Commit 0f38451f authored by Wim Taymans's avatar Wim Taymans

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.
parent 1eff7868
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),
......
......@@ -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,
......
......@@ -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),
......
......@@ -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");
......
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