diff --git a/ChangeLog b/ChangeLog
index c2c6513e99..64bf5c5f28 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,13 @@
+2006-08-22  Wim Taymans  <wim@fluendo.com>
+
+	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
+	(gst_rtspsrc_send), (gst_rtspsrc_close), (gst_rtspsrc_play),
+	(gst_rtspsrc_pause):
+	* gst/rtsp/gstrtspsrc.h:
+	* sys/oss/gstosssink.c: (gst_oss_sink_open),
+	(gst_oss_sink_prepare), (gst_oss_sink_unprepare):
+	Small documentation updates.
+
 2006-08-22  Wim Taymans  <wim@fluendo.com>
 
 	* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index 10bbedab6b..b32da110e7 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -33,7 +33,7 @@
  * </para>
  * <para>
  * rtspsrc currently understands SDP as the format of the session description.
- * For each stream listed in the SDP a new rtp_stream%d pad will be created
+ * For each stream listed in the SDP a new rtp_stream&perc;d pad will be created
  * with caps derived from the SDP media description. This is a caps of mime type
  * "application/x-rtp" that can be connected to any available rtp depayloader
  * element. 
@@ -57,7 +57,7 @@
  * </para>
  * </refsect2>
  *
- * Last reviewed on 2006-06-20 (0.10.4)
+ * Last reviewed on 2006-08-18 (0.10.5)
  */
 
 #ifdef HAVE_CONFIG_H
@@ -553,7 +553,7 @@ gst_rtspsrc_media_to_caps (SDPMedia * media)
         if (valpos) {
           /* we have a '=' and thus a value, remove the '=' with \0 */
           *valpos = '\0';
-          /* value is everything between '=' and ';' */
+          /* value is everything between '=' and ';'. FIXME, strip? */
           val = g_strstrip (valpos + 1);
         } else {
           /* simple <param>;.. is translated into <param>=1;... */
@@ -962,6 +962,7 @@ gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
 
   return TRUE;
 
+  /* ERRORS */
 send_error:
   {
     GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
@@ -1329,6 +1330,7 @@ gst_rtspsrc_close (GstRTSPSrc * src)
 
   return TRUE;
 
+  /* ERRORS */
 create_request_failed:
   {
     GST_ELEMENT_ERROR (src, LIBRARY, INIT,
@@ -1377,6 +1379,7 @@ gst_rtspsrc_play (GstRTSPSrc * src)
 
   return TRUE;
 
+  /* ERRORS */
 create_request_failed:
   {
     GST_ELEMENT_ERROR (src, LIBRARY, INIT,
@@ -1412,6 +1415,7 @@ gst_rtspsrc_pause (GstRTSPSrc * src)
 
   return TRUE;
 
+  /* ERRORS */
 create_request_failed:
   {
     GST_ELEMENT_ERROR (src, LIBRARY, INIT,
diff --git a/gst/rtsp/gstrtspsrc.h b/gst/rtsp/gstrtspsrc.h
index 424c512a90..a2e9bd2dcc 100644
--- a/gst/rtsp/gstrtspsrc.h
+++ b/gst/rtsp/gstrtspsrc.h
@@ -42,7 +42,14 @@ G_BEGIN_DECLS
 typedef struct _GstRTSPSrc GstRTSPSrc;
 typedef struct _GstRTSPSrcClass GstRTSPSrcClass;
 
-/* flags with allowed protocols */
+/**
+ * GstRTSPProto:
+ * @GST_RTSP_PROTO_UDP_UNICAST: Use unicast UDP transfer.
+ * @GST_RTSP_PROTO_UDP_MULTICAST: Use multicast UDP transfer
+ * @GST_RTSP_PROTO_TCP: Use TCP transfer.
+ *
+ * Flags with allowed protocols for the datatransfer.
+ */
 typedef enum
 {
   GST_RTSP_PROTO_UDP_UNICAST    = (1 << 0),
diff --git a/sys/oss/gstosssink.c b/sys/oss/gstosssink.c
index 9f2902a045..7e1e32bb99 100644
--- a/sys/oss/gstosssink.c
+++ b/sys/oss/gstosssink.c
@@ -396,12 +396,12 @@ gst_oss_sink_open (GstAudioSink * asink)
 
   return TRUE;
 
+  /* ERRORS */
 busy:
   {
     GST_ELEMENT_ERROR (oss, RESOURCE, BUSY, (NULL), (NULL));
     return FALSE;
   }
-
 open_failed:
   {
     GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE, (NULL), GST_ERROR_SYSTEM);
@@ -465,6 +465,7 @@ gst_oss_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
 
   return TRUE;
 
+  /* ERRORS */
 non_block:
   {
     GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
@@ -499,6 +500,7 @@ gst_oss_sink_unprepare (GstAudioSink * asink)
 
   return TRUE;
 
+  /* ERRORS */
 couldnt_close:
   {
     GST_DEBUG_OBJECT (asink, "Could not close the audio device");