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Added documentation for the rtsp plugin. Fixes #345393.

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Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init):
* gst/rtsp/gstrtspsrc.c:
* gst/rtsp/gstrtspsrc.h:
Added documentation for the rtsp plugin. Fixes #345393.
This commit is contained in:
Wim Taymans 2006-06-20 14:57:09 +00:00
parent b24f97c695
commit bfd2b35dda
7 changed files with 107 additions and 9 deletions
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10
ChangeLog
View file

@ -1,3 +1,13 @@
2006-06-20 Wim Taymans <wim@fluendo.com>
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init):
* gst/rtsp/gstrtspsrc.c:
* gst/rtsp/gstrtspsrc.h:
Added documentation for the rtsp plugin. Fixes #345393.
2006-06-20 Wim Taymans <wim@fluendo.com>
* gst/rtsp/rtspconnection.c: (inet_aton), (rtsp_connection_send),

2
docs/plugins/Makefile.am
View file

@ -82,6 +82,8 @@ EXTRA_HFILES = \
$(top_srcdir)/gst/level/gstlevel.h \
$(top_srcdir)/gst/goom/gstgoom.h \
$(top_srcdir)/gst/id3demux/gstid3demux.h \
$(top_srcdir)/gst/rtsp/gstrtpdec.h \
$(top_srcdir)/gst/rtsp/gstrtspsrc.h \
$(top_srcdir)/gst/wavparse/gstwavparse.h \
$(top_srcdir)/ext/cairo/gsttimeoverlay.h \
$(top_srcdir)/ext/cdio/gstcdiocddasrc.h \

2
docs/plugins/gst-plugins-good-plugins-docs.sgml
View file

@ -35,6 +35,8 @@
<xi:include href="xml/element-multiudpsink.xml" />
<xi:include href="xml/element-multipartmux.xml" />
<xi:include href="xml/element-multipartdemux.xml" />
<xi:include href="xml/element-rtspsrc.xml" />
<xi:include href="xml/element-rtpdec.xml" />
<xi:include href="xml/element-smokedec.xml" />
<xi:include href="xml/element-smokeenc.xml" />
<xi:include href="xml/element-videobalance.xml" />

31
docs/plugins/gst-plugins-good-plugins-sections.txt
View file

@ -156,6 +156,37 @@ GST_JPEGENC_CLASS
GST_IS_JPEGENC_CLASS
</SECTION>
<SECTION>
<FILE>element-rtspsrc</FILE>
GstRTSPProto
GstRTSPSrc
<TITLE>rtspsrc</TITLE>
<SUBSECTION Standard>
GstRTSPStream
GstRTSPSrcClass
GST_RTSPSRC
GST_IS_RTSPSRC
GST_TYPE_RTSPSRC
gst_rtspsrc_get_type
GST_RTSPSRC_CLASS
GST_IS_RTSPSRC_CLASS
</SECTION>
<SECTION>
<FILE>element-rtpdec</FILE>
GstRTPDec
<TITLE>rtpdec</TITLE>
<SUBSECTION Standard>
gst_rtpdec_plugin_init
GstRTPDecClass
GST_RTPDEC
GST_IS_RTPDEC
GST_TYPE_RTPDEC
gst_rtpdec_get_type
GST_RTPDEC_CLASS
GST_IS_RTPDEC_CLASS
</SECTION>
<SECTION>
<FILE>element-smokedec</FILE>
GstSmokeDec

18
gst/rtsp/gstrtpdec.c
View file

@ -18,6 +18,19 @@
*/
/* Element-Checklist-Version: 5 */
/**
* SECTION:element-rtpdec
*
* <refsect2>
* <para>
* A simple RTP session manager used internally by rtspsrc.
* </para>
* </refsect2>
*
* Last reviewed on 2006-06-20 (0.10.4)
*/
#include "gstrtpdec.h"
GST_DEBUG_CATEGORY (rtpdec_debug);
@ -138,7 +151,10 @@ gst_rtpdec_class_init (gpointer g_class)
gobject_class->set_property = gst_rtpdec_set_property;
gobject_class->get_property = gst_rtpdec_get_property;
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SKIP, g_param_spec_int ("skip", "skip", "skip", G_MININT, G_MAXINT, 0, G_PARAM_READWRITE)); /* CHECKME */
/* FIXME, this is unused and probably copied from somewhere */
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SKIP,
g_param_spec_int ("skip", "Skip", "skip (unused)", G_MININT, G_MAXINT, 0,
G_PARAM_READWRITE));
parent_class = g_type_class_peek_parent (klass);

43
gst/rtsp/gstrtspsrc.c
View file

@ -16,6 +16,49 @@
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-rtspsrc
*
* <refsect2>
* <para>
* Makes a connection to an RTSP server and read the data.
* rtspsrc strictly follows RFC 2326 and therefore does not (yet) support
* RealMedia/Quicktime/Microsoft extensions.
* </para>
* <para>
* RTSP supports transport over TCP or UDP in unicast or multicast mode. By
* default rtspsrc will negotiate a connection in the following order:
* UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed
* protocols can be controlled with the "protocols" property.
* </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
* 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.
* </para>
* <para>
* rtspsrc will internally instantiate an RTP session manager element
* that will handle the RTCP messages to and from the server, jitter removal,
* packet reordering along with providing a clock for the pipeline.
* This feature is however currently not yet implemented.
* </para>
* <para>
* rtspsrc acts like a live source and will therefore only generate data in the
* PLAYING state.
* </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch rtspsrc location=rtsp://some.server/url ! fakesink
* </programlisting>
* Establish a connection to an RTSP server and send the stream to a fakesink.
* </para>
* </refsect2>
*
* Last reviewed on 2006-06-20 (0.10.4)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"

10
gst/rtsp/gstrtspsrc.h
View file

@ -23,9 +23,7 @@
#include <gst/gst.h>
#ifdef __cplusplus
extern "C" {
#endif /* __cplusplus */
G_BEGIN_DECLS
#include "gstrtsp.h"
#include "rtsp.h"
@ -103,10 +101,6 @@ struct _GstRTSPSrcClass {
GType gst_rtspsrc_get_type(void);
#ifdef __cplusplus
}
#endif /* __cplusplus */
G_END_DECLS
#endif /* __GST_RTSPSRC_H__ */