gst/rtsp/gstrtspsrc.*: Small cleanups, added documentation.

Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_send), (gst_rtspsrc_parse_methods),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
(gst_rtspsrc_pause), (gst_rtspsrc_change_state),
(gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
Small cleanups, added documentation.
Try to clean up the requests and responses.
Refactor parsing the supported methods.
* gst/rtsp/rtspconnection.c: (rtsp_connection_open),
(rtsp_connection_create), (rtsp_connection_send),
(parse_response_status), (parse_request_line),
(rtsp_connection_receive), (rtsp_connection_close),
(rtsp_connection_free):
* gst/rtsp/rtsptransport.c: (rtsp_transport_new),
(rtsp_transport_init), (rtsp_transport_parse),
(rtsp_transport_free):
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
* gst/rtsp/sdpmessage.c: (sdp_message_new), (sdp_message_init),
(sdp_message_clean), (sdp_message_free), (sdp_media_new),
(sdp_media_init), (sdp_message_parse_buffer), (sdp_message_dump):
Use g_return_val some more.
* gst/rtsp/rtspdefs.h:
Add more enum values to track initial states.
* gst/rtsp/rtspmessage.c: (rtsp_message_new_request),
(rtsp_message_init_request), (rtsp_message_new_response),
(rtsp_message_init_response), (rtsp_message_init_data),
(rtsp_message_unset), (rtsp_message_free),
(rtsp_message_add_header), (rtsp_message_remove_header),
(rtsp_message_get_header), (rtsp_message_set_body),
(rtsp_message_take_body), (rtsp_message_get_body),
(rtsp_message_steal_body), (rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
Reorder arguments, object goes as the first one.
Use g_return_val some more.

10 files changed, 467 insertions, 306 deletions

diff --git a/ChangeLog b/ChangeLog
index cdd88100..1988a4d9 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,5 +1,46 @@
 2006-09-18  Wim Taymans  <wim@fluendo.com>
 
+	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
+	(gst_rtspsrc_send), (gst_rtspsrc_parse_methods),
+	(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
+	(gst_rtspsrc_pause), (gst_rtspsrc_change_state),
+	(gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
+	* gst/rtsp/gstrtspsrc.h:
+	Small cleanups, added documentation.
+	Try to clean up the requests and responses.
+	Refactor parsing the supported methods.
+
+	* gst/rtsp/rtspconnection.c: (rtsp_connection_open),
+	(rtsp_connection_create), (rtsp_connection_send),
+	(parse_response_status), (parse_request_line),
+	(rtsp_connection_receive), (rtsp_connection_close),
+	(rtsp_connection_free):
+	* gst/rtsp/rtsptransport.c: (rtsp_transport_new),
+	(rtsp_transport_init), (rtsp_transport_parse),
+	(rtsp_transport_free):
+	* gst/rtsp/rtspurl.c: (rtsp_url_parse):
+	* gst/rtsp/sdpmessage.c: (sdp_message_new), (sdp_message_init),
+	(sdp_message_clean), (sdp_message_free), (sdp_media_new),
+	(sdp_media_init), (sdp_message_parse_buffer), (sdp_message_dump):
+	Use g_return_val some more.
+
+	* gst/rtsp/rtspdefs.h:
+	Add more enum values to track initial states.
+
+	* gst/rtsp/rtspmessage.c: (rtsp_message_new_request),
+	(rtsp_message_init_request), (rtsp_message_new_response),
+	(rtsp_message_init_response), (rtsp_message_init_data),
+	(rtsp_message_unset), (rtsp_message_free),
+	(rtsp_message_add_header), (rtsp_message_remove_header),
+	(rtsp_message_get_header), (rtsp_message_set_body),
+	(rtsp_message_take_body), (rtsp_message_get_body),
+	(rtsp_message_steal_body), (rtsp_message_dump):
+	* gst/rtsp/rtspmessage.h:
+	Reorder arguments, object goes as the first one.
+	Use g_return_val some more.
+
+2006-09-18  Wim Taymans  <wim@fluendo.com>
+
 	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_base_init),
 	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
 	(gst_rtspsrc_loop), (gst_rtspsrc_uri_set_uri):
diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index 1353852c..3bc87932 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -354,6 +354,10 @@ G_STMT_START {                          \
   while (*p && g_ascii_isspace (*p))    \
     p++;
 
+/* rtpmap contains:
+ *
+ *  <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
+ */
 static gboolean
 gst_rtspsrc_parse_rtpmap (gchar * rtpmap, gint * payload, gchar ** name,
     gint * rate, gchar ** params)
@@ -413,10 +417,9 @@ gst_rtspsrc_media_to_caps (SDPMedia * media)
   GstStructure *s;
 
   payload = sdp_media_get_format (media, 0);
-  if (payload == NULL) {
-    g_warning ("payload type not given");
-    return NULL;
-  }
+  if (payload == NULL)
+    goto no_payload;
+
   pt = atoi (payload);
 
   /* dynamic payloads need rtpmap */
@@ -428,18 +431,18 @@ gst_rtspsrc_media_to_caps (SDPMedia * media)
       ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, &params);
       if (ret) {
         if (payload != pt) {
+          /* FIXME, not fatal? */
           g_warning ("rtpmap of wrong payload type");
           name = NULL;
           rate = -1;
           params = NULL;
         }
       } else {
+        /* FIXME, not fatal? */
         g_warning ("error parsing rtpmap");
       }
-    } else {
-      g_warning ("rtpmap type not given for dynamic payload %d", pt);
-      return NULL;
-    }
+    } else
+      goto no_rtpmap;
   }
 
   caps = gst_caps_new_simple ("application/x-rtp",
@@ -494,6 +497,18 @@ gst_rtspsrc_media_to_caps (SDPMedia * media)
     }
   }
   return caps;
+
+  /* ERRORS */
+no_payload:
+  {
+    g_warning ("payload type not given");
+    return NULL;
+  }
+no_rtpmap:
+  {
+    g_warning ("rtpmap type not given for dynamic payload %d", pt);
+    return NULL;
+  }
 }
 
 static gboolean
@@ -948,29 +963,48 @@ need_pause:
   }
 }
 
+/**
+ * gst_rtspsrc_send:
+ * @src: the rtsp source
+ * @request: must point to a valid request
+ * @response: must point to an empty #RTSPMessage
+ *
+ * send @request and retrieve the response in @response. optionally @code can be
+ * non-NULL in which case it will contain the status code of the response.
+ *
+ * If This function returns TRUE, @response will contain a valid response
+ * message that should be cleaned with rtsp_message_unset() after usage. 
+ *
+ * If @code is NULL, this function will return FALSE (with an invalid @response
+ * message) if the response code was not 200 (OK).
+ *
+ * Returns: TRUE if the processing was successful.
+ */
 static gboolean
 gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
     RTSPMessage * response, RTSPStatusCode * code)
 {
   RTSPResult res;
+  RTSPStatusCode thecode;
 
-  if (src->debug) {
+  if (src->debug)
     rtsp_message_dump (request);
-  }
+
   if ((res = rtsp_connection_send (src->connection, request)) < 0)
     goto send_error;
 
   if ((res = rtsp_connection_receive (src->connection, response)) < 0)
     goto receive_error;
 
-  if (code) {
-    *code = response->type_data.response.code;
-  }
-
-  if (src->debug) {
+  if (src->debug)
     rtsp_message_dump (response);
-  }
-  if (response->type_data.response.code != RTSP_STS_OK)
+
+  thecode = response->type_data.response.code;
+  /* if the caller wanted the result code, we store it. Else we check if it's
+   * OK. */
+  if (code)
+    *code = thecode;
+  else if (thecode != RTSP_STS_OK)
     goto error_response;
 
   return TRUE;
@@ -990,9 +1024,92 @@ receive_error:
   }
 error_response:
   {
-    GST_ELEMENT_ERROR (src, RESOURCE, READ, ("Got error response: %d (%s).",
-            response->type_data.response.code,
-            response->type_data.response.reason), (NULL));
+    switch (response->type_data.response.code) {
+      case RTSP_STS_NOT_FOUND:
+        GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, ("%s",
+                response->type_data.response.reason), (NULL));
+        break;
+      default:
+        GST_ELEMENT_ERROR (src, RESOURCE, READ, ("Got error response: %d (%s).",
+                response->type_data.response.code,
+                response->type_data.response.reason), (NULL));
+        break;
+    }
+    /* we return FALSE so we should unset the response ourselves */
+    rtsp_message_unset (response);
+    return FALSE;
+  }
+}
+
+/* parse the response and collect all the supported methods. We need this
+ * information so that we don't try to send an unsupported request to the
+ * server.
+ */
+static gboolean
+gst_rtspsrc_parse_methods (GstRTSPSrc * src, RTSPMessage * response)
+{
+  gchar *respoptions = NULL;
+  gchar **options;
+  gint i;
+
+  /* clear supported methods */
+  src->methods = 0;
+
+  /* Try Allow Header first */
+  rtsp_message_get_header (response, RTSP_HDR_ALLOW, &respoptions);
+  if (!respoptions)
+    /* Then maybe Public Header... */
+    rtsp_message_get_header (response, RTSP_HDR_PUBLIC, &respoptions);
+  if (!respoptions) {
+    /* this field is not required, assume the server supports
+     * DESCRIBE and SETUP*/
+    GST_DEBUG_OBJECT (src, "could not get OPTIONS");
+    src->methods = RTSP_DESCRIBE | RTSP_SETUP;
+    goto done;
+  }
+
+  /* If we get here, the server gave a list of supported methods, parse
+   * them here. The string is like: 
+   *
+   * OPTIONS, DESCRIBE, ANNOUNCE, PLAY, SETUP, ...
+   */
+  options = g_strsplit (respoptions, ",", 0);
+
+  i = 0;
+  while (options[i]) {
+    gchar *stripped;
+    gint method;
+
+    stripped = g_strstrip (options[i]);
+    method = rtsp_find_method (stripped);
+
+    /* keep bitfield of supported methods */
+    if (method != -1)
+      src->methods |= method;
+    i++;
+  }
+  g_strfreev (options);
+
+  /* we need describe and setup */
+  if (!(src->methods & RTSP_DESCRIBE))
+    goto no_describe;
+  if (!(src->methods & RTSP_SETUP))
+    goto no_setup;
+
+done:
+  return TRUE;
+
+  /* ERRORS */
+no_describe:
+  {
+    GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ,
+        ("Server does not support DESCRIBE."), (NULL));
+    return FALSE;
+  }
+no_setup:
+  {
+    GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ,
+        ("Server does not support SETUP."), (NULL));
     return FALSE;
   }
 }
@@ -1005,11 +1122,13 @@ gst_rtspsrc_open (GstRTSPSrc * src)
   RTSPMessage response = { 0 };
   guint8 *data;
   guint size;
+  gint i, n_streams;
   SDPMessage sdp = { 0 };
   GstRTSPProto protocols;
+  GstRTSPStream *stream = NULL;
 
   /* can't continue without a valid url */
-  if (src->url == NULL)
+  if (G_UNLIKELY (src->url == NULL))
     goto no_url;
 
   /* open connection */
@@ -1019,7 +1138,7 @@ gst_rtspsrc_open (GstRTSPSrc * src)
 
   /* create OPTIONS */
   GST_DEBUG_OBJECT (src, "create options...");
-  res = rtsp_message_init_request (RTSP_OPTIONS, src->location, &request);
+  res = rtsp_message_init_request (&request, RTSP_OPTIONS, src->location);
   if (res < 0)
     goto create_request_failed;
 
@@ -1028,56 +1147,13 @@ gst_rtspsrc_open (GstRTSPSrc * src)
   if (!gst_rtspsrc_send (src, &request, &response, NULL))
     goto send_error;
 
-  {
-    gchar *respoptions = NULL;
-    gchar **options;
-    gint i;
-
-    /* Try Allow Header first */
-    rtsp_message_get_header (&response, RTSP_HDR_ALLOW, &respoptions);
-    if (!respoptions) {
-      /* Then maybe Public Header... */
-      rtsp_message_get_header (&response, RTSP_HDR_PUBLIC, &respoptions);
-      if (!respoptions) {
-        /* this field is not required, assume the server supports
-         * DESCRIBE and SETUP*/
-        GST_DEBUG_OBJECT (src, "could not get OPTIONS");
-        src->options = RTSP_DESCRIBE | RTSP_SETUP;
-        goto no_options;
-      }
-    }
-
-    /* parse options */
-    options = g_strsplit (respoptions, ",", 0);
-
-    i = 0;
-    while (options[i]) {
-      gchar *stripped;
-      gint method;
-
-      stripped = g_strdup (options[i]);
-      stripped = g_strstrip (stripped);
-      method = rtsp_find_method (stripped);
-      g_free (stripped);
-
-      /* keep bitfield of supported methods */
-      if (method != -1)
-        src->options |= method;
-      i++;
-    }
-    g_strfreev (options);
-
-  no_options:
-    /* we need describe and setup */
-    if (!(src->options & RTSP_DESCRIBE))
-      goto no_describe;
-    if (!(src->options & RTSP_SETUP))
-      goto no_setup;
-  }
+  /* parse OPTIONS */
+  if (!gst_rtspsrc_parse_methods (src, &response))
+    goto methods_error;
 
   /* create DESCRIBE */
   GST_DEBUG_OBJECT (src, "create describe...");
-  res = rtsp_message_init_request (RTSP_DESCRIBE, src->location, &request);
+  res = rtsp_message_init_request (&request, RTSP_DESCRIBE, src->location);
   if (res < 0)
     goto create_request_failed;
 
@@ -1117,134 +1193,138 @@ gst_rtspsrc_open (GstRTSPSrc * src)
   protocols = src->protocols;
 
   /* setup streams */
-  {
-    gint i;
-
-    for (i = 0; i < sdp_message_medias_len (&sdp); i++) {
-      SDPMedia *media;
-      gchar *setup_url;
-      gchar *control_url;
-      gchar *transports;
-      GstRTSPStream *stream;
-
-      media = sdp_message_get_media (&sdp, i);
-
-      stream = gst_rtspsrc_create_stream (src);
-
-      GST_DEBUG_OBJECT (src, "setup media %d", i);
-      control_url = sdp_media_get_attribute_val (media, "control");
-      if (control_url == NULL) {
-        GST_DEBUG_OBJECT (src, "no control url found, skipping stream");
-        continue;
-      }
+  n_streams = sdp_message_medias_len (&sdp);
+  for (i = 0; i < n_streams; i++) {
+    SDPMedia *media;
+    gchar *setup_url;
+    gchar *control_url;
+    gchar *transports;
+
+    media = sdp_message_get_media (&sdp, i);
+
+    GST_DEBUG_OBJECT (src, "setup media %d", i);
+    control_url = sdp_media_get_attribute_val (media, "control");
+    if (control_url == NULL) {
+      GST_DEBUG_OBJECT (src, "no control url found, skipping stream %d", i);
+      continue;
+    }
 
-      /* check absolute/relative URL */
-      /* FIXME, what if the control_url starts with a '/' or a non rtsp: protocol? */
-      if (g_str_has_prefix (control_url, "rtsp://")) {
-        setup_url = g_strdup (control_url);
-      } else {
-        setup_url = g_strdup_printf ("%s/%s", src->location, control_url);
-      }
+    /* check absolute/relative URL */
+    /* FIXME, what if the control_url starts with a '/' or a non rtsp: protocol? */
+    if (g_str_has_prefix (control_url, "rtsp://")) {
+      setup_url = g_strdup (control_url);
+    } else {
+      setup_url = g_strdup_printf ("%s/%s", src->location, control_url);
+    }
 
-      GST_DEBUG_OBJECT (src, "setup %s", setup_url);
+    GST_DEBUG_OBJECT (src, "setup %s", setup_url);
 
-      /* create SETUP request */
-      res = rtsp_message_init_request (RTSP_SETUP, setup_url, &request);
-      g_free (setup_url);
-      if (res < 0)
-        goto create_request_failed;
+    /* create SETUP request */
+    res = rtsp_message_init_request (&request, RTSP_SETUP, setup_url);
+    g_free (setup_url);
+    if (res < 0)
+      goto create_request_failed;
 
-      transports = g_strdup ("");
-      if (protocols & GST_RTSP_PROTO_UDP_UNICAST) {
-        gchar *new;
-        gint rtpport, rtcpport;
-        gchar *trxparams;
+    stream = gst_rtspsrc_create_stream (src);
 
-        /* allocate two UDP ports */
-        if (!gst_rtspsrc_stream_setup_rtp (stream, media, &rtpport, &rtcpport))
-          goto setup_rtp_failed;
+    transports = g_strdup ("");
+    if (protocols & GST_RTSP_PROTO_UDP_UNICAST) {
+      gchar *new;
+      gint rtpport, rtcpport;
+      gchar *trxparams;
 
-        GST_DEBUG_OBJECT (src, "setting up RTP ports %d-%d", rtpport, rtcpport);
+      /* allocate two UDP ports */
+      if (!gst_rtspsrc_stream_setup_rtp (stream, media, &rtpport, &rtcpport))
+        goto setup_rtp_failed;
 
-        trxparams = g_strdup_printf ("client_port=%d-%d", rtpport, rtcpport);
-        new = g_strconcat (transports, "RTP/AVP/UDP;unicast;", trxparams, NULL);
-        g_free (trxparams);
-        g_free (transports);
-        transports = new;
-      }
-      if (protocols & GST_RTSP_PROTO_UDP_MULTICAST) {
-        gchar *new;
-
-        GST_DEBUG_OBJECT (src, "setting up MULTICAST");
-
-        /* we don't hav to allocate any UDP ports yet, if the selected transport
-         * turns out to be multicast we can create them and join the multicast
-         * group indicated in the transport reply */
-        new =
-            g_strconcat (transports, transports[0] ? "," : "",
-            "RTP/AVP/UDP;multicast", NULL);
-        g_free (transports);
-        transports = new;
-      }
-      if (protocols & GST_RTSP_PROTO_TCP) {
-        gchar *new;
+      GST_DEBUG_OBJECT (src, "setting up RTP ports %d-%d", rtpport, rtcpport);
 
-        GST_DEBUG_OBJECT (src, "setting up TCP");
+      trxparams = g_strdup_printf ("client_port=%d-%d", rtpport, rtcpport);
+      new = g_strconcat (transports, "RTP/AVP/UDP;unicast;", trxparams, NULL);
+      g_free (trxparams);
+      g_free (transports);
+      transports = new;
+    }
+    if (protocols & GST_RTSP_PROTO_UDP_MULTICAST) {
+      gchar *new;
 
-        new =
-            g_strconcat (transports, transports[0] ? "," : "", "RTP/AVP/TCP",
-            NULL);
-        g_free (transports);
-        transports = new;
-      }
+      GST_DEBUG_OBJECT (src, "setting up MULTICAST");
 
-      /* select transport, copy is made when adding to header */
-      rtsp_message_add_header (&request, RTSP_HDR_TRANSPORT, transports);
+      /* we don't hav to allocate any UDP ports yet, if the selected transport
+       * turns out to be multicast we can create them and join the multicast
+       * group indicated in the transport reply */
+      new =
+          g_strconcat (transports, transports[0] ? "," : "",
+          "RTP/AVP/UDP;multicast", NULL);
       g_free (transports);
+      transports = new;
+    }
+    if (protocols & GST_RTSP_PROTO_TCP) {
+      gchar *new;
 
-      if (!gst_rtspsrc_send (src, &request, &response, NULL))
-        goto send_error;
+      GST_DEBUG_OBJECT (src, "setting up TCP");
 
-      /* parse response transport */
-      {
-        gchar *resptrans = NULL;
-        RTSPTransport transport = { 0 };
+      new =
+          g_strconcat (transports, transports[0] ? "," : "", "RTP/AVP/TCP",
+          NULL);
+      g_free (transports);
+      transports = new;
+    }
 
-        rtsp_message_get_header (&response, RTSP_HDR_TRANSPORT, &resptrans);
-        if (!resptrans)
-          goto no_transport;
+    /* select transport, copy is made when adding to header */
+    rtsp_message_add_header (&request, RTSP_HDR_TRANSPORT, transports);
+    g_free (transports);
 
-        /* parse transport */
-        rtsp_transport_parse (resptrans, &transport);
+    if (!gst_rtspsrc_send (src, &request, &response, NULL))
+      goto send_error;
 
-        /* update allowed transports for other streams. once the transport of
-         * one stream has been determined, we make sure that all other streams
-         * are configured in the same way */
-        if (transport.lower_transport == RTSP_LOWER_TRANS_TCP) {
-          GST_DEBUG_OBJECT (src, "stream %d as TCP", i);
-          protocols = GST_RTSP_PROTO_TCP;
-          src->interleaved = TRUE;
+    /* parse response transport */
+    {
+      gchar *resptrans = NULL;
+      RTSPTransport transport = { 0 };
+
+      rtsp_message_get_header (&response, RTSP_HDR_TRANSPORT, &resptrans);
+      if (!resptrans)
+        goto no_transport;
+
+      /* parse transport */
+      rtsp_transport_parse (resptrans, &transport);
+
+      /* update allowed transports for other streams. once the transport of
+       * one stream has been determined, we make sure that all other streams
+       * are configured in the same way */
+      if (transport.lower_transport == RTSP_LOWER_TRANS_TCP) {
+        GST_DEBUG_OBJECT (src, "stream %d as TCP", i);
+        protocols = GST_RTSP_PROTO_TCP;
+        src->interleaved = TRUE;
+      } else {
+        if (transport.multicast) {
+          /* only allow multicast for other streams */
+          GST_DEBUG_OBJECT (src, "stream %d as MULTICAST", i);
+          protocols = GST_RTSP_PROTO_UDP_MULTICAST;
         } else {
-          if (transport.multicast) {
-            /* only allow multicast for other streams */
-            GST_DEBUG_OBJECT (src, "stream %d as MULTICAST", i);
-            protocols = GST_RTSP_PROTO_UDP_MULTICAST;
-          } else {
-            /* only allow unicast for other streams */
-            GST_DEBUG_OBJECT (src, "stream %d as UNICAST", i);
-            protocols = GST_RTSP_PROTO_UDP_UNICAST;
-          }
-        }
-        /* now configure the stream with the transport */
-        if (!gst_rtspsrc_stream_configure_transport (stream, media, &transport)) {
-          GST_DEBUG_OBJECT (src,
-              "could not configure stream transport, skipping stream");
+          /* only allow unicast for other streams */
+          GST_DEBUG_OBJECT (src, "stream %d as UNICAST", i);
+          protocols = GST_RTSP_PROTO_UDP_UNICAST;
         }
-        /* clean up our transport struct */
-        rtsp_transport_init (&transport);
       }
+      /* now configure the stream with the transport */
+      if (!gst_rtspsrc_stream_configure_transport (stream, media, &transport)) {
+        GST_DEBUG_OBJECT (src,
+            "could not configure stream %d transport, skipping stream", i);
+      }
+      /* clean up our transport struct */
+      rtsp_transport_init (&transport);
     }
   }
+  /* if we got here all was configured. We have dynamic pads so we notify that
+   * we are done */
+  gst_element_no_more_pads (GST_ELEMENT_CAST (src));
+
+  /* clean up any messages */
+  rtsp_message_unset (&request);
+  rtsp_message_unset (&response);
+
   return TRUE;
 
   /* ERRORS */
@@ -1252,53 +1332,53 @@ no_url:
   {
     GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
         ("No valid RTSP url was provided"), (NULL));
-    return FALSE;
+    goto cleanup_error;
   }
 could_not_open:
   {
     GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE,
         ("Could not open connection."), (NULL));
-    return FALSE;
+    goto cleanup_error;
   }
 create_request_failed:
   {
     GST_ELEMENT_ERROR (src, LIBRARY, INIT,
         ("Could not create request."), (NULL));
-    return FALSE;
+    goto cleanup_error;
   }
 send_error:
   {
     GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
         ("Could not send message."), (NULL));
-    return FALSE;
-  }
-no_describe:
-  {
-    GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
-        ("Server does not support DESCRIBE."), (NULL));
-    return FALSE;
+    goto cleanup_error;
   }
-no_setup:
+methods_error:
   {
-    GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
-        ("Server does not support SETUP."), (NULL));
-    return FALSE;
+    /* error was posted */
+    goto cleanup_error;
   }
 wrong_content_type:
   {
-    GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
+    GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
         ("Server does not support SDP."), (NULL));
-    return FALSE;
+    goto cleanup_error;
   }
 setup_rtp_failed:
   {
-    GST_ELEMENT_ERROR (src, RESOURCE, WRITE, ("Could not setup rtp."), (NULL));
-    return FALSE;
+    GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, ("Could not setup rtp."),
+        (NULL));
+    goto cleanup_error;
   }
 no_transport:
   {
-    GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
+    GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
         ("Server did not select transport."), (NULL));
+    goto cleanup_error;
+  }
+cleanup_error:
+  {
+    rtsp_message_unset (&request);
+    rtsp_message_unset (&response);
     return FALSE;
   }
 }
@@ -1328,14 +1408,18 @@ gst_rtspsrc_close (GstRTSPSrc * src)
     src->task = NULL;
   }
 
-  if (src->options & RTSP_PLAY) {
+  if (src->methods & RTSP_PLAY) {
     /* do TEARDOWN */
-    res = rtsp_message_init_request (RTSP_TEARDOWN, src->location, &request);
+    res = rtsp_message_init_request (&request, RTSP_TEARDOWN, src->location);
     if (res < 0)
       goto create_request_failed;
 
     if (!gst_rtspsrc_send (src, &request, &response, NULL))
       goto send_error;
+
+    /* FIXME, parse result? */
+    rtsp_message_unset (&request);
+    rtsp_message_unset (&response);
   }
 
   /* close connection */
@@ -1354,6 +1438,7 @@ create_request_failed:
   }
 send_error:
   {
+    rtsp_message_unset (&request);
     GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
         ("Could not send message."), (NULL));
     return FALSE;
@@ -1372,13 +1457,13 @@ gst_rtspsrc_play (GstRTSPSrc * src)
   RTSPMessage response = { 0 };
   RTSPResult res;
 
-  if (!(src->options & RTSP_PLAY))
+  if (!(src->methods & RTSP_PLAY))
     return TRUE;
 
   GST_DEBUG_OBJECT (src, "PLAY...");
 
   /* do play */
-  res = rtsp_message_init_request (RTSP_PLAY, src->location, &request);
+  res = rtsp_message_init_request (&request, RTSP_PLAY, src->location);
   if (res < 0)
     goto create_request_failed;
 
@@ -1387,10 +1472,16 @@ gst_rtspsrc_play (GstRTSPSrc * src)
   if (!gst_rtspsrc_send (src, &request, &response, NULL))
     goto send_error;
 
+  rtsp_message_unset (&request);
+  rtsp_message_unset (&response);
+
+  /* for interleaved transport, we receive the data on the RTSP connection
+   * instead of UDP. We start a task to select and read from that connection. */
   if (src->interleaved) {
     src->task = gst_task_create ((GstTaskFunction) gst_rtspsrc_loop, src);
 
     gst_task_set_lock (src->task, src->stream_rec_lock);
+    src->running = TRUE;
     gst_task_start (src->task);
   }
 
@@ -1405,6 +1496,7 @@ create_request_failed:
   }
 send_error:
   {
+    rtsp_message_unset (&request);
     GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
         ("Could not send message."), (NULL));
     return FALSE;
@@ -1418,18 +1510,21 @@ gst_rtspsrc_pause (GstRTSPSrc * src)
   RTSPMessage response = { 0 };
   RTSPResult res;
 
-  if (!(src->options & RTSP_PAUSE))
+  if (!(src->methods & RTSP_PAUSE))
     return TRUE;
 
   GST_DEBUG_OBJECT (src, "PAUSE...");
   /* do pause */
-  res = rtsp_message_init_request (RTSP_PAUSE, src->location, &request);
+  res = rtsp_message_init_request (&request, RTSP_PAUSE, src->location);
   if (res < 0)
     goto create_request_failed;
 
   if (!gst_rtspsrc_send (src, &request, &response, NULL))
     goto send_error;
 
+  rtsp_message_unset (&request);
+  rtsp_message_unset (&response);
+
   return TRUE;
 
   /* ERRORS */
@@ -1441,6 +1536,7 @@ create_request_failed:
   }
 send_error:
   {
+    rtsp_message_unset (&request);
     GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
         ("Could not send message."), (NULL));
     return FALSE;
@@ -1461,7 +1557,7 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
       break;
     case GST_STATE_CHANGE_READY_TO_PAUSED:
       rtspsrc->interleaved = FALSE;
-      rtspsrc->options = 0;
+      gst_segment_init (&rtspsrc->segment, GST_FORMAT_TIME);
       if (!gst_rtspsrc_open (rtspsrc))
         goto open_failed;
       break;
@@ -1521,6 +1617,7 @@ gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
 {
   GstRTSPSrc *src = GST_RTSPSRC (handler);
 
+  /* should not dup */
   return src->location;
 }
 
@@ -1560,8 +1657,8 @@ was_ok:
   }
 parse_error:
   {
-    GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)", GST_STR_NULL (uri),
-        res);
+    GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
+        GST_STR_NULL (uri), res);
     return FALSE;
   }
 }
diff --git a/gst/rtsp/gstrtspsrc.h b/gst/rtsp/gstrtspsrc.h
index 73267edd..3b3f7240 100644
--- a/gst/rtsp/gstrtspsrc.h
+++ b/gst/rtsp/gstrtspsrc.h
@@ -103,8 +103,8 @@ struct _GstRTSPSrc {
   guint   	   retry;
 
   GstRTSPProto     protocols;
-  /* supported options */
-  gint             options;
+  /* supported methods */
+  gint             methods;
 
   RTSPConnection  *connection;
   RTSPMessage     *request;
diff --git a/gst/rtsp/rtspconnection.c b/gst/rtsp/rtspconnection.c
index f773fa28..5d1ab22c 100644
--- a/gst/rtsp/rtspconnection.c
+++ b/gst/rtsp/rtspconnection.c
@@ -71,8 +71,8 @@ rtsp_connection_open (RTSPUrl * url, RTSPConnection ** conn)
   struct in_addr addr;
   gint ret;
 
-  if (url == NULL || conn == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (url != NULL, RTSP_EINVAL);
+  g_return_val_if_fail (conn != NULL, RTSP_EINVAL);
 
   if (url->protocol != RTSP_PROTO_TCP)
     return RTSP_ENOTIMPL;
@@ -128,6 +128,8 @@ rtsp_connection_create (gint fd, RTSPConnection ** conn)
 {
   RTSPConnection *newconn;
 
+  g_return_val_if_fail (conn != NULL, RTSP_EINVAL);
+
   /* FIXME check fd, must be connected SOCK_STREAM */
 
   newconn = g_new (RTSPConnection, 1);
@@ -157,8 +159,8 @@ rtsp_connection_send (RTSPConnection * conn, RTSPMessage * message)
   gint towrite;
   gchar *data;
 
-  if (conn == NULL || message == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (conn != NULL, RTSP_EINVAL);
+  g_return_val_if_fail (message != NULL, RTSP_EINVAL);
 
 #ifdef G_OS_WIN32
   WSADATA w;
@@ -333,7 +335,7 @@ parse_response_status (gchar * buffer, RTSPMessage * msg)
   while (g_ascii_isspace (*bptr))
     bptr++;
 
-  rtsp_message_init_response (code, bptr, NULL, msg);
+  rtsp_message_init_response (msg, code, bptr, NULL);
 
   return RTSP_OK;
 
@@ -365,7 +367,7 @@ parse_request_line (gchar * buffer, RTSPMessage * msg)
   if (strcmp (versionstr, "RTSP/1.0") != 0)
     goto wrong_version;
 
-  rtsp_message_init_request (method, urlstr, msg);
+  rtsp_message_init_request (msg, method, urlstr);
 
   return RTSP_OK;
 
@@ -456,8 +458,8 @@ rtsp_connection_receive (RTSPConnection * conn, RTSPMessage * msg)
   RTSPResult res;
   gboolean need_body;
 
-  if (conn == NULL || msg == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (conn != NULL, RTSP_EINVAL);
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
 
   line = 0;
 
@@ -490,7 +492,7 @@ rtsp_connection_receive (RTSPConnection * conn, RTSPMessage * msg)
         goto error;
 
       /* now we create a data message */
-      rtsp_message_init_data ((gint) c, msg);
+      rtsp_message_init_data (msg, (gint) c);
 
       /* next two bytes are the length of the data */
       ret = read (conn->fd, &size, 2);
@@ -590,8 +592,7 @@ rtsp_connection_close (RTSPConnection * conn)
 {
   gint res;
 
-  if (conn == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (conn != NULL, RTSP_EINVAL);
 
   res = CLOSE_SOCKET (conn->fd);
 #ifdef G_OS_WIN32
@@ -612,8 +613,7 @@ sys_error:
 RTSPResult
 rtsp_connection_free (RTSPConnection * conn)
 {
-  if (conn == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (conn != NULL, RTSP_EINVAL);
 
 #ifdef G_OS_WIN32
   WSACleanup ();
diff --git a/gst/rtsp/rtspdefs.h b/gst/rtsp/rtspdefs.h
index 169ce48c..a7e89320 100644
--- a/gst/rtsp/rtspdefs.h
+++ b/gst/rtsp/rtspdefs.h
@@ -50,6 +50,7 @@ typedef enum {
 } RTSPFamily;
 
 typedef enum {
+  RTSP_STATE_INVALID,
   RTSP_STATE_INIT,
   RTSP_STATE_READY,
   RTSP_STATE_PLAYING,
@@ -57,6 +58,7 @@ typedef enum {
 } RTSPState;
 
 typedef enum {
+  RTSP_INVALID          = 0,
   RTSP_DESCRIBE         = (1 <<  0),
   RTSP_ANNOUNCE         = (1 <<  1),
   RTSP_GET_PARAMETER    = (1 <<  2),
@@ -120,6 +122,7 @@ typedef enum {
 } RTSPHeaderField;
 
 typedef enum {
+  RTSP_STS_INVALID                              = 0, 
   RTSP_STS_CONTINUE                             = 100, 
   RTSP_STS_OK                                   = 200, 
   RTSP_STS_CREATED                              = 201, 
diff --git a/gst/rtsp/rtspmessage.c b/gst/rtsp/rtspmessage.c
index fd45f599..bd1473bb 100644
--- a/gst/rtsp/rtspmessage.c
+++ b/gst/rtsp/rtspmessage.c
@@ -20,84 +20,69 @@
 #include "rtspmessage.h"
 
 RTSPResult
-rtsp_message_new_request (RTSPMethod method, gchar * uri, RTSPMessage ** msg)
+rtsp_message_new_request (RTSPMessage ** msg, RTSPMethod method, gchar * uri)
 {
   RTSPMessage *newmsg;
 
-  if (msg == NULL || uri == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
+  g_return_val_if_fail (uri != NULL, RTSP_EINVAL);
 
   newmsg = g_new0 (RTSPMessage, 1);
 
   *msg = newmsg;
 
-  return rtsp_message_init_request (method, uri, newmsg);
+  return rtsp_message_init_request (newmsg, method, uri);
 }
 
 RTSPResult
-rtsp_message_init_request (RTSPMethod method, gchar * uri, RTSPMessage * msg)
+rtsp_message_init_request (RTSPMessage * msg, RTSPMethod method, gchar * uri)
 {
-  if (msg == NULL || uri == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (uri != NULL, RTSP_EINVAL);
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
+
+  rtsp_message_unset (msg);
 
   msg->type = RTSP_MESSAGE_REQUEST;
   msg->type_data.request.method = method;
-  g_free (msg->type_data.request.uri);
   msg->type_data.request.uri = g_strdup (uri);
-
-  if (msg->hdr_fields != NULL)
-    g_hash_table_destroy (msg->hdr_fields);
   msg->hdr_fields =
       g_hash_table_new_full (g_direct_hash, g_direct_equal, NULL, g_free);
 
-  if (msg->body) {
-    g_free (msg->body);
-    msg->body = NULL;
-  }
-  msg->body_size = 0;
-
   return RTSP_OK;
 }
 
 RTSPResult
-rtsp_message_new_response (RTSPStatusCode code, gchar * reason,
-    RTSPMessage * request, RTSPMessage ** msg)
+rtsp_message_new_response (RTSPMessage ** msg, RTSPStatusCode code,
+    gchar * reason, RTSPMessage * request)
 {
   RTSPMessage *newmsg;
 
-  if (msg == NULL || reason == NULL || request == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (reason != NULL, RTSP_EINVAL);
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
+  g_return_val_if_fail (request != NULL, RTSP_EINVAL);
 
   newmsg = g_new0 (RTSPMessage, 1);
 
   *msg = newmsg;
 
-  return rtsp_message_init_response (code, reason, request, newmsg);
+  return rtsp_message_init_response (newmsg, code, reason, request);
 }
 
 RTSPResult
-rtsp_message_init_response (RTSPStatusCode code, gchar * reason,
-    RTSPMessage * request, RTSPMessage * msg)
+rtsp_message_init_response (RTSPMessage * msg, RTSPStatusCode code,
+    gchar * reason, RTSPMessage * request)
 {
-  if (reason == NULL || msg == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
+  g_return_val_if_fail (reason != NULL, RTSP_EINVAL);
+
+  rtsp_message_unset (msg);
 
   msg->type = RTSP_MESSAGE_RESPONSE;
   msg->type_data.response.code = code;
-  g_free (msg->type_data.response.reason);
   msg->type_data.response.reason = g_strdup (reason);
-
-  if (msg->hdr_fields != NULL)
-    g_hash_table_destroy (msg->hdr_fields);
   msg->hdr_fields =
       g_hash_table_new_full (g_direct_hash, g_direct_equal, NULL, g_free);
 
-  if (msg->body) {
-    g_free (msg->body);
-    msg->body = NULL;
-  }
-  msg->body_size = 0;
-
   if (request) {
     /* FIXME copy headers */
   }
@@ -106,10 +91,11 @@ rtsp_message_init_response (RTSPStatusCode code, gchar * reason,
 }
 
 RTSPResult
-rtsp_message_init_data (gint channel, RTSPMessage * msg)
+rtsp_message_init_data (RTSPMessage * msg, gint channel)
 {
-  if (msg == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
+
+  rtsp_message_unset (msg);
 
   msg->type = RTSP_MESSAGE_DATA;
   msg->type_data.data.channel = channel;
@@ -117,13 +103,51 @@ rtsp_message_init_data (gint channel, RTSPMessage * msg)
   return RTSP_OK;
 }
 
+RTSPResult
+rtsp_message_unset (RTSPMessage * msg)
+{
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
+
+  msg->type = RTSP_MESSAGE_INVALID;
+  msg->type_data.request.method = RTSP_INVALID;
+  g_free (msg->type_data.request.uri);
+  msg->type_data.request.uri = NULL;
+
+  msg->type_data.response.code = RTSP_STS_INVALID;
+  g_free (msg->type_data.response.reason);
+  msg->type_data.response.reason = NULL;
+
+  if (msg->hdr_fields != NULL)
+    g_hash_table_destroy (msg->hdr_fields);
+  msg->hdr_fields = NULL;
+
+  g_free (msg->body);
+  msg->body = NULL;
+  msg->body_size = 0;
+
+  return RTSP_OK;
+}
+
+RTSPResult
+rtsp_message_free (RTSPMessage * msg)
+{
+  RTSPResult res;
+
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
+
+  res = rtsp_message_unset (msg);
+  if (res == RTSP_OK)
+    g_free (msg);
+
+  return res;
+}
 
 RTSPResult
 rtsp_message_add_header (RTSPMessage * msg, RTSPHeaderField field,
     gchar * value)
 {
-  if (msg == NULL || value == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
+  g_return_val_if_fail (value != NULL, RTSP_EINVAL);
 
   g_hash_table_insert (msg->hdr_fields, GINT_TO_POINTER (field),
       g_strdup (value));
@@ -134,8 +158,7 @@ rtsp_message_add_header (RTSPMessage * msg, RTSPHeaderField field,
 RTSPResult
 rtsp_message_remove_header (RTSPMessage * msg, RTSPHeaderField field)
 {
-  if (msg == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
 
   g_hash_table_remove (msg->hdr_fields, GINT_TO_POINTER (field));
 
@@ -148,8 +171,8 @@ rtsp_message_get_header (RTSPMessage * msg, RTSPHeaderField field,
 {
   gchar *val;
 
-  if (msg == NULL || value == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
+  g_return_val_if_fail (value != NULL, RTSP_EINVAL);
 
   val = g_hash_table_lookup (msg->hdr_fields, GINT_TO_POINTER (field));
   if (val == NULL)
@@ -163,8 +186,7 @@ rtsp_message_get_header (RTSPMessage * msg, RTSPHeaderField field,
 RTSPResult
 rtsp_message_set_body (RTSPMessage * msg, guint8 * data, guint size)
 {
-  if (msg == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
 
   return rtsp_message_take_body (msg, g_memdup (data, size), size);
 }
@@ -172,8 +194,7 @@ rtsp_message_set_body (RTSPMessage * msg, guint8 * data, guint size)
 RTSPResult
 rtsp_message_take_body (RTSPMessage * msg, guint8 * data, guint size)
 {
-  if (msg == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
 
   if (msg->body)
     g_free (msg->body);
@@ -187,8 +208,9 @@ rtsp_message_take_body (RTSPMessage * msg, guint8 * data, guint size)
 RTSPResult
 rtsp_message_get_body (RTSPMessage * msg, guint8 ** data, guint * size)
 {
-  if (msg == NULL || data == NULL || size == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
+  g_return_val_if_fail (data != NULL, RTSP_EINVAL);
+  g_return_val_if_fail (size != NULL, RTSP_EINVAL);
 
   *data = msg->body;
   *size = msg->body_size;
@@ -199,8 +221,9 @@ rtsp_message_get_body (RTSPMessage * msg, guint8 ** data, guint * size)
 RTSPResult
 rtsp_message_steal_body (RTSPMessage * msg, guint8 ** data, guint * size)
 {
-  if (msg == NULL || data == NULL || size == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
+  g_return_val_if_fail (data != NULL, RTSP_EINVAL);
+  g_return_val_if_fail (size != NULL, RTSP_EINVAL);
 
   *data = msg->body;
   *size = msg->body_size;
@@ -257,8 +280,7 @@ rtsp_message_dump (RTSPMessage * msg)
   guint8 *data;
   guint size;
 
-  if (msg == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
 
   if (msg->type == RTSP_MESSAGE_REQUEST) {
     g_print ("request message %p\n", msg);
diff --git a/gst/rtsp/rtspmessage.h b/gst/rtsp/rtspmessage.h
index 3488cac1..ced3f840 100644
--- a/gst/rtsp/rtspmessage.h
+++ b/gst/rtsp/rtspmessage.h
@@ -28,6 +28,7 @@ G_BEGIN_DECLS
 
 typedef enum
 {
+  RTSP_MESSAGE_INVALID,
   RTSP_MESSAGE_REQUEST,
   RTSP_MESSAGE_RESPONSE,
   RTSP_MESSAGE_DATA,
@@ -58,15 +59,16 @@ typedef struct _RTSPMessage
 
 } RTSPMessage;
 
-RTSPResult      rtsp_message_new_request        (RTSPMethod method, gchar *uri, RTSPMessage **msg);
-RTSPResult      rtsp_message_init_request       (RTSPMethod method, gchar *uri, RTSPMessage *msg);
+RTSPResult      rtsp_message_new_request        (RTSPMessage **msg, RTSPMethod method, gchar *uri);
+RTSPResult      rtsp_message_init_request       (RTSPMessage *msg, RTSPMethod method, gchar *uri);
 
-RTSPResult      rtsp_message_new_response       (RTSPStatusCode code, gchar *reason, 
-                                                 RTSPMessage *request, RTSPMessage **msg);
-RTSPResult      rtsp_message_init_response      (RTSPStatusCode code, gchar *reason, 
-                                                 RTSPMessage *request, RTSPMessage *msg);
-RTSPResult      rtsp_message_init_data          (gint channel, RTSPMessage *msg);
+RTSPResult      rtsp_message_new_response       (RTSPMessage **msg, RTSPStatusCode code, gchar *reason, 
+                                                 RTSPMessage *request);
+RTSPResult      rtsp_message_init_response      (RTSPMessage *msg, RTSPStatusCode code, gchar *reason, 
+                                                 RTSPMessage *request);
+RTSPResult      rtsp_message_init_data          (RTSPMessage *msg, gint channel);
 
+RTSPResult      rtsp_message_unset              (RTSPMessage *msg);
 RTSPResult      rtsp_message_free               (RTSPMessage *msg);
 
 
diff --git a/gst/rtsp/rtsptransport.c b/gst/rtsp/rtsptransport.c
index 3fdea275..57ec9204 100644
--- a/gst/rtsp/rtsptransport.c
+++ b/gst/rtsp/rtsptransport.c
@@ -27,8 +27,7 @@ rtsp_transport_new (RTSPTransport ** transport)
 {
   RTSPTransport *trans;
 
-  if (transport == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (transport != NULL, RTSP_EINVAL);
 
   trans = g_new0 (RTSPTransport, 1);
 
@@ -40,6 +39,8 @@ rtsp_transport_new (RTSPTransport ** transport)
 RTSPResult
 rtsp_transport_init (RTSPTransport * transport)
 {
+  g_return_val_if_fail (transport != NULL, RTSP_EINVAL);
+
   g_free (transport->destination);
   g_free (transport->source);
   g_free (transport->ssrc);
@@ -83,8 +84,8 @@ rtsp_transport_parse (gchar * str, RTSPTransport * transport)
   gchar **split, *down;
   gint i;
 
-  if (str == NULL || transport == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (transport != NULL, RTSP_EINVAL);
+  g_return_val_if_fail (str != NULL, RTSP_EINVAL);
 
   rtsp_transport_init (transport);
 
@@ -141,6 +142,8 @@ rtsp_transport_parse (gchar * str, RTSPTransport * transport)
 RTSPResult
 rtsp_transport_free (RTSPTransport * transport)
 {
+  g_return_val_if_fail (transport != NULL, RTSP_EINVAL);
+
   rtsp_transport_init (transport);
   g_free (transport);
   return RTSP_OK;
diff --git a/gst/rtsp/rtspurl.c b/gst/rtsp/rtspurl.c
index c5e429f7..dc861fd0 100644
--- a/gst/rtsp/rtspurl.c
+++ b/gst/rtsp/rtspurl.c
@@ -34,10 +34,10 @@ rtsp_url_parse (const gchar * urlstr, RTSPUrl ** url)
   RTSPUrl *res;
   gchar *p, *slash, *at, *col;
 
-  res = g_new0 (RTSPUrl, 1);
+  g_return_val_if_fail (urlstr != NULL, RTSP_EINVAL);
+  g_return_val_if_fail (url != NULL, RTSP_EINVAL);
 
-  if (urlstr == NULL)
-    goto invalid;
+  res = g_new0 (RTSPUrl, 1);
 
   p = (gchar *) urlstr;
   if (g_str_has_prefix (p, RTSP_PROTO)) {
diff --git a/gst/rtsp/sdpmessage.c b/gst/rtsp/sdpmessage.c
index 5ab9b85f..23aab927 100644
--- a/gst/rtsp/sdpmessage.c
+++ b/gst/rtsp/sdpmessage.c
@@ -76,8 +76,7 @@ sdp_message_new (SDPMessage ** msg)
 {
   SDPMessage *newmsg;
 
-  if (msg == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
 
   newmsg = g_new0 (SDPMessage, 1);
 
@@ -89,8 +88,7 @@ sdp_message_new (SDPMessage ** msg)
 RTSPResult
 sdp_message_init (SDPMessage * msg)
 {
-  if (msg == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
 
   FREE_STRING (msg->version);
   FREE_STRING (msg->origin.username);
@@ -123,8 +121,7 @@ sdp_message_init (SDPMessage * msg)
 RTSPResult
 sdp_message_clean (SDPMessage * msg)
 {
-  if (msg == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
 
   FREE_ARRAY (msg->emails);
   FREE_ARRAY (msg->phones);
@@ -140,8 +137,7 @@ sdp_message_clean (SDPMessage * msg)
 RTSPResult
 sdp_message_free (SDPMessage * msg)
 {
-  if (msg == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
 
   sdp_message_clean (msg);
 
@@ -156,8 +152,7 @@ sdp_media_new (SDPMedia ** media)
 {
   SDPMedia *newmedia;
 
-  if (media == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (media != NULL, RTSP_EINVAL);
 
   newmedia = g_new0 (SDPMedia, 1);
 
@@ -169,8 +164,7 @@ sdp_media_new (SDPMedia ** media)
 RTSPResult
 sdp_media_init (SDPMedia * media)
 {
-  if (media == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (media != NULL, RTSP_EINVAL);
 
   FREE_STRING (media->media);
   media->port = 0;
@@ -578,8 +572,9 @@ sdp_message_parse_buffer (guint8 * data, guint size, SDPMessage * msg)
   gchar buffer[4096];
   gint idx = 0;
 
-  if (msg == NULL || data == NULL || size == 0)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
+  g_return_val_if_fail (data != NULL, RTSP_EINVAL);
+  g_return_val_if_fail (size != 0, RTSP_EINVAL);
 
   c.state = SDP_SESSION;
   c.msg = msg;
@@ -651,8 +646,7 @@ print_media (SDPMedia * media)
 RTSPResult
 sdp_message_dump (SDPMessage * msg)
 {
-  if (msg == NULL)
-    return RTSP_EINVAL;
+  g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
 
   g_print ("sdp packet %p:\n", msg);
   g_print (" version:       '%s'\n", msg->version);
@@ -666,6 +660,7 @@ sdp_message_dump (SDPMessage * msg)
   g_print (" session_name:  '%s'\n", msg->session_name);
   g_print (" information:   '%s'\n", msg->information);
   g_print (" uri:           '%s'\n", msg->uri);
+
   if (msg->emails->len > 0) {
     gint i;
 
@@ -710,7 +705,5 @@ sdp_message_dump (SDPMessage * msg)
       print_media (&g_array_index (msg->medias, SDPMedia, i));
     }
   }
-
-
   return RTSP_OK;
 }