Commit e5019de8 authored by Wim Taymans's avatar Wim Taymans

docs: update docs with 1.0 element names

parent 3c69d65b
...@@ -633,13 +633,13 @@ create_session (GstRtpBin * rtpbin, gint id) ...@@ -633,13 +633,13 @@ create_session (GstRtpBin * rtpbin, gint id)
/* ERRORS */ /* ERRORS */
no_session: no_session:
{ {
g_warning ("rtpbin: could not create gstrtpsession element"); g_warning ("rtpbin: could not create rtpsession element");
return NULL; return NULL;
} }
no_demux: no_demux:
{ {
gst_object_unref (session); gst_object_unref (session);
g_warning ("rtpbin: could not create gstrtpssrcdemux element"); g_warning ("rtpbin: could not create rtpssrcdemux element");
return NULL; return NULL;
} }
} }
...@@ -1465,13 +1465,13 @@ create_stream (GstRtpBinSession * session, guint32 ssrc) ...@@ -1465,13 +1465,13 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
/* ERRORS */ /* ERRORS */
no_jitterbuffer: no_jitterbuffer:
{ {
g_warning ("rtpbin: could not create gstrtpjitterbuffer element"); g_warning ("rtpbin: could not create rtpjitterbuffer element");
return NULL; return NULL;
} }
no_demux: no_demux:
{ {
gst_object_unref (buffer); gst_object_unref (buffer);
g_warning ("rtpbin: could not create gstrtpptdemux element"); g_warning ("rtpbin: could not create rtpptdemux element");
return NULL; return NULL;
} }
} }
...@@ -2598,7 +2598,7 @@ new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad, ...@@ -2598,7 +2598,7 @@ new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
stream->demux_ptchange_sig = g_signal_connect (stream->demux, stream->demux_ptchange_sig = g_signal_connect (stream->demux,
"payload-type-change", (GCallback) payload_type_change, session); "payload-type-change", (GCallback) payload_type_change, session);
} else { } else {
/* add gstrtpjitterbuffer src pad to pads */ /* add rtpjitterbuffer src pad to pads */
GstElementClass *klass; GstElementClass *klass;
GstPadTemplate *templ; GstPadTemplate *templ;
gchar *padname; gchar *padname;
......
...@@ -24,7 +24,7 @@ ...@@ -24,7 +24,7 @@
*/ */
/** /**
* SECTION:element-gstrtpjitterbuffer * SECTION:element-rtpjitterbuffer
* *
* This element reorders and removes duplicate RTP packets as they are received * 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 * from a network source. It will also wait for missing packets up to a
...@@ -39,12 +39,12 @@ ...@@ -39,12 +39,12 @@
* when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal. * 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. * 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> * <refsect2>
* <title>Example pipelines</title> * <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 * ]| 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 * inserted into the pipeline to smooth out network jitter and to reorder the
* out-of-order RTP packets. * out-of-order RTP packets.
...@@ -593,7 +593,7 @@ gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass) ...@@ -593,7 +593,7 @@ gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active); klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
GST_DEBUG_CATEGORY_INIT GST_DEBUG_CATEGORY_INIT
(rtpjitterbuffer_debug, "gstrtpjitterbuffer", 0, "RTP Jitter Buffer"); (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
} }
static void static void
...@@ -801,12 +801,12 @@ gst_rtp_jitter_buffer_request_new_pad (GstElement * element, ...@@ -801,12 +801,12 @@ gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
/* ERRORS */ /* ERRORS */
wrong_template: wrong_template:
{ {
g_warning ("gstrtpjitterbuffer: this is not our template"); g_warning ("rtpjitterbuffer: this is not our template");
return NULL; return NULL;
} }
exists: exists:
{ {
g_warning ("gstrtpjitterbuffer: pad already requested"); g_warning ("rtpjitterbuffer: pad already requested");
return NULL; return NULL;
} }
} }
......
...@@ -24,9 +24,9 @@ ...@@ -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 * the packets. Its main purpose is to allow an application to easily receive
* and decode an RTP stream with multiple payload types. * and decode an RTP stream with multiple payload types.
* *
...@@ -42,7 +42,7 @@ ...@@ -42,7 +42,7 @@
* <refsect2> * <refsect2>
* <title>Example pipelines</title> * <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 * ]| Takes an RTP stream and send the RTP packets with the first detected
* payload type to fakesink, discarding the other payload types. * payload type to fakesink, discarding the other payload types.
* </refsect2> * </refsect2>
......
...@@ -18,8 +18,8 @@ ...@@ -18,8 +18,8 @@
*/ */
/** /**
* SECTION:element-gstrtpsession * SECTION:element-rtpsession
* @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpptdemux, gstrtpssrcdemux * @see_also: rtpjitterbuffer, rtpbin, rtpptdemux, rtpssrcdemux
* *
* The RTP session manager models one participant with a unique SSRC in an RTP * 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. * session. This session can be used to send and receive RTP and RTCP packets.
...@@ -42,7 +42,7 @@ ...@@ -42,7 +42,7 @@
* </listitem> * </listitem>
* </itemizedlist> * </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, * it correct for packet reordering and jitter. Use #GstRtpsSrcDemux,
* #GstRtpPtDemux and GstRtpJitterBuffer in addition to #GstRtpSession to * #GstRtpPtDemux and GstRtpJitterBuffer in addition to #GstRtpSession to
* perform these tasks. It is usually a good idea to use #GstRtpBin, which * perform these tasks. It is usually a good idea to use #GstRtpBin, which
...@@ -76,13 +76,13 @@ ...@@ -76,13 +76,13 @@
* <refsect2> * <refsect2>
* <title>Example pipelines</title> * <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, * ]| 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 * 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 * configured based on some negotiation process such as RTSP for this pipeline
* to work correctly. * 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 \ * .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink \
* udpsrc port=5001 caps="application/x-rtcp" ! session.recv_rtcp_sink * udpsrc port=5001 caps="application/x-rtcp" ! session.recv_rtcp_sink
* ]| Receive theora RTP packets from port 5000 and send them to the depayloader, * ]| Receive theora RTP packets from port 5000 and send them to the depayloader,
...@@ -92,11 +92,11 @@ ...@@ -92,11 +92,11 @@
* configured based on some negotiation process such as RTSP for this pipeline * configured based on some negotiation process such as RTSP for this pipeline
* to work correctly. * 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 * ]| Send theora RTP packets through the session manager and out on UDP port
* 5000. * 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 * ! udpsink port=5000 session.send_rtcp_src ! udpsink port=5001
* ]| Send theora RTP packets through the session manager and out on UDP port * ]| 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 * 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, ...@@ -2269,13 +2269,13 @@ gst_rtp_session_request_new_pad (GstElement * element,
wrong_template: wrong_template:
{ {
GST_RTP_SESSION_UNLOCK (rtpsession); GST_RTP_SESSION_UNLOCK (rtpsession);
g_warning ("gstrtpsession: this is not our template"); g_warning ("rtpsession: this is not our template");
return NULL; return NULL;
} }
exists: exists:
{ {
GST_RTP_SESSION_UNLOCK (rtpsession); GST_RTP_SESSION_UNLOCK (rtpsession);
g_warning ("gstrtpsession: pad already requested"); g_warning ("rtpsession: pad already requested");
return NULL; return NULL;
} }
} }
...@@ -2313,7 +2313,7 @@ gst_rtp_session_release_pad (GstElement * element, GstPad * pad) ...@@ -2313,7 +2313,7 @@ gst_rtp_session_release_pad (GstElement * element, GstPad * pad)
wrong_pad: wrong_pad:
{ {
GST_RTP_SESSION_UNLOCK (rtpsession); GST_RTP_SESSION_UNLOCK (rtpsession);
g_warning ("gstrtpsession: asked to release an unknown pad"); g_warning ("rtpsession: asked to release an unknown pad");
return; return;
} }
} }
......
...@@ -20,9 +20,9 @@ ...@@ -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 * packets. Its main purpose is to allow an application to easily receive and
* decode an RTP stream with multiple SSRCs. * decode an RTP stream with multiple SSRCs.
* *
...@@ -32,7 +32,7 @@ ...@@ -32,7 +32,7 @@
* <refsect2> * <refsect2>
* <title>Example pipelines</title> * <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 * ]| Takes an RTP stream and send the RTP packets with the first detected SSRC
* to fakesink, discarding the other SSRCs. * to fakesink, discarding the other SSRCs.
* </refsect2> * </refsect2>
......
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