From 0f38451f208637683b47694f74bf09da10d26a3a Mon Sep 17 00:00:00 2001
From: Wim Taymans <wim.taymans@gmail.com>
Date: Tue, 22 Aug 2006 16:45:37 +0000
Subject: Small documentation updates.

Original commit message from CVS:
* 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.
---
 gst/rtsp/gstrtspsrc.c | 10 +++++++---
 gst/rtsp/gstrtspsrc.h |  9 ++++++++-
 2 files changed, 15 insertions(+), 4 deletions(-)

(limited to 'gst')

diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index 10bbedab..b32da110 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 424c512a..a2e9bd2d 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),
-- 
cgit