gst/rtsp/gstrtspsrc.*: Use boilerplate.

Original commit message from CVS:
Based on patch by: Lutz Mueller <lutz at topfrose dot de>
* gst/rtsp/gstrtspsrc.c: (_do_init), (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Use boilerplate.
Make rtspsrc subclass GstBin to make state changes easier.
Add Range header field on the PLAY request.

ChangeLog

@@ -1,3 +1,16 @@
+2006-09-18  Wim Taymans  <wim@fluendo.com>
+
+	Based on patch by: Lutz Mueller <lutz at topfrose dot de>
+
+	* gst/rtsp/gstrtspsrc.c: (_do_init), (gst_rtspsrc_class_init),
+	(gst_rtspsrc_init), (gst_rtspsrc_stream_setup_rtp),
+	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_play),
+	(gst_rtspsrc_change_state):
+	* gst/rtsp/gstrtspsrc.h:
+	Use boilerplate.
+	Make rtspsrc subclass GstBin to make state changes easier.
+	Add Range header field on the PLAY request.
+
 2006-09-18  Wim Taymans  <wim@fluendo.com>
 
 	Based on patch by: Thijs Vermeir <thijs dot vermeir at barco dot com>

gst/rtsp/gstrtspsrc.c

@@ -78,7 +78,9 @@ static const GstElementDetails gst_rtspsrc_details =
 GST_ELEMENT_DETAILS ("RTSP packet receiver",
     "Source/Network",
     "Receive data over the network via RTSP (RFC 2326)",
-    "Wim Taymans <wim@fluendo.com>");
+    "Wim Taymans <wim@fluendo.com>\n"
+    "Thijs Vermeir <thijs.vermeir@barco.com>\n"
+    "Lutz Mueller <lutz@topfrose.de>");
 
 static GstStaticPadTemplate rtptemplate =
 GST_STATIC_PAD_TEMPLATE ("rtp_stream%d",
@@ -131,10 +133,7 @@ gst_rtsp_proto_get_type (void)
   return rtsp_proto_type;
 }
 
-
 static void gst_rtspsrc_base_init (gpointer g_class);
-static void gst_rtspsrc_class_init (GstRTSPSrc * klass);
-static void gst_rtspsrc_init (GstRTSPSrc * rtspsrc);
 static void gst_rtspsrc_finalize (GObject * object);
 
 static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
@@ -150,46 +149,25 @@ static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
 
 static void gst_rtspsrc_loop (GstRTSPSrc * src);
 
-static GstElementClass *parent_class = NULL;
-
 /*static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 }; */
 
-GType
+static void
-gst_rtspsrc_get_type (void)
+_do_init (GType rtspsrc_type)
 {
-  static GType rtspsrc_type = 0;
+  static const GInterfaceInfo urihandler_info = {
+    gst_rtspsrc_uri_handler_init,
+    NULL,
+    NULL
+  };
 
-  if (!rtspsrc_type) {
+  GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
-    static const GTypeInfo rtspsrc_info = {
-      sizeof (GstRTSPSrcClass),
-      gst_rtspsrc_base_init,
-      NULL,
-      (GClassInitFunc) gst_rtspsrc_class_init,
-      NULL,
-      NULL,
-      sizeof (GstRTSPSrc),
-      0,
-      (GInstanceInitFunc) gst_rtspsrc_init,
-      NULL
-    };
-    static const GInterfaceInfo urihandler_info = {
-      gst_rtspsrc_uri_handler_init,
-      NULL,
-      NULL
-    };
 
-    GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
+  g_type_add_interface_static (rtspsrc_type, GST_TYPE_URI_HANDLER,
-
+      &urihandler_info);
-    rtspsrc_type =
-        g_type_register_static (GST_TYPE_ELEMENT, "GstRTSPSrc", &rtspsrc_info,
-        0);
-
-    g_type_add_interface_static (rtspsrc_type, GST_TYPE_URI_HANDLER,
-        &urihandler_info);
-  }
-  return rtspsrc_type;
 }
 
+GST_BOILERPLATE_FULL (GstRTSPSrc, gst_rtspsrc, GstBin, GST_TYPE_BIN, _do_init);
+
 static void
 gst_rtspsrc_base_init (gpointer g_class)
 {
@@ -204,15 +182,15 @@ gst_rtspsrc_base_init (gpointer g_class)
 }
 
 static void
-gst_rtspsrc_class_init (GstRTSPSrc * klass)
+gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
 {
   GObjectClass *gobject_class;
   GstElementClass *gstelement_class;
+  GstBinClass *gstbin_class;
 
   gobject_class = (GObjectClass *) klass;
   gstelement_class = (GstElementClass *) klass;
-
+  gstbin_class = (GstBinClass *) klass;
-  parent_class = g_type_class_peek_parent (klass);
 
   gobject_class->set_property = gst_rtspsrc_set_property;
   gobject_class->get_property = gst_rtspsrc_get_property;
@@ -244,7 +222,7 @@ gst_rtspsrc_class_init (GstRTSPSrc * klass)
 }
 
 static void
-gst_rtspsrc_init (GstRTSPSrc * src)
+gst_rtspsrc_init (GstRTSPSrc * src, GstRTSPSrcClass * g_class)
 {
   src->stream_rec_lock = g_new (GStaticRecMutex, 1);
   g_static_rec_mutex_init (src->stream_rec_lock);
@@ -348,52 +326,6 @@ gst_rtspsrc_free_stream (GstRTSPSrc * src, GstRTSPStream * stream)
 }
 #endif
 
-static gboolean
-gst_rtspsrc_add_element (GstRTSPSrc * src, GstElement * element)
-{
-  gst_object_set_parent (GST_OBJECT (element), GST_OBJECT (src));
-
-  return TRUE;
-}
-
-static GstStateChangeReturn
-gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
-{
-  GstStateChangeReturn ret;
-  GList *streams;
-
-  ret = GST_STATE_CHANGE_SUCCESS;
-
-  /* for all streams */
-  for (streams = src->streams; streams; streams = g_list_next (streams)) {
-    GstRTSPStream *stream;
-
-    stream = (GstRTSPStream *) streams->data;
-
-    /* first our RTP session manager */
-    if (stream->rtpdec) {
-      ret = gst_element_set_state (stream->rtpdec, state);
-      if (ret == GST_STATE_CHANGE_FAILURE)
-        goto done;
-    }
-
-    /* then our sources */
-    if (stream->rtpsrc) {
-      ret = gst_element_set_state (stream->rtpsrc, state);
-      if (ret == GST_STATE_CHANGE_FAILURE)
-        goto done;
-    }
-    if (stream->rtcpsrc) {
-      ret = gst_element_set_state (stream->rtcpsrc, state);
-      if (ret == GST_STATE_CHANGE_FAILURE)
-        goto done;
-    }
-  }
-
-done:
-  return ret;
-}
-
 #define PARSE_INT(p, del, res)          \
 G_STMT_START {                          \
   gchar *t = p;                         \
@@ -648,9 +580,10 @@ again:
 
   /* we manage these elements, we set the caps in configure_transport */
   stream->rtpsrc = rtpsrc;
-  gst_rtspsrc_add_element (src, stream->rtpsrc);
   stream->rtcpsrc = rtcpsrc;
-  gst_rtspsrc_add_element (src, stream->rtcpsrc);
+
+  gst_bin_add (GST_BIN_CAST (src), stream->rtpsrc);
+  gst_bin_add (GST_BIN_CAST (src), stream->rtcpsrc);
 
   return TRUE;
 
@@ -722,7 +655,7 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
     goto no_element;
 
   /* we manage this element */
-  gst_rtspsrc_add_element (src, stream->rtpdec);
+  gst_bin_add (GST_BIN_CAST (src), stream->rtpdec);
 
   ret = gst_element_set_state (stream->rtpdec, GST_STATE_PAUSED);
   if (ret != GST_STATE_CHANGE_SUCCESS)
@@ -773,8 +706,8 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
       gst_element_set_state (stream->rtcpsrc, GST_STATE_PAUSED);
 
       /* we manage these elements */
-      gst_rtspsrc_add_element (src, stream->rtpsrc);
+      gst_bin_add (GST_BIN_CAST (src), stream->rtpsrc);
-      gst_rtspsrc_add_element (src, stream->rtcpsrc);
+      gst_bin_add (GST_BIN_CAST (src), stream->rtcpsrc);
     }
 
     /* configure caps on the RTP source element */
@@ -1398,6 +1331,8 @@ gst_rtspsrc_play (GstRTSPSrc * src)
   if (res < 0)
     goto create_request_failed;
 
+  rtsp_message_add_header (&request, RTSP_HDR_RANGE, "npt=0.000-");
+
   if (!gst_rtspsrc_send (src, &request, &response, NULL))
     goto send_error;
 
@@ -1490,10 +1425,6 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
   if (ret == GST_STATE_CHANGE_FAILURE)
     goto done;
 
-  ret = gst_rtspsrc_set_state (rtspsrc, GST_STATE_PENDING (rtspsrc));
-  if (ret == GST_STATE_CHANGE_FAILURE)
-    goto done;
-
   switch (transition) {
     case GST_STATE_CHANGE_READY_TO_PAUSED:
       ret = GST_STATE_CHANGE_NO_PREROLL;

gst/rtsp/gstrtspsrc.h

@@ -85,7 +85,7 @@ struct _GstRTSPStream {
 };
 
 struct _GstRTSPSrc {
-  GstElement element;
+  GstBin           parent;
 
   /* task and mutex for interleaved mode */
   gboolean         interleaved;
@@ -109,7 +109,7 @@ struct _GstRTSPSrc {
 };
 
 struct _GstRTSPSrcClass {
-  GstElementClass parent_class;
+  GstBinClass parent_class;
 };
 
 GType gst_rtspsrc_get_type(void);