mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2025-01-11 09:55:36 +00:00
Code Issues 1 Projects Releases Wiki Activity

docs: update docs with 1.0 element names

Browse source
This commit is contained in:
Wim Taymans 2013-09-23 15:36:32 +02:00
parent 3c69d65b85
commit e5019de80d
5 changed files with 27 additions and 27 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

10
gst/rtpmanager/gstrtpbin.c
View file

@ -633,13 +633,13 @@ create_session (GstRtpBin * rtpbin, gint id)
/* ERRORS */
no_session:
{
g_warning ("rtpbin: could not create gstrtpsession element");
g_warning ("rtpbin: could not create rtpsession element");
return NULL;
}
no_demux:
{
gst_object_unref (session);
g_warning ("rtpbin: could not create gstrtpssrcdemux element");
g_warning ("rtpbin: could not create rtpssrcdemux element");
return NULL;
}
}
@ -1465,13 +1465,13 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
/* ERRORS */
no_jitterbuffer:
{
g_warning ("rtpbin: could not create gstrtpjitterbuffer element");
g_warning ("rtpbin: could not create rtpjitterbuffer element");
return NULL;
}
no_demux:
{
gst_object_unref (buffer);
g_warning ("rtpbin: could not create gstrtpptdemux element");
g_warning ("rtpbin: could not create rtpptdemux element");
return NULL;
}
}
@ -2598,7 +2598,7 @@ new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
stream->demux_ptchange_sig = g_signal_connect (stream->demux,
"payload-type-change", (GCallback) payload_type_change, session);
} else {
/* add gstrtpjitterbuffer src pad to pads */
/* add rtpjitterbuffer src pad to pads */
GstElementClass *klass;
GstPadTemplate *templ;
gchar *padname;

12
gst/rtpmanager/gstrtpjitterbuffer.c
View file

@ -24,7 +24,7 @@
*/
/**
* SECTION:element-gstrtpjitterbuffer
* SECTION:element-rtpjitterbuffer
*
* This element reorders and removes duplicate RTP packets as they are received
* from a network source. It will also wait for missing packets up to a
@ -39,12 +39,12 @@
* when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal.
* To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal.
*
* This element will automatically be used inside gstrtpbin.
* This element will automatically be used inside rtpbin.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! gstrtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
* gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
* ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is
* inserted into the pipeline to smooth out network jitter and to reorder the
* out-of-order RTP packets.
@ -593,7 +593,7 @@ gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
GST_DEBUG_CATEGORY_INIT
(rtpjitterbuffer_debug, "gstrtpjitterbuffer", 0, "RTP Jitter Buffer");
(rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
}
static void
@ -801,12 +801,12 @@ gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
/* ERRORS */
wrong_template:
{
g_warning ("gstrtpjitterbuffer: this is not our template");
g_warning ("rtpjitterbuffer: this is not our template");
return NULL;
}
exists:
{
g_warning ("gstrtpjitterbuffer: pad already requested");
g_warning ("rtpjitterbuffer: pad already requested");
return NULL;
}
}

6
gst/rtpmanager/gstrtpptdemux.c
View file

@ -24,9 +24,9 @@
*/
/**
* SECTION:element-gstrtpptdemux
* SECTION:element-rtpptdemux
*
* gstrtpptdemux acts as a demuxer for RTP packets based on the payload type of
* rtpptdemux acts as a demuxer for RTP packets based on the payload type of
* the packets. Its main purpose is to allow an application to easily receive
* and decode an RTP stream with multiple payload types.
*
@ -42,7 +42,7 @@
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch-1.0 udpsrc caps="application/x-rtp" ! gstrtpptdemux ! fakesink
* gst-launch-1.0 udpsrc caps="application/x-rtp" ! rtpptdemux ! fakesink
* ]| Takes an RTP stream and send the RTP packets with the first detected
* payload type to fakesink, discarding the other payload types.
* </refsect2>

20
gst/rtpmanager/gstrtpsession.c
View file

@ -18,8 +18,8 @@
*/
/**
* SECTION:element-gstrtpsession
* @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpptdemux, gstrtpssrcdemux
* SECTION:element-rtpsession
* @see_also: rtpjitterbuffer, rtpbin, rtpptdemux, rtpssrcdemux
*
* The RTP session manager models one participant with a unique SSRC in an RTP
* session. This session can be used to send and receive RTP and RTCP packets.
@ -42,7 +42,7 @@
* </listitem>
* </itemizedlist>
*
* The gstrtpsession will not demux packets based on SSRC or payload type, nor will
* The rtpsession will not demux packets based on SSRC or payload type, nor will
* it correct for packet reordering and jitter. Use #GstRtpsSrcDemux,
* #GstRtpPtDemux and GstRtpJitterBuffer in addition to #GstRtpSession to
* perform these tasks. It is usually a good idea to use #GstRtpBin, which
@ -76,13 +76,13 @@
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink
* gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink
* ]| Receive theora RTP packets from port 5000 and send them to the depayloader,
* decoder and display. Note that the application/x-rtp caps on udpsrc should be
* configured based on some negotiation process such as RTSP for this pipeline
* to work correctly.
* |[
* gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession name=session \
* gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession name=session \
* .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink \
* udpsrc port=5001 caps="application/x-rtcp" ! session.recv_rtcp_sink
* ]| Receive theora RTP packets from port 5000 and send them to the depayloader,
@ -92,11 +92,11 @@
* configured based on some negotiation process such as RTSP for this pipeline
* to work correctly.
* |[
* gst-launch-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession .send_rtp_src ! udpsink port=5000
* gst-launch-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession .send_rtp_src ! udpsink port=5000
* ]| Send theora RTP packets through the session manager and out on UDP port
* 5000.
* |[
* gst-launch-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession name=session .send_rtp_src \
* gst-launch-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession name=session .send_rtp_src \
* ! udpsink port=5000 session.send_rtcp_src ! udpsink port=5001
* ]| Send theora RTP packets through the session manager and out on UDP port
* 5000. Send RTCP packets on port 5001. Note that this pipeline will not preroll
@ -2269,13 +2269,13 @@ gst_rtp_session_request_new_pad (GstElement * element,
wrong_template:
{
GST_RTP_SESSION_UNLOCK (rtpsession);
g_warning ("gstrtpsession: this is not our template");
g_warning ("rtpsession: this is not our template");
return NULL;
}
exists:
{
GST_RTP_SESSION_UNLOCK (rtpsession);
g_warning ("gstrtpsession: pad already requested");
g_warning ("rtpsession: pad already requested");
return NULL;
}
}
@ -2313,7 +2313,7 @@ gst_rtp_session_release_pad (GstElement * element, GstPad * pad)
wrong_pad:
{
GST_RTP_SESSION_UNLOCK (rtpsession);
g_warning ("gstrtpsession: asked to release an unknown pad");
g_warning ("rtpsession: asked to release an unknown pad");
return;
}
}

6
gst/rtpmanager/gstrtpssrcdemux.c
View file

@ -20,9 +20,9 @@
*/
/**
* SECTION:element-gstrtpssrcdemux
* SECTION:element-rtpssrcdemux
*
* gstrtpssrcdemux acts as a demuxer for RTP packets based on the SSRC of the
* rtpssrcdemux acts as a demuxer for RTP packets based on the SSRC of the
* packets. Its main purpose is to allow an application to easily receive and
* decode an RTP stream with multiple SSRCs.
*
@ -32,7 +32,7 @@
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch-1.0 udpsrc caps="application/x-rtp" ! gstrtpssrcdemux ! fakesink
* gst-launch-1.0 udpsrc caps="application/x-rtp" ! rtpssrcdemux ! fakesink
* ]| Takes an RTP stream and send the RTP packets with the first detected SSRC
* to fakesink, discarding the other SSRCs.
* </refsect2>