Revert "Add new API for setting/getting maximum multicast ttl value"

This reverts commit 7f0ae77e40.

Commits where accidentially squashed together

gst/rtsp-server/rtsp-client.c

@@ -1981,27 +1981,26 @@ default_configure_client_transport (GstRTSPClient * client,
      */
     /* 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,
+    if (!gst_rtsp_stream_allocate_udp_sockets (ctx->stream, G_SOCKET_FAMILY_IPV4, ct,
-            G_SOCKET_FAMILY_IPV4, ct, use_client_settings)
+            use_client_settings) && family == G_SOCKET_FAMILY_IPV4)
-        && family == G_SOCKET_FAMILY_IPV4)
       goto error_allocating_ports;
 
-    if (!gst_rtsp_stream_allocate_udp_sockets (ctx->stream,
+    if (!gst_rtsp_stream_allocate_udp_sockets (ctx->stream, G_SOCKET_FAMILY_IPV6, ct,
-            G_SOCKET_FAMILY_IPV6, ct, use_client_settings)
+            use_client_settings) && family == G_SOCKET_FAMILY_IPV6)
-        && 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
+      GstRTSPAddress *addr = NULL;
-       * 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) {
+      if (use_client_settings) {
-        GstRTSPAddress *addr = NULL;
+        /* 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)
@@ -2010,8 +2009,9 @@ default_configure_client_transport (GstRTSPClient * client,
         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;
 

gst/rtsp-server/rtsp-media-factory.c

@@ -59,7 +59,6 @@ struct _GstRTSPMediaFactoryPrivate
   GstRTSPTransportMode transport_mode;
   gboolean stop_on_disconnect;
   gchar *multicast_iface;
-  guint max_mcast_ttl;
 
   GstClockTime rtx_time;
   guint latency;
@@ -84,7 +83,6 @@ struct _GstRTSPMediaFactoryPrivate
                                         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
@@ -103,7 +101,6 @@ enum
   PROP_TRANSPORT_MODE,
   PROP_STOP_ON_DISCONNECT,
   PROP_CLOCK,
-  PROP_MAX_MCAST_TTL,
   PROP_LAST
 };
 
@@ -225,12 +222,6 @@ gst_rtsp_media_factory_class_init (GstRTSPMediaFactoryClass * klass)
           "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,
@@ -272,7 +263,6 @@ gst_rtsp_media_factory_init (GstRTSPMediaFactory * factory)
   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);
@@ -347,9 +337,6 @@ gst_rtsp_media_factory_get_property (GObject * object, guint propid,
     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);
   }
@@ -400,9 +387,6 @@ gst_rtsp_media_factory_set_property (GObject * object, guint propid,
     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);
   }
@@ -1479,62 +1463,6 @@ gst_rtsp_media_factory_get_publish_clock_mode (GstRTSPMediaFactory * factory)
   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)
 {
@@ -1685,7 +1613,6 @@ default_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
   GstClock *clock;
   gchar *multicast_iface;
   GstRTSPPublishClockMode publish_clock_mode;
-  guint ttl;
 
   /* configure the sharedness */
   GST_RTSP_MEDIA_FACTORY_LOCK (factory);
@@ -1701,7 +1628,6 @@ default_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
   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);
@@ -1716,7 +1642,6 @@ default_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
   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/rtsp-media-factory.h

@@ -239,13 +239,6 @@ void                    gst_rtsp_media_factory_set_publish_clock_mode (GstRTSPMe
 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

gst/rtsp-server/rtsp-media.c

@@ -98,7 +98,6 @@ struct _GstRTSPMediaPrivate
   guint buffer_size;
   GstRTSPAddressPool *pool;
   gchar *multicast_iface;
-  guint max_mcast_ttl;
   gboolean blocked;
   GstRTSPTransportMode transport_mode;
   gboolean stop_on_disconnect;
@@ -159,7 +158,6 @@ struct _GstRTSPMediaPrivate
 #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
 
@@ -182,7 +180,6 @@ enum
   PROP_TRANSPORT_MODE,
   PROP_STOP_ON_DISCONNECT,
   PROP_CLOCK,
-  PROP_MAX_MCAST_TTL,
   PROP_LAST
 };
 
@@ -378,12 +375,6 @@ gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
           "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,
@@ -454,7 +445,6 @@ gst_rtsp_media_init (GstRTSPMedia * media)
   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
@@ -541,9 +531,6 @@ gst_rtsp_media_get_property (GObject * object, guint propid,
     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);
   }
@@ -597,9 +584,6 @@ gst_rtsp_media_set_property (GObject * object, guint propid,
     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);
   }
@@ -1839,70 +1823,6 @@ gst_rtsp_media_get_multicast_iface (GstRTSPMedia * media)
   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)
 {
@@ -2220,7 +2140,6 @@ gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
   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/rtsp-media.h

@@ -314,12 +314,6 @@ void                    gst_rtsp_media_set_publish_clock_mode (GstRTSPMedia * me
 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

gst/rtsp-server/rtsp-stream.c

@@ -145,7 +145,6 @@ struct _GstRTSPStreamPrivate
   GstRTSPAddress *mcast_addr_v6;
 
   gchar *multicast_iface;
-  guint max_mcast_ttl;
 
   /* the caps of the stream */
   gulong caps_sig;
@@ -182,7 +181,6 @@ struct _GstRTSPStreamPrivate
 #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
 {
@@ -282,7 +280,6 @@ gst_rtsp_stream_init (GstRTSPStream * stream)
   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);
 
@@ -1887,54 +1884,6 @@ gst_rtsp_stream_get_buffer_size (GstRTSPStream * stream)
   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)

gst/rtsp-server/rtsp-stream.h

@@ -292,12 +292,6 @@ void                    gst_rtsp_stream_set_publish_clock_mode (GstRTSPStream *
 GST_RTSP_SERVER_API
 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);
 

tests/check/gst/client.c

@@ -525,6 +525,7 @@ test_setup_response_200_multicast (GstRTSPClient * client,
   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);
 
@@ -726,6 +727,117 @@ GST_START_TEST (test_client_multicast_ignore_transport_specific)
 
 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;
@@ -747,7 +859,7 @@ 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\"";
 
-  /* 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);
@@ -919,156 +1031,6 @@ GST_START_TEST (test_client_sdp_with_no_bitrate_tags)
 
 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)
 {
@@ -1084,13 +1046,12 @@ 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;
 }

tests/check/gst/mediafactory.c

@@ -314,63 +314,6 @@ GST_START_TEST (test_reset)
 
 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)
 {
@@ -386,7 +329,6 @@ 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;
 }