diff options
Diffstat (limited to 'gst/rtsp')
| -rw-r--r-- | gst/rtsp/gstrtpdec.c | 12 | ||||
| -rw-r--r-- | gst/rtsp/gstrtspsrc.c | 27 |
2 files changed, 14 insertions, 25 deletions
diff --git a/gst/rtsp/gstrtpdec.c b/gst/rtsp/gstrtpdec.c index 0adfc7aa..c6a942e4 100644 --- a/gst/rtsp/gstrtpdec.c +++ b/gst/rtsp/gstrtpdec.c @@ -44,11 +44,7 @@ /** * 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) */ @@ -252,8 +248,8 @@ gst_rtp_dec_marshal_BOXED__UINT_UINT (GClosure * closure, data2 = closure->data; } callback = - (GMarshalFunc_BOXED__UINT_UINT) (marshal_data ? marshal_data : cc-> - callback); + (GMarshalFunc_BOXED__UINT_UINT) (marshal_data ? marshal_data : + cc->callback); v_return = callback (data1, g_marshal_value_peek_uint (param_values + 1), @@ -285,8 +281,8 @@ gst_rtp_dec_marshal_VOID__UINT_UINT (GClosure * closure, data2 = closure->data; } callback = - (GMarshalFunc_VOID__UINT_UINT) (marshal_data ? marshal_data : cc-> - callback); + (GMarshalFunc_VOID__UINT_UINT) (marshal_data ? marshal_data : + cc->callback); callback (data1, g_marshal_value_peek_uint (param_values + 1), diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index 79455cd1..e677c991 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -43,43 +43,36 @@ /** * 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> + * protocols can be controlled with the #GstRTSPSrc:protocols property. + * * 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 currently fully implemented with the gstrtpbin in the * gst-plugins-bad module. - * </para> - * <para> + * * rtspsrc acts like a live source and will therefore only generate data in the * PLAYING state. - * </para> + * + * <refsect2> * <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 raw RTP packets to a fakesink. - * </para> + * ]| Establish a connection to an RTSP server and send the raw RTP packets to a + * fakesink. * </refsect2> * * Last reviewed on 2006-08-18 (0.10.5) |