From 3a496fd7ebb90d12bad86c6ded97a75e134794f6 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 28 May 2007 16:37:47 +0000 Subject: Rename elements to avoid conflict with farsight elements with the same name. Fixes #430664. Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * gst/rtpmanager/gstrtpbin.c: (create_session), (create_stream), (gst_rtp_bin_class_init), (create_recv_rtp), (create_recv_rtcp), (create_send_rtp), (create_rtcp), (gst_rtp_bin_request_new_pad): * gst/rtpmanager/gstrtpclient.c: (create_stream), (gst_rtp_client_request_new_pad): * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/gstrtpmanager.c: (plugin_init): * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (gst_rtp_session_request_new_pad): * gst/rtpmanager/gstrtpssrcdemux.c: Rename elements to avoid conflict with farsight elements with the same name. Fixes #430664. --- gst/rtpmanager/gstrtpsession.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) (limited to 'gst/rtpmanager/gstrtpsession.c') diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c index 431098d9..3e33cf6a 100644 --- a/gst/rtpmanager/gstrtpsession.c +++ b/gst/rtpmanager/gstrtpsession.c @@ -18,9 +18,9 @@ */ /** - * SECTION:element-rtpsession + * SECTION:element-gstrtpsession * @short_description: an RTP session manager - * @see_also: rtpjitterbuffer, rtpbin, rtpptdemux, rtpssrcdemux + * @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpptdemux, gstrtpssrcdemux * * * @@ -47,20 +47,20 @@ * * * - * The rtpsession will not demux packets based on SSRC or payload type, nor will - * it correct for packet reordering and jitter. Use rtpssrcdemux, rtpptdemux and - * rtpjitterbuffer in addition to rtpsession to perform these tasks. It is - * usually a good idea to use rtpbin, which combines all these features in one + * The gstrtpsession 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 combines all these features in one * element. * * - * To use rtpsession as an RTP receiver, request a recv_rtp_sink pad, which will + * To use gstrtpsession as an RTP receiver, request a recv_rtp_sink pad, which will * automatically create recv_rtp_src pad. Data received on the recv_rtp_sink pad * will be processed in the session and after being validated forwarded on the * recv_rtp_src pad. * * - * To also use rtpsession as an RTCP receiver, request a recv_rtcp_sink pad, + * To also use gstrtpsession as an RTCP receiver, request a recv_rtcp_sink pad, * which will automatically create a sync_src pad. Packets received on the RTCP * pad will be used by the session manager to update the stats and database of * the other participants. SR packets will be forwarded on the sync_src pad @@ -72,7 +72,7 @@ * that should be sent to all participants in the session. * * - * To use rtpsession as a sender, request a send_rtp_sink pad, which will + * To use gstrtpsession as a sender, request a send_rtp_sink pad, which will * automatically create a send_rtp_src pad. The session manager will modify the * SSRC in the RTP packets to its own SSRC and wil forward the packets on the * send_rtp_src pad after updating its internal state. @@ -86,7 +86,7 @@ * Example pipelines * * - * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink + * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession .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 @@ -95,7 +95,7 @@ * * * - * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession name=session \ + * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession name=session \ * .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink \ * udpsrc port=5001 caps="application/x-rtcp" ! session.recv_rtcp_sink * @@ -108,24 +108,24 @@ * * * - * gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession .send_rtp_src ! udpsink port=5000 + * gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession .send_rtp_src ! udpsink port=5000 * * Send theora RTP packets through the session manager and out on UDP port 5000. * * * - * gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession name=session .send_rtp_src \ + * gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession 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. Not that this pipeline will not preroll + * Send RTCP packets on port 5001. Note that this pipeline will not preroll * correctly because the second udpsink will not preroll correctly (no RTCP * packets are sent in the PAUSED state). Applications should manually set and * keep (see #gst_element_set_locked_state()) the RTCP udpsink to the PLAYING state. * * * - * Last reviewed on 2007-05-23 (0.10.6) + * Last reviewed on 2007-05-28 (0.10.5) */ #ifdef HAVE_CONFIG_H @@ -1001,13 +1001,13 @@ gst_rtp_session_request_new_pad (GstElement * element, wrong_template: { GST_RTP_SESSION_UNLOCK (rtpsession); - g_warning ("rtpsession: this is not our template"); + g_warning ("gstrtpsession: this is not our template"); return NULL; } exists: { GST_RTP_SESSION_UNLOCK (rtpsession); - g_warning ("rtpsession: pad already requested"); + g_warning ("gstrtpsession: pad already requested"); return NULL; } } -- cgit v1.2.1