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authorWim Taymans <wim.taymans@gmail.com>2006-08-22 16:45:37 +0000
committerWim Taymans <wim.taymans@gmail.com>2006-08-22 16:45:37 +0000
commit0f38451f208637683b47694f74bf09da10d26a3a (patch)
treefb974d5fb55dfa7e497dc088bca5cb6a954c0e3e /gst/rtsp
parent1eff78685b38ebf0c0b589535066da428a708feb (diff)
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.
Diffstat (limited to 'gst/rtsp')
-rw-r--r--gst/rtsp/gstrtspsrc.c10
-rw-r--r--gst/rtsp/gstrtspsrc.h9
2 files changed, 15 insertions, 4 deletions
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),