From e5019de80d307cf9b6d906fd5a51c395bb70aa69 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 23 Sep 2013 15:36:32 +0200 Subject: [PATCH] docs: update docs with 1.0 element names --- gst/rtpmanager/gstrtpbin.c | 10 +++++----- gst/rtpmanager/gstrtpjitterbuffer.c | 12 ++++++------ gst/rtpmanager/gstrtpptdemux.c | 6 +++--- gst/rtpmanager/gstrtpsession.c | 20 ++++++++++---------- gst/rtpmanager/gstrtpssrcdemux.c | 6 +++--- 5 files changed, 27 insertions(+), 27 deletions(-) diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c index 4236cd160b..e615c94e11 100644 --- a/gst/rtpmanager/gstrtpbin.c +++ b/gst/rtpmanager/gstrtpbin.c @@ -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; diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c index 2c37fc3b45..b0640436a9 100644 --- a/gst/rtpmanager/gstrtpjitterbuffer.c +++ b/gst/rtpmanager/gstrtpjitterbuffer.c @@ -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. * * * Example pipelines * |[ - * 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; } } diff --git a/gst/rtpmanager/gstrtpptdemux.c b/gst/rtpmanager/gstrtpptdemux.c index dbc0016ab2..2c80ae0337 100644 --- a/gst/rtpmanager/gstrtpptdemux.c +++ b/gst/rtpmanager/gstrtpptdemux.c @@ -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 @@ * * Example pipelines * |[ - * 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. * diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c index a445be97e5..dc1c34d181 100644 --- a/gst/rtpmanager/gstrtpsession.c +++ b/gst/rtpmanager/gstrtpsession.c @@ -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 @@ * * * - * 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 @@ * * Example pipelines * |[ - * 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; } } diff --git a/gst/rtpmanager/gstrtpssrcdemux.c b/gst/rtpmanager/gstrtpssrcdemux.c index b04d4365e9..75730a4438 100644 --- a/gst/rtpmanager/gstrtpssrcdemux.c +++ b/gst/rtpmanager/gstrtpssrcdemux.c @@ -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 @@ * * Example pipelines * |[ - * 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. *