X-Git-Url: http://review.tizen.org/git/?a=blobdiff_plain;f=gst%2Frtsp-server%2Frtsp-media.c;h=6b81e254e5dd1bdcb465c02b636869dde684d2cb;hb=376488d8c7d0a92c56065070f9003e699533c3e5;hp=f37258dda55be3bb03dab148296321688909b7fc;hpb=050b16ad8437313a98944b8d9e4d36e1e659991d;p=platform%2Fupstream%2Fgstreamer.git

diff --git a/gst/rtsp-server/rtsp-media.c b/gst/rtsp-server/rtsp-media.c
index f37258d..6b81e25 100644
--- a/gst/rtsp-server/rtsp-media.c
+++ b/gst/rtsp-server/rtsp-media.c
@@ -80,12 +80,15 @@ struct _GstRTSPMediaPrivate
   /* protected by lock */
   GstRTSPPermissions *permissions;
   gboolean shared;
+  gboolean suspend_mode;
   gboolean reusable;
+  GstRTSPProfile profiles;
   GstRTSPLowerTrans protocols;
   gboolean reused;
   gboolean eos_shutdown;
   guint buffer_size;
   GstRTSPAddressPool *pool;
+  gboolean blocked;
 
   GstElement *element;
   GRecMutex state_lock;         /* locking order: state lock, lock */
@@ -121,9 +124,11 @@ struct _GstRTSPMediaPrivate
 };
 
 #define DEFAULT_SHARED          FALSE
+#define DEFAULT_SUSPEND_MODE    GST_RTSP_SUSPEND_MODE_NONE
 #define DEFAULT_REUSABLE        FALSE
-#define DEFAULT_PROTOCOLS       GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_TCP
-//#define DEFAULT_PROTOCOLS      GST_RTSP_LOWER_TRANS_UDP_MCAST
+#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_EOS_SHUTDOWN    FALSE
 #define DEFAULT_BUFFER_SIZE     0x80000
 #define DEFAULT_TIME_PROVIDER   FALSE
@@ -135,7 +140,9 @@ enum
 {
   PROP_0,
   PROP_SHARED,
+  PROP_SUSPEND_MODE,
   PROP_REUSABLE,
+  PROP_PROFILES,
   PROP_PROTOCOLS,
   PROP_EOS_SHUTDOWN,
   PROP_BUFFER_SIZE,
@@ -150,6 +157,7 @@ enum
   SIGNAL_REMOVED_STREAM,
   SIGNAL_PREPARED,
   SIGNAL_UNPREPARED,
+  SIGNAL_TARGET_STATE,
   SIGNAL_NEW_STATE,
   SIGNAL_LAST
 };
@@ -166,16 +174,46 @@ static void gst_rtsp_media_finalize (GObject * obj);
 static gboolean default_handle_message (GstRTSPMedia * media,
     GstMessage * message);
 static void finish_unprepare (GstRTSPMedia * media);
+static gboolean default_prepare (GstRTSPMedia * media, GstRTSPThread * thread);
 static gboolean default_unprepare (GstRTSPMedia * media);
-static gboolean
-default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
-    GstRTSPRangeUnit unit);
+static gboolean default_suspend (GstRTSPMedia * media);
+static gboolean default_unsuspend (GstRTSPMedia * media);
+static gboolean default_convert_range (GstRTSPMedia * media,
+    GstRTSPTimeRange * range, GstRTSPRangeUnit unit);
 static gboolean default_query_position (GstRTSPMedia * media,
     gint64 * position);
 static gboolean default_query_stop (GstRTSPMedia * media, gint64 * stop);
+static GstElement *default_create_rtpbin (GstRTSPMedia * media);
+static gboolean default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
+    GstSDPInfo * info);
+
+static gboolean wait_preroll (GstRTSPMedia * media);
 
 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
 
+#define C_ENUM(v) ((gint) v)
+
+#define GST_TYPE_RTSP_SUSPEND_MODE (gst_rtsp_suspend_mode_get_type())
+GType
+gst_rtsp_suspend_mode_get_type (void)
+{
+  static gsize id = 0;
+  static const GEnumValue values[] = {
+    {C_ENUM (GST_RTSP_SUSPEND_MODE_NONE), "GST_RTSP_SUSPEND_MODE_NONE", "none"},
+    {C_ENUM (GST_RTSP_SUSPEND_MODE_PAUSE), "GST_RTSP_SUSPEND_MODE_PAUSE",
+        "pause"},
+    {C_ENUM (GST_RTSP_SUSPEND_MODE_RESET), "GST_RTSP_SUSPEND_MODE_RESET",
+        "reset"},
+    {0, NULL, NULL}
+  };
+
+  if (g_once_init_enter (&id)) {
+    GType tmp = g_enum_register_static ("GstRTSPSuspendMode", values);
+    g_once_init_leave (&id, tmp);
+  }
+  return (GType) id;
+}
+
 G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
 
 static void
@@ -196,11 +234,21 @@ gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
           "If this media pipeline can be shared", DEFAULT_SHARED,
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
+      g_param_spec_enum ("suspend-mode", "Suspend Mode",
+          "How to suspend the media in PAUSED", GST_TYPE_RTSP_SUSPEND_MODE,
+          DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
   g_object_class_install_property (gobject_class, PROP_REUSABLE,
       g_param_spec_boolean ("reusable", "Reusable",
           "If this media pipeline can be reused after an unprepare",
           DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  g_object_class_install_property (gobject_class, PROP_PROFILES,
+      g_param_spec_flags ("profiles", "Profiles",
+          "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
+          DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
   g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
       g_param_spec_flags ("protocols", "Protocols",
           "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
@@ -240,25 +288,35 @@ gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
   gst_rtsp_media_signals[SIGNAL_PREPARED] =
       g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
       G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
-      g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
+      g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
 
   gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
       g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
       G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
-      g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
+      g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
+
+  gst_rtsp_media_signals[SIGNAL_TARGET_STATE] =
+      g_signal_new ("target-state", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, target_state),
+      NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
 
   gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
       g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
       G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
-      g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 0, G_TYPE_INT);
+      g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
 
   GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
 
   klass->handle_message = default_handle_message;
+  klass->prepare = default_prepare;
   klass->unprepare = default_unprepare;
+  klass->suspend = default_suspend;
+  klass->unsuspend = default_unsuspend;
   klass->convert_range = default_convert_range;
   klass->query_position = default_query_position;
   klass->query_stop = default_query_stop;
+  klass->create_rtpbin = default_create_rtpbin;
+  klass->setup_sdp = default_setup_sdp;
 }
 
 static void
@@ -274,7 +332,9 @@ gst_rtsp_media_init (GstRTSPMedia * media)
   g_rec_mutex_init (&priv->state_lock);
 
   priv->shared = DEFAULT_SHARED;
+  priv->suspend_mode = DEFAULT_SUSPEND_MODE;
   priv->reusable = DEFAULT_REUSABLE;
+  priv->profiles = DEFAULT_PROFILES;
   priv->protocols = DEFAULT_PROTOCOLS;
   priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
   priv->buffer_size = DEFAULT_BUFFER_SIZE;
@@ -292,6 +352,9 @@ gst_rtsp_media_finalize (GObject * obj)
 
   GST_INFO ("finalize media %p", media);
 
+  if (priv->permissions)
+    gst_rtsp_permissions_unref (priv->permissions);
+
   g_ptr_array_unref (priv->streams);
 
   g_list_free_full (priv->dynamic, gst_object_unref);
@@ -323,9 +386,15 @@ gst_rtsp_media_get_property (GObject * object, guint propid,
     case PROP_SHARED:
       g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
       break;
+    case PROP_SUSPEND_MODE:
+      g_value_set_enum (value, gst_rtsp_media_get_suspend_mode (media));
+      break;
     case PROP_REUSABLE:
       g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
       break;
+    case PROP_PROFILES:
+      g_value_set_flags (value, gst_rtsp_media_get_profiles (media));
+      break;
     case PROP_PROTOCOLS:
       g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
       break;
@@ -357,9 +426,15 @@ gst_rtsp_media_set_property (GObject * object, guint propid,
     case PROP_SHARED:
       gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
       break;
+    case PROP_SUSPEND_MODE:
+      gst_rtsp_media_set_suspend_mode (media, g_value_get_enum (value));
+      break;
     case PROP_REUSABLE:
       gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
       break;
+    case PROP_PROFILES:
+      gst_rtsp_media_set_profiles (media, g_value_get_flags (value));
+      break;
     case PROP_PROTOCOLS:
       gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
       break;
@@ -377,12 +452,92 @@ gst_rtsp_media_set_property (GObject * object, guint propid,
   }
 }
 
+typedef struct
+{
+  gint64 position;
+  gboolean ret;
+} DoQueryPositionData;
+
+static void
+do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
+{
+  gint64 tmp;
+
+  if (gst_rtsp_stream_query_position (stream, &tmp)) {
+    data->position = MAX (data->position, tmp);
+    data->ret = TRUE;
+  }
+}
+
+static gboolean
+default_query_position (GstRTSPMedia * media, gint64 * position)
+{
+  GstRTSPMediaPrivate *priv;
+  DoQueryPositionData data;
+
+  priv = media->priv;
+
+  data.position = -1;
+  data.ret = FALSE;
+
+  g_ptr_array_foreach (priv->streams, (GFunc) do_query_position, &data);
+
+  *position = data.position;
+
+  return data.ret;
+}
+
+typedef struct
+{
+  gint64 stop;
+  gboolean ret;
+} DoQueryStopData;
+
+static void
+do_query_stop (GstRTSPStream * stream, DoQueryStopData * data)
+{
+  gint64 tmp;
+
+  if (gst_rtsp_stream_query_stop (stream, &tmp)) {
+    data->stop = MAX (data->stop, tmp);
+    data->ret = TRUE;
+  }
+}
+
+static gboolean
+default_query_stop (GstRTSPMedia * media, gint64 * stop)
+{
+  GstRTSPMediaPrivate *priv;
+  DoQueryStopData data;
+
+  priv = media->priv;
+
+  data.stop = -1;
+  data.ret = FALSE;
+
+  g_ptr_array_foreach (priv->streams, (GFunc) do_query_stop, &data);
+
+  *stop = data.stop;
+
+  return data.ret;
+}
+
+static GstElement *
+default_create_rtpbin (GstRTSPMedia * media)
+{
+  GstElement *rtpbin;
+
+  rtpbin = gst_element_factory_make ("rtpbin", NULL);
+
+  return rtpbin;
+}
+
 /* must be called with state lock */
 static void
 collect_media_stats (GstRTSPMedia * media)
 {
   GstRTSPMediaPrivate *priv = media->priv;
-  gint64 position, stop;
+  gint64 position = 0, stop = -1;
 
   if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
       priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
@@ -460,7 +615,7 @@ collect_media_stats (GstRTSPMedia * media)
  *
  * Ownership is taken of @element.
  *
- * Returns: a new #GstRTSPMedia object.
+ * Returns: (transfer full): a new #GstRTSPMedia object.
  */
 GstRTSPMedia *
 gst_rtsp_media_new (GstElement * element)
@@ -480,7 +635,7 @@ gst_rtsp_media_new (GstElement * element)
  *
  * Get the element that was used when constructing @media.
  *
- * Returns: a #GstElement. Unref after usage.
+ * Returns: (transfer full): a #GstElement. Unref after usage.
  */
 GstElement *
 gst_rtsp_media_get_element (GstRTSPMedia * media)
@@ -529,7 +684,7 @@ gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
 /**
  * gst_rtsp_media_set_permissions:
  * @media: a #GstRTSPMedia
- * @permissions: a #GstRTSPPermissions
+ * @permissions: (transfer none): a #GstRTSPPermissions
  *
  * Set @permissions on @media.
  */
@@ -578,6 +733,66 @@ gst_rtsp_media_get_permissions (GstRTSPMedia * media)
 }
 
 /**
+ * gst_rtsp_media_set_suspend_mode:
+ * @media: a #GstRTSPMedia
+ * @mode: the new #GstRTSPSuspendMode
+ *
+ * Control how @ media will be suspended after the SDP has been generated and
+ * after a PAUSE request has been performed.
+ *
+ * Media must be unprepared when setting the suspend mode.
+ */
+void
+gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
+{
+  GstRTSPMediaPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  priv = media->priv;
+
+  g_rec_mutex_lock (&priv->state_lock);
+  if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
+    goto was_prepared;
+  priv->suspend_mode = mode;
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  return;
+
+  /* ERRORS */
+was_prepared:
+  {
+    GST_WARNING ("media %p was prepared", media);
+    g_rec_mutex_unlock (&priv->state_lock);
+  }
+}
+
+/**
+ * gst_rtsp_media_get_suspend_mode:
+ * @media: a #GstRTSPMedia
+ *
+ * Get how @media will be suspended.
+ *
+ * Returns: #GstRTSPSuspendMode.
+ */
+GstRTSPSuspendMode
+gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv;
+  GstRTSPSuspendMode res;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
+
+  priv = media->priv;
+
+  g_rec_mutex_lock (&priv->state_lock);
+  res = priv->suspend_mode;
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  return res;
+}
+
+/**
  * gst_rtsp_media_set_shared:
  * @media: a #GstRTSPMedia
  * @shared: the new value
@@ -672,6 +887,65 @@ gst_rtsp_media_is_reusable (GstRTSPMedia * media)
   return res;
 }
 
+static void
+do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
+{
+  gst_rtsp_stream_set_profiles (stream, *profiles);
+}
+
+/**
+ * gst_rtsp_media_set_profiles:
+ * @media: a #GstRTSPMedia
+ * @profiles: the new flags
+ *
+ * Configure the allowed lower transport for @media.
+ */
+void
+gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
+{
+  GstRTSPMediaPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  priv->profiles = profiles;
+  g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_profiles:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the allowed profiles of @media.
+ *
+ * Returns: a #GstRTSPProfile
+ */
+GstRTSPProfile
+gst_rtsp_media_get_profiles (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv;
+  GstRTSPProfile res;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  res = priv->profiles;
+  g_mutex_unlock (&priv->lock);
+
+  return res;
+}
+
+static void
+do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
+{
+  gst_rtsp_stream_set_protocols (stream, *protocols);
+}
+
 /**
  * gst_rtsp_media_set_protocols:
  * @media: a #GstRTSPMedia
@@ -690,6 +964,7 @@ gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
 
   g_mutex_lock (&priv->lock);
   priv->protocols = protocols;
+  g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
   g_mutex_unlock (&priv->lock);
 }
 
@@ -866,7 +1141,7 @@ gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
 /**
  * gst_rtsp_media_set_address_pool:
  * @media: a #GstRTSPMedia
- * @pool: a #GstRTSPAddressPool
+ * @pool: (transfer none): a #GstRTSPAddressPool
  *
  * configure @pool to be used as the address pool of @media.
  */
@@ -927,10 +1202,10 @@ gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
  * gst_rtsp_media_collect_streams:
  * @media: a #GstRTSPMedia
  *
- * Find all payloader elements, they should be named pay%d in the
+ * Find all payloader elements, they should be named pay\%d in the
  * element of @media, and create #GstRTSPStreams for them.
  *
- * Collect all dynamic elements, named dynpay%d, and add them to
+ * Collect all dynamic elements, named dynpay\%d, and add them to
  * the list of dynamic elements.
  */
 void
@@ -971,7 +1246,6 @@ gst_rtsp_media_collect_streams (GstRTSPMedia * media)
     name = g_strdup_printf ("dynpay%d", i);
     if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
       /* a stream that will dynamically create pads to provide RTP packets */
-
       GST_INFO ("found dynamic element %d, %p", i, elem);
 
       g_mutex_lock (&priv->lock);
@@ -994,7 +1268,7 @@ gst_rtsp_media_collect_streams (GstRTSPMedia * media)
  * @srcpad should be a pad of an element inside @media->element.
  *
  * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
- *          as @media exists.
+ * as @media exists.
  */
 GstRTSPStream *
 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
@@ -1027,6 +1301,8 @@ gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
   stream = gst_rtsp_stream_new (idx, payloader, srcpad);
   if (priv->pool)
     gst_rtsp_stream_set_address_pool (stream, priv->pool);
+  gst_rtsp_stream_set_profiles (stream, priv->profiles);
+  gst_rtsp_stream_set_protocols (stream, priv->protocols);
 
   g_ptr_array_add (priv->streams, stream);
   g_mutex_unlock (&priv->lock);
@@ -1093,8 +1369,8 @@ gst_rtsp_media_n_streams (GstRTSPMedia * media)
  *
  * Retrieve the stream with index @idx from @media.
  *
- * Returns: (transfer none): the #GstRTSPStream at index @idx or %NULL when a stream with
- * that index did not exist.
+ * Returns: (nullable) (transfer none): the #GstRTSPStream at index
+ * @idx or %NULL when a stream with that index did not exist.
  */
 GstRTSPStream *
 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
@@ -1123,8 +1399,9 @@ gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
  *
  * Find a stream in @media with @control as the control uri.
  *
- * Returns: (transfer none): the #GstRTSPStream with control uri @control
- * or %NULL when a stream with that control did not exist.
+ * Returns: (nullable) (transfer none): the #GstRTSPStream with
+ * control uri @control or %NULL when a stream with that control did
+ * not exist.
  */
 GstRTSPStream *
 gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
@@ -1155,6 +1432,14 @@ gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
   return res;
 }
 
+/* called with state-lock */
+static gboolean
+default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
+    GstRTSPRangeUnit unit)
+{
+  return gst_rtsp_range_convert_units (range, unit);
+}
+
 /**
  * gst_rtsp_media_get_range_string:
  * @media: a #GstRTSPMedia
@@ -1164,7 +1449,7 @@ gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
  * Get the current range as a string. @media must be prepared with
  * gst_rtsp_media_prepare ().
  *
- * Returns: The range as a string, g_free() after usage.
+ * Returns: (transfer full): The range as a string, g_free() after usage.
  */
 gchar *
 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
@@ -1182,7 +1467,8 @@ gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
   priv = media->priv;
 
   g_rec_mutex_lock (&priv->state_lock);
-  if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
+  if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
+      priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
     goto not_prepared;
 
   g_mutex_lock (&priv->lock);
@@ -1221,10 +1507,72 @@ conversion_failed:
   }
 }
 
+static void
+stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
+{
+  gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
+}
+
+static void
+media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+
+  GST_DEBUG ("media %p set blocked %d", media, blocked);
+  priv->blocked = blocked;
+  g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
+}
+
+static void
+gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  priv->status = status;
+  GST_DEBUG ("setting new status to %d", status);
+  g_cond_broadcast (&priv->cond);
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_status:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the status of @media. When @media is busy preparing, this function waits
+ * until @media is prepared or in error.
+ *
+ * Returns: the status of @media.
+ */
+GstRTSPMediaStatus
+gst_rtsp_media_get_status (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  GstRTSPMediaStatus result;
+  gint64 end_time;
+
+  g_mutex_lock (&priv->lock);
+  end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
+  /* while we are preparing, wait */
+  while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
+    GST_DEBUG ("waiting for status change");
+    if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
+      GST_DEBUG ("timeout, assuming error status");
+      priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
+    }
+  }
+  /* could be success or error */
+  result = priv->status;
+  GST_DEBUG ("got status %d", result);
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
 /**
  * gst_rtsp_media_seek:
  * @media: a #GstRTSPMedia
- * @range: a #GstRTSPTimeRange
+ * @range: (transfer none): a #GstRTSPTimeRange
  *
  * Seek the pipeline of @media to @range. @media must be prepared with
  * gst_rtsp_media_prepare().
@@ -1236,10 +1584,10 @@ gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
 {
   GstRTSPMediaClass *klass;
   GstRTSPMediaPrivate *priv;
-  GstSeekFlags flags;
   gboolean res;
   GstClockTime start, stop;
   GstSeekType start_type, stop_type;
+  GstQuery *query;
 
   klass = GST_RTSP_MEDIA_GET_CLASS (media);
 
@@ -1253,13 +1601,21 @@ gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
   if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
     goto not_prepared;
 
+  /* Update the seekable state of the pipeline in case it changed */
+  query = gst_query_new_seeking (GST_FORMAT_TIME);
+  if (gst_element_query (priv->pipeline, query)) {
+    GstFormat format;
+    gboolean seekable;
+    gint64 start, end;
+
+    gst_query_parse_seeking (query, &format, &seekable, &start, &end);
+    priv->seekable = seekable;
+  }
+  gst_query_unref (query);
+
   if (!priv->seekable)
     goto not_seekable;
 
-  /* depends on the current playing state of the pipeline. We might need to
-   * queue this until we get EOS. */
-  flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_KEY_UNIT;
-
   start_type = stop_type = GST_SEEK_TYPE_NONE;
 
   if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
@@ -1271,9 +1627,7 @@ gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
   GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
       GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
 
-  if (priv->range_start == start)
-    start = GST_CLOCK_TIME_NONE;
-  else if (start != GST_CLOCK_TIME_NONE)
+  if (start != GST_CLOCK_TIME_NONE)
     start_type = GST_SEEK_TYPE_SET;
 
   if (priv->range_stop == stop)
@@ -1282,18 +1636,58 @@ gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
     stop_type = GST_SEEK_TYPE_SET;
 
   if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
+    GstSeekFlags flags;
+
     GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
         GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
 
+    gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
+    if (priv->blocked)
+      media_streams_set_blocked (media, TRUE);
+
+    /* depends on the current playing state of the pipeline. We might need to
+     * queue this until we get EOS. */
+    flags = GST_SEEK_FLAG_FLUSH;
+
+    /* if range start was not supplied we must continue from current position.
+     * but since we're doing a flushing seek, let us query the current position
+     * so we end up at exactly the same position after the seek. */
+    if (range->min.type == GST_RTSP_TIME_END) { /* Yepp, that's right! */
+      gint64 position;
+      gboolean ret = FALSE;
+
+      if (klass->query_position)
+        ret = klass->query_position (media, &position);
+
+      if (!ret) {
+        GST_WARNING ("position query failed");
+      } else {
+        GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
+            GST_TIME_ARGS (position));
+        start = position;
+        start_type = GST_SEEK_TYPE_SET;
+        flags |= GST_SEEK_FLAG_ACCURATE;
+      }
+    } else {
+      /* only set keyframe flag when modifying start */
+      if (start_type != GST_SEEK_TYPE_NONE)
+        flags |= GST_SEEK_FLAG_KEY_UNIT;
+    }
+
+    /* FIXME, we only do forwards playback, no trick modes yet */
     res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
         flags, start_type, start, stop_type, stop);
 
     /* and block for the seek to complete */
     GST_INFO ("done seeking %d", res);
-    gst_element_get_state (priv->pipeline, NULL, NULL, -1);
-    GST_INFO ("prerolled again");
+    g_rec_mutex_unlock (&priv->state_lock);
 
-    collect_media_stats (media);
+    /* wait until pipeline is prerolled again, this will also collect stats */
+    if (!wait_preroll (media))
+      goto preroll_failed;
+
+    g_rec_mutex_lock (&priv->state_lock);
+    GST_INFO ("prerolled again");
   } else {
     GST_INFO ("no seek needed");
     res = TRUE;
@@ -1313,7 +1707,7 @@ not_seekable:
   {
     g_rec_mutex_unlock (&priv->state_lock);
     GST_INFO ("pipeline is not seekable");
-    return TRUE;
+    return FALSE;
   }
 not_supported:
   {
@@ -1321,52 +1715,62 @@ not_supported:
     GST_WARNING ("conversion to npt not supported");
     return FALSE;
   }
+preroll_failed:
+  {
+    GST_WARNING ("failed to preroll after seek");
+    return FALSE;
+  }
 }
 
 static void
-gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
+stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
 {
-  GstRTSPMediaPrivate *priv = media->priv;
-
-  g_mutex_lock (&priv->lock);
-  priv->status = status;
-  GST_DEBUG ("setting new status to %d", status);
-  g_cond_broadcast (&priv->cond);
-  g_mutex_unlock (&priv->lock);
+  *blocked &= gst_rtsp_stream_is_blocking (stream);
 }
 
-/**
- * gst_rtsp_media_get_status:
- * @media: a #GstRTSPMedia
- *
- * Get the status of @media. When @media is busy preparing, this function waits
- * until @media is prepared or in error.
- *
- * Returns: the status of @media.
- */
-GstRTSPMediaStatus
-gst_rtsp_media_get_status (GstRTSPMedia * media)
+static gboolean
+media_streams_blocking (GstRTSPMedia * media)
+{
+  gboolean blocking = TRUE;
+
+  g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
+      &blocking);
+
+  return blocking;
+}
+
+static GstStateChangeReturn
+set_state (GstRTSPMedia * media, GstState state)
 {
   GstRTSPMediaPrivate *priv = media->priv;
-  GstRTSPMediaStatus result;
-  gint64 end_time;
+  GstStateChangeReturn ret;
 
-  g_mutex_lock (&priv->lock);
-  end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
-  /* while we are preparing, wait */
-  while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
-    GST_DEBUG ("waiting for status change");
-    if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
-      GST_DEBUG ("timeout, assuming error status");
-      priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
-    }
-  }
-  /* could be success or error */
-  result = priv->status;
-  GST_DEBUG ("got status %d", result);
-  g_mutex_unlock (&priv->lock);
+  GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
+      media);
+  ret = gst_element_set_state (priv->pipeline, state);
 
-  return result;
+  return ret;
+}
+
+static GstStateChangeReturn
+set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  GstStateChangeReturn ret;
+
+  GST_INFO ("set target state to %s for media %p",
+      gst_element_state_get_name (state), media);
+  priv->target_state = state;
+
+  g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
+      priv->target_state, NULL);
+
+  if (do_state)
+    ret = set_state (media, state);
+  else
+    ret = GST_STATE_CHANGE_SUCCESS;
+
+  return ret;
 }
 
 /* called with state-lock */
@@ -1397,7 +1801,7 @@ default_handle_message (GstRTSPMedia * media, GstMessage * message)
         /* if the desired state is playing, go back */
         if (priv->target_state == GST_STATE_PLAYING) {
           GST_INFO ("Buffering done, setting pipeline to PLAYING");
-          gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
+          set_state (media, GST_STATE_PLAYING);
         } else {
           GST_INFO ("Buffering done");
         }
@@ -1407,7 +1811,7 @@ default_handle_message (GstRTSPMedia * media, GstMessage * message)
           if (priv->target_state == GST_STATE_PLAYING) {
             /* we were not buffering but PLAYING, PAUSE  the pipeline. */
             GST_INFO ("Buffering, setting pipeline to PAUSED ...");
-            gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
+            set_state (media, GST_STATE_PAUSED);
           } else {
             GST_INFO ("Buffering ...");
           }
@@ -1446,7 +1850,22 @@ default_handle_message (GstRTSPMedia * media, GstMessage * message)
       break;
     }
     case GST_MESSAGE_ELEMENT:
+    {
+      const GstStructure *s;
+
+      s = gst_message_get_structure (message);
+      if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
+        GST_DEBUG ("media received blocking message");
+        if (priv->blocked && media_streams_blocking (media)) {
+          GST_DEBUG ("media is blocking");
+          collect_media_stats (media);
+
+          if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
+            gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+        }
+      }
       break;
+    }
     case GST_MESSAGE_STREAM_STATUS:
       break;
     case GST_MESSAGE_ASYNC_DONE:
@@ -1505,21 +1924,62 @@ watch_destroyed (GstRTSPMedia * media)
   g_object_unref (media);
 }
 
+static GstElement *
+find_payload_element (GstElement * payloader)
+{
+  GstElement *pay = NULL;
+
+  if (GST_IS_BIN (payloader)) {
+    GstIterator *iter;
+    GValue item = { 0 };
+
+    iter = gst_bin_iterate_recurse (GST_BIN (payloader));
+    while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
+      GstElement *element = (GstElement *) g_value_get_object (&item);
+      GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
+      const gchar *klass;
+
+      klass =
+          gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
+      if (klass == NULL)
+        continue;
+
+      if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
+        pay = gst_object_ref (element);
+        g_value_unset (&item);
+        break;
+      }
+      g_value_unset (&item);
+    }
+    gst_iterator_free (iter);
+  } else {
+    pay = g_object_ref (payloader);
+  }
+
+  return pay;
+}
+
 /* called from streaming threads */
 static void
 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
 {
   GstRTSPMediaPrivate *priv = media->priv;
   GstRTSPStream *stream;
+  GstElement *pay;
 
-  /* FIXME, element is likely not a payloader, find the payloader here */
-  stream = gst_rtsp_media_create_stream (media, element, pad);
-
-  g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
+  /* find the real payload element */
+  pay = find_payload_element (element);
+  stream = gst_rtsp_media_create_stream (media, pay, pad);
+  gst_object_unref (pay);
 
   GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
 
   g_rec_mutex_lock (&priv->state_lock);
+  if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
+    goto not_preparing;
+
+  g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
+
   /* we will be adding elements below that will cause ASYNC_DONE to be
    * posted in the bus. We want to ignore those messages until the
    * pipeline really prerolled. */
@@ -1527,11 +1987,24 @@ pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
 
   /* join the element in the PAUSED state because this callback is
    * called from the streaming thread and it is PAUSED */
-  gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
-      priv->rtpbin, GST_STATE_PAUSED);
+  if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
+          priv->rtpbin, GST_STATE_PAUSED)) {
+    GST_WARNING ("failed to join bin element");
+  }
 
   priv->adding = FALSE;
   g_rec_mutex_unlock (&priv->state_lock);
+
+  return;
+
+  /* ERRORS */
+not_preparing:
+  {
+    gst_rtsp_media_remove_stream (media, stream);
+    g_rec_mutex_unlock (&priv->state_lock);
+    GST_INFO ("ignore pad because we are not preparing");
+    return;
+  }
 }
 
 static void
@@ -1591,10 +2064,75 @@ struct _DynPaySignalHandlers
 };
 
 static gboolean
-start_prepare (GstRTSPMedia * media)
+start_preroll (GstRTSPMedia * media)
 {
   GstRTSPMediaPrivate *priv = media->priv;
   GstStateChangeReturn ret;
+
+  GST_INFO ("setting pipeline to PAUSED for media %p", media);
+  /* first go to PAUSED */
+  ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
+
+  switch (ret) {
+    case GST_STATE_CHANGE_SUCCESS:
+      GST_INFO ("SUCCESS state change for media %p", media);
+      priv->seekable = TRUE;
+      break;
+    case GST_STATE_CHANGE_ASYNC:
+      GST_INFO ("ASYNC state change for media %p", media);
+      priv->seekable = TRUE;
+      break;
+    case GST_STATE_CHANGE_NO_PREROLL:
+      /* we need to go to PLAYING */
+      GST_INFO ("NO_PREROLL state change: live media %p", media);
+      /* FIXME we disable seeking for live streams for now. We should perform a
+       * seeking query in preroll instead */
+      priv->seekable = FALSE;
+      priv->is_live = TRUE;
+      /* start blocked  to make sure nothing goes to the sink */
+      media_streams_set_blocked (media, TRUE);
+      ret = set_state (media, GST_STATE_PLAYING);
+      if (ret == GST_STATE_CHANGE_FAILURE)
+        goto state_failed;
+      break;
+    case GST_STATE_CHANGE_FAILURE:
+      goto state_failed;
+  }
+
+  return TRUE;
+
+state_failed:
+  {
+    GST_WARNING ("failed to preroll pipeline");
+    return FALSE;
+  }
+}
+
+static gboolean
+wait_preroll (GstRTSPMedia * media)
+{
+  GstRTSPMediaStatus status;
+
+  GST_DEBUG ("wait to preroll pipeline");
+
+  /* wait until pipeline is prerolled */
+  status = gst_rtsp_media_get_status (media);
+  if (status == GST_RTSP_MEDIA_STATUS_ERROR)
+    goto preroll_failed;
+
+  return TRUE;
+
+preroll_failed:
+  {
+    GST_WARNING ("failed to preroll pipeline");
+    return FALSE;
+  }
+}
+
+static gboolean
+start_prepare (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
   guint i;
   GList *walk;
 
@@ -1605,8 +2143,10 @@ start_prepare (GstRTSPMedia * media)
 
     stream = g_ptr_array_index (priv->streams, i);
 
-    gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
-        priv->rtpbin, GST_STATE_NULL);
+    if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
+            priv->rtpbin, GST_STATE_NULL)) {
+      goto join_bin_failed;
+    }
   }
 
   for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
@@ -1630,38 +2170,18 @@ start_prepare (GstRTSPMedia * media)
     gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink);
   }
 
-  GST_INFO ("setting pipeline to PAUSED for media %p", media);
-  /* first go to PAUSED */
-  ret = gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
-  priv->target_state = GST_STATE_PAUSED;
-
-  switch (ret) {
-    case GST_STATE_CHANGE_SUCCESS:
-      GST_INFO ("SUCCESS state change for media %p", media);
-      priv->seekable = TRUE;
-      break;
-    case GST_STATE_CHANGE_ASYNC:
-      GST_INFO ("ASYNC state change for media %p", media);
-      priv->seekable = TRUE;
-      break;
-    case GST_STATE_CHANGE_NO_PREROLL:
-      /* we need to go to PLAYING */
-      GST_INFO ("NO_PREROLL state change: live media %p", media);
-      /* FIXME we disable seeking for live streams for now. We should perform a
-       * seeking query in preroll instead */
-      priv->seekable = FALSE;
-      priv->is_live = TRUE;
-      ret = gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
-      if (ret == GST_STATE_CHANGE_FAILURE)
-        goto state_failed;
-      break;
-    case GST_STATE_CHANGE_FAILURE:
-      goto state_failed;
-  }
+  if (!start_preroll (media))
+    goto preroll_failed;
 
   return FALSE;
 
-state_failed:
+join_bin_failed:
+  {
+    GST_WARNING ("failed to join bin element");
+    gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+    return FALSE;
+  }
+preroll_failed:
   {
     GST_WARNING ("failed to preroll pipeline");
     gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
@@ -1669,10 +2189,82 @@ state_failed:
   }
 }
 
+static gboolean
+default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
+{
+  GstRTSPMediaPrivate *priv;
+  GstRTSPMediaClass *klass;
+  GstBus *bus;
+  GMainContext *context;
+  GSource *source;
+
+  priv = media->priv;
+
+  klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+  if (!klass->create_rtpbin)
+    goto no_create_rtpbin;
+
+  priv->rtpbin = klass->create_rtpbin (media);
+  if (priv->rtpbin != NULL) {
+    gboolean success = TRUE;
+
+    if (klass->setup_rtpbin)
+      success = klass->setup_rtpbin (media, priv->rtpbin);
+
+    if (success == FALSE) {
+      gst_object_unref (priv->rtpbin);
+      priv->rtpbin = NULL;
+    }
+  }
+  if (priv->rtpbin == NULL)
+    goto no_rtpbin;
+
+  priv->thread = thread;
+  context = (thread != NULL) ? (thread->context) : NULL;
+
+  bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
+
+  /* add the pipeline bus to our custom mainloop */
+  priv->source = gst_bus_create_watch (bus);
+  gst_object_unref (bus);
+
+  g_source_set_callback (priv->source, (GSourceFunc) bus_message,
+      g_object_ref (media), (GDestroyNotify) watch_destroyed);
+
+  priv->id = g_source_attach (priv->source, context);
+
+  /* add stuff to the bin */
+  gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
+
+  /* do remainder in context */
+  source = g_idle_source_new ();
+  g_source_set_callback (source, (GSourceFunc) start_prepare, media, NULL);
+  g_source_attach (source, context);
+  g_source_unref (source);
+
+  return TRUE;
+
+  /* ERRORS */
+no_create_rtpbin:
+  {
+    GST_ERROR ("no create_rtpbin function");
+    g_critical ("no create_rtpbin vmethod function set");
+    return FALSE;
+  }
+no_rtpbin:
+  {
+    GST_WARNING ("no rtpbin element");
+    g_warning ("failed to create element 'rtpbin', check your installation");
+    return FALSE;
+  }
+}
+
 /**
  * gst_rtsp_media_prepare:
  * @media: a #GstRTSPMedia
- * @thread: a #GstRTSPThread to run the bus handler or %NULL
+ * @thread: (transfer full) (allow-none): a #GstRTSPThread to run the
+ *   bus handler or %NULL
  *
  * Prepare @media for streaming. This function will create the objects
  * to manage the streaming. A pipeline must have been set on @media with
@@ -1687,23 +2279,21 @@ gboolean
 gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
 {
   GstRTSPMediaPrivate *priv;
-  GstRTSPMediaStatus status;
-  GstBus *bus;
-  GSource *source;
+  GstRTSPMediaClass *klass;
 
   g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
-  g_return_val_if_fail (GST_IS_RTSP_THREAD (thread), FALSE);
 
   priv = media->priv;
 
   g_rec_mutex_lock (&priv->state_lock);
   priv->prepare_count++;
 
-  if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
+  if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
+      priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
     goto was_prepared;
 
   if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
-    goto wait_status;
+    goto is_preparing;
 
   if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
     goto not_unprepared;
@@ -1711,48 +2301,29 @@ gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
   if (!priv->reusable && priv->reused)
     goto is_reused;
 
-  priv->rtpbin = gst_element_factory_make ("rtpbin", NULL);
-  if (priv->rtpbin == NULL)
-    goto no_rtpbin;
-
   GST_INFO ("preparing media %p", media);
 
   /* reset some variables */
   priv->is_live = FALSE;
   priv->seekable = FALSE;
   priv->buffering = FALSE;
-  priv->thread = thread;
-  /* we're preparing now */
-  priv->status = GST_RTSP_MEDIA_STATUS_PREPARING;
-
-  bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
-
-  /* add the pipeline bus to our custom mainloop */
-  priv->source = gst_bus_create_watch (bus);
-  gst_object_unref (bus);
-
-  g_source_set_callback (priv->source, (GSourceFunc) bus_message,
-      g_object_ref (media), (GDestroyNotify) watch_destroyed);
-
-  priv->id = g_source_attach (priv->source, thread->context);
 
-  /* add stuff to the bin */
-  gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
+  /* we're preparing now */
+  gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
 
-  /* do remainder in context */
-  source = g_idle_source_new ();
-  g_source_set_callback (source, (GSourceFunc) start_prepare, media, NULL);
-  g_source_attach (source, thread->context);
-  g_source_unref (source);
+  klass = GST_RTSP_MEDIA_GET_CLASS (media);
+  if (klass->prepare) {
+    if (!klass->prepare (media, thread))
+      goto prepare_failed;
+  }
 
 wait_status:
   g_rec_mutex_unlock (&priv->state_lock);
 
   /* now wait for all pads to be prerolled, FIXME, we should somehow be
    * able to do this async so that we don't block the server thread. */
-  status = gst_rtsp_media_get_status (media);
-  if (status == GST_RTSP_MEDIA_STATUS_ERROR)
-    goto state_failed;
+  if (!wait_preroll (media))
+    goto preroll_failed;
 
   g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
 
@@ -1761,15 +2332,28 @@ wait_status:
   return TRUE;
 
   /* OK */
+is_preparing:
+  {
+    /* we are not going to use the giving thread, so stop it. */
+    if (thread)
+      gst_rtsp_thread_stop (thread);
+    goto wait_status;
+  }
 was_prepared:
   {
     GST_LOG ("media %p was prepared", media);
+    /* we are not going to use the giving thread, so stop it. */
+    if (thread)
+      gst_rtsp_thread_stop (thread);
     g_rec_mutex_unlock (&priv->state_lock);
     return TRUE;
   }
   /* ERRORS */
 not_unprepared:
   {
+    /* we are not going to use the giving thread, so stop it. */
+    if (thread)
+      gst_rtsp_thread_stop (thread);
     GST_WARNING ("media %p was not unprepared", media);
     priv->prepare_count--;
     g_rec_mutex_unlock (&priv->state_lock);
@@ -1777,20 +2361,25 @@ not_unprepared:
   }
 is_reused:
   {
+    /* we are not going to use the giving thread, so stop it. */
+    if (thread)
+      gst_rtsp_thread_stop (thread);
     priv->prepare_count--;
     g_rec_mutex_unlock (&priv->state_lock);
     GST_WARNING ("can not reuse media %p", media);
     return FALSE;
   }
-no_rtpbin:
+prepare_failed:
   {
+    /* we are not going to use the giving thread, so stop it. */
+    if (thread)
+      gst_rtsp_thread_stop (thread);
     priv->prepare_count--;
     g_rec_mutex_unlock (&priv->state_lock);
-    GST_WARNING ("no rtpbin element");
-    g_warning ("failed to create element 'rtpbin', check your installation");
+    GST_ERROR ("failed to prepare media");
     return FALSE;
   }
-state_failed:
+preroll_failed:
   {
     GST_WARNING ("failed to preroll pipeline");
     gst_rtsp_media_unprepare (media);
@@ -1808,7 +2397,11 @@ finish_unprepare (GstRTSPMedia * media)
 
   GST_DEBUG ("shutting down");
 
-  gst_element_set_state (priv->pipeline, GST_STATE_NULL);
+  /* release the lock on shutdown, otherwise pad_added_cb might try to
+   * acquire the lock and then we deadlock */
+  g_rec_mutex_unlock (&priv->state_lock);
+  set_state (media, GST_STATE_NULL);
+  g_rec_mutex_lock (&priv->state_lock);
   remove_fakesink (priv);
 
   for (i = 0; i < priv->streams->len; i++) {
@@ -1847,7 +2440,7 @@ finish_unprepare (GstRTSPMedia * media)
   priv->nettime = NULL;
 
   priv->reused = TRUE;
-  priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARED;
+  gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
 
   /* when the media is not reusable, this will effectively unref the media and
    * recreate it */
@@ -1877,8 +2470,7 @@ default_unprepare (GstRTSPMedia * media)
     gst_element_send_event (priv->pipeline, gst_event_new_eos ());
     /* we need to go to playing again for the EOS to propagate, normally in this
      * state, nothing is receiving data from us anymore so this is ok. */
-    gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
-    priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARING;
+    set_state (media, GST_STATE_PLAYING);
   } else {
     finish_unprepare (media);
   }
@@ -1914,9 +2506,13 @@ gst_rtsp_media_unprepare (GstRTSPMedia * media)
     goto is_busy;
 
   GST_INFO ("unprepare media %p", media);
-  priv->target_state = GST_STATE_NULL;
+  if (priv->blocked)
+    media_streams_set_blocked (media, FALSE);
+  set_target_state (media, GST_STATE_NULL, FALSE);
   success = TRUE;
 
+  gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
+
   if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
     GstRTSPMediaClass *klass;
 
@@ -1963,7 +2559,7 @@ get_clock_unlocked (GstRTSPMedia * media)
  *
  * @media must be prepared before this method returns a valid clock object.
  *
- * Returns: the #GstClock used by @media. unref after usage.
+ * Returns: (transfer full): the #GstClock used by @media. unref after usage.
  */
 GstClock *
 gst_rtsp_media_get_clock (GstRTSPMedia * media)
@@ -2023,13 +2619,13 @@ not_prepared:
 /**
  * gst_rtsp_media_get_time_provider:
  * @media: a #GstRTSPMedia
- * @address: an address or NULL
+ * @address: (allow-none): an address or %NULL
  * @port: a port or 0
  *
  * Get the #GstNetTimeProvider for the clock used by @media. The time provider
  * will listen on @address and @port for client time requests.
  *
- * Returns: the #GstNetTimeProvider of @media.
+ * Returns: (transfer full): the #GstNetTimeProvider of @media.
  */
 GstNetTimeProvider *
 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
@@ -2063,8 +2659,261 @@ gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
   return provider;
 }
 
-void
-gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
+static gboolean
+default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
+{
+  return gst_rtsp_sdp_from_media (sdp, info, media);
+}
+
+/**
+ * gst_rtsp_media_setup_sdp:
+ * @media: a #GstRTSPMedia
+ * @sdp: (transfer none): a #GstSDPMessage
+ * @info: (transfer none): a #GstSDPInfo
+ *
+ * Add @media specific info to @sdp. @info is used to configure the connection
+ * information in the SDP.
+ *
+ * Returns: TRUE on success.
+ */
+gboolean
+gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
+    GstSDPInfo * info)
+{
+  GstRTSPMediaPrivate *priv;
+  GstRTSPMediaClass *klass;
+  gboolean res;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+  g_return_val_if_fail (sdp != NULL, FALSE);
+  g_return_val_if_fail (info != NULL, FALSE);
+
+  priv = media->priv;
+
+  g_rec_mutex_lock (&priv->state_lock);
+
+  klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+  if (!klass->setup_sdp)
+    goto no_setup_sdp;
+
+  res = klass->setup_sdp (media, sdp, info);
+
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  return res;
+
+  /* ERRORS */
+no_setup_sdp:
+  {
+    g_rec_mutex_unlock (&priv->state_lock);
+    GST_ERROR ("no setup_sdp function");
+    g_critical ("no setup_sdp vmethod function set");
+    return FALSE;
+  }
+}
+
+static void
+do_set_seqnum (GstRTSPStream * stream)
+{
+  guint16 seq_num;
+  seq_num = gst_rtsp_stream_get_current_seqnum (stream);
+  gst_rtsp_stream_set_seqnum_offset (stream, seq_num + 1);
+}
+
+/* call with state_lock */
+gboolean
+default_suspend (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  GstStateChangeReturn ret;
+
+  switch (priv->suspend_mode) {
+    case GST_RTSP_SUSPEND_MODE_NONE:
+      GST_DEBUG ("media %p no suspend", media);
+      break;
+    case GST_RTSP_SUSPEND_MODE_PAUSE:
+      GST_DEBUG ("media %p suspend to PAUSED", media);
+      ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
+      if (ret == GST_STATE_CHANGE_FAILURE)
+        goto state_failed;
+      break;
+    case GST_RTSP_SUSPEND_MODE_RESET:
+      GST_DEBUG ("media %p suspend to NULL", media);
+      ret = set_target_state (media, GST_STATE_NULL, TRUE);
+      if (ret == GST_STATE_CHANGE_FAILURE)
+        goto state_failed;
+      /* Because payloader needs to set the sequence number as
+       * monotonic, we need to preserve the sequence number
+       * after pause. (otherwise going from pause to play,  which
+       * is actually from NULL to PLAY will create a new sequence
+       * number. */
+      g_ptr_array_foreach (priv->streams, (GFunc) do_set_seqnum, NULL);
+      break;
+    default:
+      break;
+  }
+
+  /* let the streams do the state changes freely, if any */
+  media_streams_set_blocked (media, FALSE);
+
+  return TRUE;
+
+  /* ERRORS */
+state_failed:
+  {
+    GST_WARNING ("failed changing pipeline's state for media %p", media);
+    return FALSE;
+  }
+}
+
+/**
+ * gst_rtsp_media_suspend:
+ * @media: a #GstRTSPMedia
+ *
+ * Suspend @media. The state of the pipeline managed by @media is set to
+ * GST_STATE_NULL but all streams are kept. @media can be prepared again
+ * with gst_rtsp_media_unsuspend()
+ *
+ * @media must be prepared with gst_rtsp_media_prepare();
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_suspend (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  GstRTSPMediaClass *klass;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+  GST_FIXME ("suspend for dynamic pipelines needs fixing");
+
+  g_rec_mutex_lock (&priv->state_lock);
+  if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
+    goto not_prepared;
+
+  /* don't attempt to suspend when something is busy */
+  if (priv->n_active > 0)
+    goto done;
+
+  klass = GST_RTSP_MEDIA_GET_CLASS (media);
+  if (klass->suspend) {
+    if (!klass->suspend (media))
+      goto suspend_failed;
+  }
+
+  gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
+done:
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  return TRUE;
+
+  /* ERRORS */
+not_prepared:
+  {
+    g_rec_mutex_unlock (&priv->state_lock);
+    GST_WARNING ("media %p was not prepared", media);
+    return FALSE;
+  }
+suspend_failed:
+  {
+    g_rec_mutex_unlock (&priv->state_lock);
+    gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+    GST_WARNING ("failed to suspend media %p", media);
+    return FALSE;
+  }
+}
+
+/* call with state_lock */
+gboolean
+default_unsuspend (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+
+  switch (priv->suspend_mode) {
+    case GST_RTSP_SUSPEND_MODE_NONE:
+      gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+      break;
+    case GST_RTSP_SUSPEND_MODE_PAUSE:
+      gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+      break;
+    case GST_RTSP_SUSPEND_MODE_RESET:
+    {
+      gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
+      if (!start_preroll (media))
+        goto start_failed;
+      g_rec_mutex_unlock (&priv->state_lock);
+
+      if (!wait_preroll (media))
+        goto preroll_failed;
+
+      g_rec_mutex_lock (&priv->state_lock);
+    }
+    default:
+      break;
+  }
+
+  return TRUE;
+
+  /* ERRORS */
+start_failed:
+  {
+    GST_WARNING ("failed to preroll pipeline");
+    return FALSE;
+  }
+preroll_failed:
+  {
+    GST_WARNING ("failed to preroll pipeline");
+    return FALSE;
+  }
+}
+
+/**
+ * gst_rtsp_media_unsuspend:
+ * @media: a #GstRTSPMedia
+ *
+ * Unsuspend @media if it was in a suspended state. This method does nothing
+ * when the media was not in the suspended state.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_unsuspend (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  GstRTSPMediaClass *klass;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+  g_rec_mutex_lock (&priv->state_lock);
+  if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
+    goto done;
+
+  klass = GST_RTSP_MEDIA_GET_CLASS (media);
+  if (klass->unsuspend) {
+    if (!klass->unsuspend (media))
+      goto unsuspend_failed;
+  }
+
+done:
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  return TRUE;
+
+  /* ERRORS */
+unsuspend_failed:
+  {
+    g_rec_mutex_unlock (&priv->state_lock);
+    GST_WARNING ("failed to unsuspend media %p", media);
+    gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+    return FALSE;
+  }
+}
+
+/* must be called with state-lock */
+static void
+media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
 {
   GstRTSPMediaPrivate *priv = media->priv;
 
@@ -2072,22 +2921,48 @@ gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
     gst_rtsp_media_unprepare (media);
   } else {
     GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
-    priv->target_state = state;
+    set_target_state (media, state, FALSE);
     /* when we are buffering, don't update the state yet, this will be done
      * when buffering finishes */
     if (priv->buffering) {
       GST_INFO ("Buffering busy, delay state change");
     } else {
-      gst_element_set_state (priv->pipeline, state);
+      if (state == GST_STATE_PLAYING)
+        /* make sure pads are not blocking anymore when going to PLAYING */
+        media_streams_set_blocked (media, FALSE);
+
+      set_state (media, state);
+
+      /* and suspend after pause */
+      if (state == GST_STATE_PAUSED)
+        gst_rtsp_media_suspend (media);
     }
   }
 }
 
 /**
+ * gst_rtsp_media_set_pipeline_state:
+ * @media: a #GstRTSPMedia
+ * @state: the target state of the pipeline
+ *
+ * Set the state of the pipeline managed by @media to @state
+ */
+void
+gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
+{
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  g_rec_mutex_lock (&media->priv->state_lock);
+  media_set_pipeline_state_locked (media, state);
+  g_rec_mutex_unlock (&media->priv->state_lock);
+}
+
+/**
  * gst_rtsp_media_set_state:
  * @media: a #GstRTSPMedia
  * @state: the target state of the media
- * @transports: a #GPtrArray of #GstRTSPStreamTransport pointers
+ * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
+ * a #GPtrArray of #GstRTSPStreamTransport pointers
  *
  * Set the state of @media to @state and for the transports in @transports.
  *
@@ -2110,7 +2985,10 @@ gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
   priv = media->priv;
 
   g_rec_mutex_lock (&priv->state_lock);
-  if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
+  if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
+    goto error_status;
+  if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
+      priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
     goto not_prepared;
 
   /* NULL and READY are the same */
@@ -2169,7 +3047,7 @@ gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
 
   if (priv->target_state != state) {
     if (do_state)
-      gst_rtsp_media_set_pipeline_state (media, state);
+      media_set_pipeline_state_locked (media, state);
 
     g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
         NULL);
@@ -2191,36 +3069,24 @@ not_prepared:
     g_rec_mutex_unlock (&priv->state_lock);
     return FALSE;
   }
-}
-
-/* called with state-lock */
-static gboolean
-default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
-    GstRTSPRangeUnit unit)
-{
-  return gst_rtsp_range_convert_units (range, unit);
-}
-
-static gboolean
-default_query_position (GstRTSPMedia * media, gint64 * position)
-{
-  return gst_element_query_position (media->priv->pipeline, GST_FORMAT_TIME,
-      position);
-}
-
-static gboolean
-default_query_stop (GstRTSPMedia * media, gint64 * stop)
-{
-  GstQuery *query;
-  gboolean res;
-
-  query = gst_query_new_segment (GST_FORMAT_TIME);
-  if ((res = gst_element_query (media->priv->pipeline, query))) {
-    GstFormat format;
-    gst_query_parse_segment (query, NULL, &format, NULL, stop);
-    if (format != GST_FORMAT_TIME)
-      *stop = -1;
+error_status:
+  {
+    GST_WARNING ("media %p in error status while changing to state %d",
+        media, state);
+    if (state == GST_STATE_NULL) {
+      for (i = 0; i < transports->len; i++) {
+        GstRTSPStreamTransport *trans;
+
+        /* we need a non-NULL entry in the array */
+        trans = g_ptr_array_index (transports, i);
+        if (trans == NULL)
+          continue;
+
+        gst_rtsp_stream_transport_set_active (trans, FALSE);
+      }
+      priv->n_active = 0;
+    }
+    g_rec_mutex_unlock (&priv->state_lock);
+    return FALSE;
   }
-  gst_query_unref (query);
-  return res;
 }