*/
/* 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) {
- /* 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. */
+ GstRTSPAddress *addr = NULL;
- if (!use_client_settings) {
- GstRTSPAddress *addr = NULL;
+ 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 (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 */
+ /* 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_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