diff options
author | Wim Taymans <wim.taymans@gmail.com> | 2006-06-20 14:57:09 +0000 |
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committer | Wim Taymans <wim.taymans@gmail.com> | 2006-06-20 14:57:09 +0000 |
commit | bfd2b35dda34bd54346adf0293f9947dfa32ec06 (patch) | |
tree | 6094c5e93b19a19d2df38b37c7ff7f9c7dc5b96d | |
parent | b24f97c6950a0760762ff89a25621bce48ab9efe (diff) |
Added documentation for the rtsp plugin. Fixes #345393.
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.
-rw-r--r-- | ChangeLog | 10 | ||||
-rw-r--r-- | docs/plugins/Makefile.am | 2 | ||||
-rw-r--r-- | docs/plugins/gst-plugins-good-plugins-docs.sgml | 2 | ||||
-rw-r--r-- | docs/plugins/gst-plugins-good-plugins-sections.txt | 31 | ||||
-rw-r--r-- | gst/rtsp/gstrtpdec.c | 18 | ||||
-rw-r--r-- | gst/rtsp/gstrtspsrc.c | 43 | ||||
-rw-r--r-- | gst/rtsp/gstrtspsrc.h | 10 |
7 files changed, 107 insertions, 9 deletions
@@ -1,5 +1,15 @@ 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), (rtsp_connection_close), (rtsp_connection_free): Use better G_OS_* macros. Fixes #345301 some more. diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am index 6e3589f7..cb3decc4 100644 --- a/docs/plugins/Makefile.am +++ b/docs/plugins/Makefile.am @@ -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 \ diff --git a/docs/plugins/gst-plugins-good-plugins-docs.sgml b/docs/plugins/gst-plugins-good-plugins-docs.sgml index b77f46f6..41cdc902 100644 --- a/docs/plugins/gst-plugins-good-plugins-docs.sgml +++ b/docs/plugins/gst-plugins-good-plugins-docs.sgml @@ -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" /> diff --git a/docs/plugins/gst-plugins-good-plugins-sections.txt b/docs/plugins/gst-plugins-good-plugins-sections.txt index 81007495..ac57b606 100644 --- a/docs/plugins/gst-plugins-good-plugins-sections.txt +++ b/docs/plugins/gst-plugins-good-plugins-sections.txt @@ -157,6 +157,37 @@ 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 <TITLE>smokedec</TITLE> diff --git a/gst/rtsp/gstrtpdec.c b/gst/rtsp/gstrtpdec.c index c22d4cca..39917423 100644 --- a/gst/rtsp/gstrtpdec.c +++ b/gst/rtsp/gstrtpdec.c @@ -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); diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index 4554dc03..c909415f 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -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" diff --git a/gst/rtsp/gstrtspsrc.h b/gst/rtsp/gstrtspsrc.h index 32a7735b..fa19677d 100644 --- a/gst/rtsp/gstrtspsrc.h +++ b/gst/rtsp/gstrtspsrc.h @@ -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__ */ |