst_rtsp_media_get_max_mcast_ttl (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ guint res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->max_mcast_ttl;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
static GList *
_find_payload_types (GstRTSPMedia * media)
{
if (priv->pool)
gst_rtsp_stream_set_address_pool (stream, priv->pool);
gst_rtsp_stream_set_multicast_iface (stream, priv->multicast_iface);
+ gst_rtsp_stream_set_max_mcast_ttl (stream, priv->max_mcast_ttl);
gst_rtsp_stream_set_profiles (stream, priv->profiles);
gst_rtsp_stream_set_protocols (stream, priv->protocols);
gst_rtsp_stream_set_retransmission_time (stream, priv->rtx_time);
GST_RTSP_SERVER_API
GstRTSPPublishClockMode gst_rtsp_media_get_publish_clock_mode (GstRTSPMedia * media);
+GST_RTSP_SERVER_API
+gboolean gst_rtsp_media_set_max_mcast_ttl (GstRTSPMedia *media, guint ttl);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_media_get_max_mcast_ttl (GstRTSPMedia *media);
+
/* prepare the media for playback */
GST_RTSP_SERVER_API
GstRTSPAddress *mcast_addr_v6;
gchar *multicast_iface;
+ guint max_mcast_ttl;
/* the caps of the stream */
gulong caps_sig;
#define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
GST_RTSP_LOWER_TRANS_TCP
+#define DEFAULT_MAX_MCAST_TTL 255
enum
{
priv->allowed_protocols = DEFAULT_PROTOCOLS;
priv->configured_protocols = 0;
priv->publish_clock_mode = GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK;
+ priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
g_mutex_init (&priv->lock);
return buffer_size;
}
+/**
+ * gst_rtsp_stream_set_max_mcast_ttl:
+ * @stream: a #GstRTSPStream
+ * @ttl: the new multicast ttl value
+ *
+ * Set the maximum time-to-live value of outgoing multicast packets.
+ *
+ * Returns: %TRUE if the requested ttl has been set successfully.
+ *
+ */
+gboolean
+gst_rtsp_stream_set_max_mcast_ttl (GstRTSPStream * stream, guint ttl)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ g_mutex_lock (&stream->priv->lock);
+ if (ttl == 0 || ttl > DEFAULT_MAX_MCAST_TTL) {
+ GST_WARNING_OBJECT (stream, "The reqested mcast TTL value is not valid.");
+ g_mutex_unlock (&stream->priv->lock);
+ return FALSE;
+ }
+ stream->priv->max_mcast_ttl = ttl;
+ g_mutex_unlock (&stream->priv->lock);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_stream_get_max_mcast_ttl:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the the maximum time-to-live value of outgoing multicast packets.
+ *
+ * Returns: the maximum time-to-live value of outgoing multicast packets.
+ *
+ */
+guint
+gst_rtsp_stream_get_max_mcast_ttl (GstRTSPStream * stream)
+{
+ guint ttl;
+
+ g_mutex_lock (&stream->priv->lock);
+ ttl = stream->priv->max_mcast_ttl;
+ g_mutex_unlock (&stream->priv->lock);
+
+ return ttl;
+}
+
/* executed from streaming thread */
static void
caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
GstRTSPPublishClockMode gst_rtsp_stream_get_publish_clock_mode (GstRTSPStream * stream);
GST_RTSP_SERVER_API
+gboolean gst_rtsp_stream_set_max_mcast_ttl (GstRTSPStream *stream, guint ttl);
+
+GST_RTSP_SERVER_API
+guint gst_rtsp_stream_get_max_mcast_ttl (GstRTSPStream *stream);
+
+GST_RTSP_SERVER_API
gboolean gst_rtsp_stream_complete_stream (GstRTSPStream * stream, const GstRTSPTransport * transport);
GST_RTSP_SERVER_API
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);
session = gst_rtsp_session_pool_find (session_pool, session_hdr_params[0]);
g_strfreev (session_hdr_params);
GST_END_TEST;
-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;
-}
-
-GST_START_TEST (test_client_multicast_invalid_transport_specific)
-{
- GstRTSPClient *client;
- GstRTSPMessage request = { 0, };
- gchar *str;
- GstRTSPSessionPool *session_pool;
- GstRTSPContext ctx = { NULL };
-
- client = setup_multicast_client ();
-
- 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 and multicast, but an invalid ip */
- 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=1;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);
-
-
- /* simple SETUP with a valid URI and multicast, but an invalid prt */
- 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=1;port=6000-6001;");
-
- 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);
-
-
- /* simple SETUP with a valid URI and multicast, but an invalid ttl */
- 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=2;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;
-
GST_START_TEST (test_client_multicast_transport_specific)
{
GstRTSPClient *client;
expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=1;port=5000-5001;mode=\"PLAY\"";
- /* simple SETUP with a valid URI and multicast, but an invalid ip */
+ /* 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_END_TEST;
+static void
+mcast_transport_specific_two_clients (gboolean shared)
+{
+ GstRTSPClient *client, *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;
+
+ 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_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 */
+ client = gst_rtsp_client_new ();
+ gst_rtsp_client_set_session_pool (client, session_pool);
+ gst_rtsp_client_set_mount_points (client, mount_points);
+ gst_rtsp_client_set_thread_pool (client, thread_pool);
+
+ 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_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
+ "ttl=1;port=5000-5001;mode=\"PLAY\"";
+
+ /* 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,
+ expected_transport);
+
+ gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
+ NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &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 (client, test_response_200, NULL, NULL);
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+ gst_rtsp_context_pop_current (&ctx);
+ session_id1 = 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 = "RTP/AVP;multicast;destination=233.252.0.2;"
+ "ttl=1;port=5002-5003;mode=\"PLAY\"";
+
+ /* 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,
+ expected_transport);
+
+ gst_rtsp_client_set_send_func (client2, test_setup_response_200_multicast,
+ 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);
+
+ send_teardown (client2);
+ gst_rtsp_context_pop_current (&ctx2);
+
+ gst_rtsp_context_push_current (&ctx);
+ session_id = session_id1;
+ send_teardown (client);
+ gst_rtsp_context_pop_current (&ctx);
+
+ teardown_client (client);
+ 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 */
+GST_START_TEST
+ (test_client_multicast_transport_specific_two_clients_shared_media) {
+ mcast_transport_specific_two_clients (TRUE);
+}
+
+GST_END_TEST;
+
static Suite *
rtspclient_suite (void)
{
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_invalid_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);
return s;
}
GST_END_TEST;
+GST_START_TEST (test_mcast_ttl)
+{
+ GstRTSPMediaFactory *factory;
+ GstElement *element;
+ GstRTSPMedia *media;
+ GstRTSPUrl *url;
+ GstRTSPStream *stream;
+
+ factory = gst_rtsp_media_factory_new ();
+ gst_rtsp_media_factory_set_shared (factory, TRUE);
+ fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+ &url) == GST_RTSP_OK);
+
+ gst_rtsp_media_factory_set_launch (factory,
+ "( videotestsrc ! rtpvrawpay pt=96 name=pay0 "
+ " audiotestsrc ! audioconvert ! rtpL16pay name=pay1 )");
+
+ /* try to set an invalid ttl and make sure that the default ttl value (255) is
+ * set */
+ gst_rtsp_media_factory_set_max_mcast_ttl (factory, 0);
+ fail_unless (gst_rtsp_media_factory_get_max_mcast_ttl (factory) == 255);
+ gst_rtsp_media_factory_set_max_mcast_ttl (factory, 300);
+ fail_unless (gst_rtsp_media_factory_get_max_mcast_ttl (factory) == 255);
+
+ /* set a valid ttl value */
+ gst_rtsp_media_factory_set_max_mcast_ttl (factory, 3);
+ fail_unless (gst_rtsp_media_factory_get_max_mcast_ttl (factory) == 3);
+
+ element = gst_rtsp_media_factory_create_element (factory, url);
+ fail_unless (GST_IS_BIN (element));
+ fail_if (GST_IS_PIPELINE (element));
+ gst_object_unref (element);
+
+ media = gst_rtsp_media_factory_construct (factory, url);
+ fail_unless (GST_IS_RTSP_MEDIA (media));
+
+ fail_unless (gst_rtsp_media_n_streams (media) == 2);
+ fail_unless (gst_rtsp_media_get_max_mcast_ttl (media) == 3);
+
+ /* verify that the correct ttl value has been propageted to the media
+ * streams */
+ stream = gst_rtsp_media_get_stream (media, 0);
+ fail_unless (stream != NULL);
+ fail_unless (gst_rtsp_stream_get_max_mcast_ttl (stream) == 3);
+
+ stream = gst_rtsp_media_get_stream (media, 1);
+ fail_unless (stream != NULL);
+ fail_unless (gst_rtsp_stream_get_max_mcast_ttl (stream) == 3);
+
+ g_object_unref (media);
+
+ gst_rtsp_url_free (url);
+ g_object_unref (factory);
+}
+
+GST_END_TEST;
+
static Suite *
rtspmediafactory_suite (void)
{
tcase_add_test (tc, test_addresspool);
tcase_add_test (tc, test_permissions);
tcase_add_test (tc, test_reset);
+ tcase_add_test (tc, test_mcast_ttl);
return s;
}
projects / platform / upstream / gstreamer.git / commitdiff